THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.

THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.

The Cisco implementation of TCP header compression is an adaptation of a program developed by the University of California, Berkeley (UCB) as part of UCB's public domain version of the UNIX operating system. All rights reserved. Copyright © 1981, Regents of the University of California.

NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED "AS IS" WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE.

IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.

Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks. Third-party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (1110R)

Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.

© 2012 Cisco Systems, Inc. All rights reserved.

MGCP Configuration Guide, Cisco IOS Release 15M&T

- Overview of MGCP and Related Protocols

- Finding Feature Information

- Prerequisites for MGCP and Related Protocols

- Information About MGCP and Related Protocols

- Supported Gateways

- Residential Gateway

- Trunking Gateway

- Toll Fraud Prevention

- Additional References

- Basic MGCP Configuration

- Finding Feature Information

- How to Configure MGCP and Related Protocols

- Configuring a TGW for MGCP

- Configuring a TGW for SGCP

- Configuring an RGW

- Configuring a SDP Aware NSE Mode

- Verifying NSE Mode Configuration

- Verifying the TGW or RGW Configuration

- Blocking New Calls

- Configuration Examples for MGCP and Related Protocols

- Configuring a Cisco AS5300 as a TGW with MGCP Example

- Configuring a Cisco AS5300 as a TGW with SGCP Example

- Configuring a Cisco 3660 as a TGW with MGCP Example

- Configuring a Cisco uBR924 as an RGW Example

- Configuring a Cisco 2620 as an RGW Example

- Additional References

- Feature Information for Basic MGCP Configuration

- Configuring MGCP 1.0

- Finding Feature Information

- Prerequisites for MGCP 1.0

- Information About MGCP 1.0

- MGCP Model

- How to Configure MGCP 1.0

- Identifying Endpoints and Configuring the MGCP Application

- Analog CAS and POTS Lines

- Digital CAS Trunks

- ISUP Signaling Trunks

- FGD-OS Trunks

- Digital VoATM with AAL2 PVC

- Configuring Global MGCP Parameters

- Configuring an MGCP Profile and Profile-Related MGCP Parameters

- Verifying the Configuration

- Troubleshooting Tips

- Configuration Examples for MGCP 1.0

- Cisco uBR925 Using Radio Frequency Interface Example

- Cisco uBR925 Using Ethernet0 Interface Example

- Cisco CVA122 Using Radio Frequency Interface Example

- Cisco 2600 Series as a Residential Gateway Example

- Cisco 3660 Platform as a Trunking Gateway Example

- Cisco MC3810 as a Residential Gateway Example

- Cisco MC3810 as a VoAAL2 Gateway using AAL2 PVCs Example

- Configuring MGCP Basic CLASS and Operator Services

- Finding Feature Information

- Prerequisites for MGCP Basic CLASS and Operator Services

- Restrictions for MGCP Basic CLASS and Operator Services

- Information About MGCP Basic CLASS and Operator Services

- Distinctive Power Ring

- Visual Message Waiting Indicator

- Caller ID

- Caller ID with Call Waiting

- Call Forwarding

- Ring Splash

- Distinctive Call-Waiting Tone

- Message-Waiting Tone

- Stutter Dial Tone

- Off-Hook Warning Tone

- 911 Calls

- Three-Way Calling

- Considerations for Three-way Calling

- Examples of Service-Provider Solutions

- Troubleshooting MGCP Basic CLASS and Operator Services

- Configuration Examples for MGCP Basic CLASS and Operator Services

- Configuring NAS Package for MGCP

- Finding Feature Information

- Prerequisites for NAS Package for MGCP

- Information About NAS Package for MGCP

- How to Configure NAS Package for MGCP

- Configuring the NAS for MGCP

- Configuring Controllers

- Configuring Dialer Interfaces and Routing

- Verifying the NAS Package for MGCP

- Troubleshooting Tips

- MGCP Troubleshooting

- Example Output for show mgcp nas info Command

- Example Output for show mgcp nas dump Command

- Example Output for show mgcp connection Command

- Example Output for show xcsp slot Command

- Example Output for show xcsp port Command

- Example Output for show cdapi Command

- MGCP Debugging

- Example Output for debug mgcp all Command

- Example Output for debug mgcp events Command

- Example Output for debug mgcp packets Command

- Example Output for debug mgcp parser Command

- Example Output for debug mgcp nas Command

- Example Output for debug xcsp Command

- Example Output for debug cdapi Command

- Controller Troubleshooting

- Example Output for show controllers e1 or t1 Command

- Example Output for show voice port Command

- Dialer Interface and Routing Troubleshooting

- Example Output for show dialer map Command

- Example Output for show dialer Command

- Example Output for show interface Command

- Example Output for show ip route Command

- Example Output for debug dialer Command

- Example Output for clear interface Command

- Example Output for debug ppp negotiation Command

- Example Output for debug ppp authentication Command

- Configuration Examples for NAC Package for MGCP

- NAS Package for MGCP Example

- Configuring SGCP RSIP and AUEP Enhancements

- Finding Feature Information

- Prerequisites for SGCP RSIP and AUEP Enhancements

- Restrictions for SGCP RSIP and AUEP Enhancements

- Information About SGCP RSIP and AUEP Enhancements

- How to Configure SGCP RSIP and AUEP Enhancements

- Configuring SGCP RSIP and AUEP Enhancements

- Verifying SGCP RSIP Configuration

- Configuration Examples for SGCP RSIP and AUEP Enhancements

- Disconnected RSIP Messaging Example

- Configuring MGCP Gateway Support

- Finding Feature Information

- Prerequisites for Configuring MGCP Gateway Support

- Information About MGCP Gateway Support

- How to Configure MGCP Gateway Support

- Configuring the MGCP Application

- Configuring the bind Command

- Troubleshooting Tips

- Verifying MGCP Gateway Support

- Configuration Examples for MGCP Gateway Support

- Configuring MGCP CAS MD Package

- Finding Feature Information

- Prerequisites for MGCP CAS MD Package

- Restrictions for MGCP CAS MD Package

- Information About MGCP CAS MD Package

- MD Package

- How to Configure the MGCP CAS MD Package

- Configuring the Incoming Called Number in the MGCP Dial Peer

- Modifying ANI and DNIS Order when Using CAS MD Package

- Configuration Examples for MGCP CAS MD Package

- CAS MD Package Configuration Example

- Cisco AS5850 Configuration Example

- Media and Signaling Authentication and Encryption

- Finding Feature Information

- Prerequisites for Media and Signaling Authentication and Encryption

- Restrictions for Media and Signaling Authentication and Encryption

- Information About Media and Signaling Authentication and Encryption

- Benefits of Media and Signaling Authentication and Encryption

- Feature Design

- MGCP Gateway Behavior and Voice Security Features

- Voice Security Features Interoperability with Endpoints

- How to Configure Media and Signaling Authentication and Encryption Feature

- Installing Cisco CallManager

- Configuring IPsec on Cisco CallManager

- Configuring the Cisco PGW

- Configuring Voice Security Features

- Configuring Secure IP Telephony Calls

- Verifying Voice Security Features

- Configuration Examples for Media and Signaling Authentication and Encryption

- Voice Security Features Example

- Additional References

- Feature Information for Media and Signaling Authentication and Encryption

- Configuring MGCP CAS PBX and AAL2 PVC

- Finding Feature Information

- Prerequisites for MGCP CAS PBX and AAL2 PVC

- Restrictions for MGCP CAS PBX and AAL2 PVC

- Information About MGCP CAS PBX and AAL2 PVC

- How to Configure MGCP CAS PBX and AAL2 PVC

- Configuring the Gateway

- Configuring Subcell Multiplexing for AAL2 Voice

- Configuring the Cable Access Router for SGCP and MGCP

- Verifying the MGCP CAS PBX and AAL2 PVC Configurations

- Configuration Examples for MGCP CAS PBX and AAL2 PVC

- Example 1 MGCP Residential Gateway

- Example 2 MGCP Gateway using Voice over ATM AAL2

- Example 3 MGCP and SGCP EM Wink-Start

- Example 4 SGCP 1.5 CAS PBX using Voice over ATM AAL2

- Example 5 SGCP 1.5 CAS PBX using Voice over IP over ATM AAL5

- Example 6 SGCP 1.5 Analog EM PBX using Voice over ATM AAL2

- Example 7 SGCP 1.5 Analog EM PBX using Voice over IP over ATM AAL5

- Example 8 SGCP 1.5 RGW using Voice over ATM AAL2

- Example 9 SGCP 1.5 RGW using Voice over IP over ATM AAL5

- Secure Tone on MGCP TDM Gateways

- Finding Feature Information

- Prerequisites for Secure Tone on MGCP TDM Gateways

- Restrictions for Secure Tone on MGCP TDM Gateways

- Secure Tone on MGCP TDM Gateways

- How to Configure Secure Tone on MGCP TDM Gateways

- Configuring Secure Tone on MGCP TDM Gateways

- Verifying and Troubleshooting Secure Tone on MGCP TDM Gateways

- Configuration Examples for Secure Tone on MGCP TDM Gateways

- Example Configuring Secure Tone for MGCP TDM Gateways

- Example Verifying Secure Tone for MGCP TDM Gateways

- Additional References

- Feature Information for Secure Tone on MGCP TDM Gateways

- DSP Voice Quality Statistics in DLCX Messages

- Finding Feature Information

- Prerequisites for DSP Voice Quality Statistics in DLCX Messages

- Restrictions for DSP Voice Quality Statistics in DLCX Messages

- Information About DSP Voice Quality Statistics in DLCX Messages

- Cisco PGW 2200

- MGCP

- Voice Quality Statistics

- Quality of Service for Voice

- Voice Quality Parameters for Cisco IOS Release 12.4(4)T and Later Releases

- DSP/EC

- DSP/KF

- DSP/CS

- DSP/RF

- DSP/UC

- DSP/DL

- How to Configure DSP Voice Quality Statistics in DLCX Messages

- Configuring DSP Voice Quality Statistics in DLCX Messages

- What to Do Next

- Troubleshooting Tips

- Verifying DSP Voice Quality Statistics in DLCX Messages

- Configuration Examples for DSP Voice Quality Statistics in DLCX Messages

- Example: Configuring DSP Voice Quality Statistics in DLCX Messages

- Additional References

- Feature Information for DLCP Voice Quality Statistics in DLCX Messages

- Glossary

Overview of MGCP and Related Protocols

This chapter provides overview information on Media Gateway Control Protocol (MGCP) and related protocols.

Note

For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration Library--including library preface and glossary, feature documents, and troubleshooting information--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for MGCP and Related Protocols

- Configure IP routing.

- Configure voice ports.

- Configure VoIP.

- Configure the call agent. (For information on setting up call agents, see the documentation that accompanies the call agents used in your network configuration.)

Information About MGCP and Related Protocols

MGCP is an extension of the earlier version of the protocol Simple Gateway Control Protocol (SGCP) and supports SGCP functionality in addition to several enhancements. Systems using SGCP can easily migrate to MGCP, and MGCP commands are available to enable SGCP capabilities.

An MGCP gateway handles translation between audio signals and the packet network. Gateways interact with a call agent (CA)--also called a media gateway controller (MGC)--that performs signal and call processing on gateway calls. In the MGCP configurations that Cisco IOS supports, a gateway can be a Cisco router, access server, or cable modem, and the CA is a server from a third-party vendor.

Configuration commands for MGCP define the path between the call agent and the gateway, the type of gateway, and the type of calls handled by the gateway.

MGCP uses endpoints and connections to construct a call. Endpoints are sources of or destinations for data, and can be physical or logical locations in a device. Connections can be point-to-point or multipoint.

Similar to SGCP, MGCP uses User Datagram Protocol (UDP) for establishing audio connections over IP networks. However, MGCP also uses hairpinning to return a call to the PSTN when the packet network is not available.

Package Types

A call connection involves a series of events and signals--such as off-hook status, a ringing signal, or a signal to play an announcement--that are specific to the type of endpoint involved in the call.

MGCP groups these events and signals into packages. A trunk package, for example, is a group of events and signals relevant to a trunking gateway; an announcement package is a group of events and signals relevant to an announcement server. MGCP supports the following seven package types:

- Trunk

- Line

- Dual-tone multifrequency (DTMF)

- Generic media

- Real-Time Transport Protocol (RTP)

- Announcement server

- Script

The trunk package and line package are supported by default on certain types of gateways. Although configuring a gateway with additional endpoint package information is optional, you may want to specify packages for your endpoints to add to or override the defaults.

Protocol Benefits

MGCP provides the following benefits:

- Alternative dial tone for VoIP environments--Deregulation in the telecommunications industry gives competitive local-exchange carriers (CLECs) opportunities to provide toll bypass from the incumbent local-exchange carriers (ILECs) by means of VoIP. MGCP enables a VoIP system to control call setup and teardown and Custom Local Area Subscriber Services (CLASS) features for less sophisticated gateways.

- Simplified configuration for static VoIP network dial peers--When you use MGCP as the call agent in a VoIP environment, you need not configure static VoIP network dial peers. The MGCP call agent provides functions similar to VoIP-network dial peers.

Note

Plain old telephone service (POTS) dial peer configuration is still required.

- Migration paths--Systems using earlier versions of the protocol can migrate easily to MGCP.

- Varied network needs supported for the following:

- Interexchange carriers (IXCs) who have no legacy time-division multiplexing (TDM) equipment in their networks and want to deploy a fully featured network that offers both long-distance services to corporate customers and connectivity to local exchange carriers or other IXCs with traditional TDM equipment.

- IXCs who have TDM equipment in their networks and want to relieve network congestion using data technologies to carry voice traffic or to cap the growth of TDM ports. In these situations, the packet network provides basic switched trunking without services or features.

- Competitive competitive local-exchange carriers (CLECs) who want to provide residential and enhanced services.

- Dial-access customers who want enhanced Signaling System 7 (SS7) access capabilities and increased performance, reliability, scalability, and economy.

Supported Gateways

MGCP supports both residential and trunking gateways.

Residential Gateway

A residential gateway (RGW) provides an interface between analog (RJ-11) calls from a telephone and the VoIP network. Examples of RGWs include cable modems and Cisco 2600 series routers. The figure below shows an RGW configuration.

Figure 1

Residential and Trunking Gateways

RGW functionality supports analog POTS calls for both SGCP and MGCP on the Cisco 2600 series routers and Cisco uBR924 cable access router as shown in the table below.

Table 1

RGW Functionality

Functionality

Platform

Cisco 2600 Series

Cisco uBR924

Call waiting

Yes

Yes

Default call-agent address specifiable for each foreign exchange station (FXS) port

--

Yes

Distinctive ringing

--

Yes

Fax and modem calls

Yes

Yes

On-hook caller identification (ID)

--

Yes

Ring splash

--

Yes

Stutter dial tone

Yes

Yes

Trunking Gateway

A trunking gateway (TGW) provides an interface between PSTN trunks and a VoIP network. A trunk can be a DS0, T1, or E1 line. Examples of TGWs include access servers and routers. The figure below shows a TGW configuration.

Figure 2

Trunking Gateways

TGW functionality supports SGCP and MGCP as shown in the table below.

Table 2

TGW Functionality

Functionality

Platform

Cisco AS5300

Cisco 3660

911 outgoing calls on T1 lines

Yes1

Fax and modem calls

Yes

Yes

PRI/ISDN signaling (calls are backhauled to the call agent)

Yes

SS7

Yes

Yes

T1 and E1 interfaces

Yes

Yes

1 Server must have SGCP 1.1+ protocol for Feature Group D Operator Services (FGD-OS)

Toll Fraud Prevention

When a Cisco router platform is installed with a voice-capable Cisco IOS software image, appropriate features must be enabled on the platform to prevent potential toll fraud exploitation by unauthorized users. Deploy these features on all Cisco router Unified Communications applications that process voice calls, such as Cisco Unified Communications Manager Express (CME), Cisco Survivable Remote Site Telephony (SRST), Cisco Unified Border Element (UBE), Cisco IOS-based router and standalone analog and digital PBX and public-switched telephone network (PSTN) gateways, and Cisco contact-center VoiceXML gateways. These features include, but are not limited to, the following:

- Disable secondary dial tone on voice ports--By default, secondary dial tone is presented on voice ports on Cisco router gateways. Use private line automatic ringdown (PLAR) for foreign exchange office (FXO) ports and direct-inward-dial (DID) for T1/E1 ports to prevent secondary dial tone from being presented to inbound callers.

- Cisco router access control lists (ACLs)--Define ACLs to allow only explicitly valid sources of calls to the router or gateway, and therefore to prevent unauthorized Session Initiation Protocol (SIP) or H.323 calls from unknown parties to be processed and connected by the router or gateway.

- Close unused SIP and H.323 ports--If either the SIP or H.323 protocol is not used in your deployment, close the associated protocol ports. If a Cisco voice gateway has dial peers configured to route calls outbound to the PSTN using either time division multiplex (TDM) trunks or IP, close the unused H.323 or SIP ports so that calls from unauthorized endpoints cannot connect calls. If the protocols are used and the ports must remain open, use ACLs to limit access to legitimate sources.

- Change SIP port 5060--If SIP is actively used, consider changing the port to something other than well-known port 5060.

- SIP registration--If SIP registration is available on SIP trunks, turn on this feature because it provides an extra level of authentication and validation that only legitimate sources can connect calls. If it is not available, ensure that the appropriate ACLs are in place.

- SIP Digest Authentication--If the SIP Digest Authentication feature is available for either registrations or invites, turn this feature on because it provides an extra level of authentication and validation that only legitimate sources can connect calls.

- Explicit incoming and outgoing dial peers--Use explicit dial peers to control the types and parameters of calls allowed by the router, especially in IP-to-IP connections used on CME, SRST, and Cisco UBE. Incoming dial peers offer additional control on the sources of calls, and outgoing dial peers on the destinations. Incoming dial peers are always used for calls. If a dial peer is not explicitly defined, the implicit dial peer 0 is used to allow all calls.

- Explicit destination patterns--Use dial peers with more granularity than .T for destination patterns to block disallowed off-net call destinations. Use class of restriction (COR) on dial peers with specific destination patterns to allow even more granular control of calls to different destinations on the PSTN.

- Translation rules--Use translation rules to manipulate dialed digits before calls connect to the PSTN to provide better control over who may dial PSTN destinations. Legitimate users dial an access code and an augmented number for PSTN for certain PSTN (for example, international) locations.

- Tcl and VoiceXML scripts--Attach a Tcl/VoiceXML script to dial peers to do database lookups or additional off-router authorization checks to allow or deny call flows based on origination or destination numbers. Tcl/VoiceXML scripts can also be used to add a prefix to inbound DID calls. If the prefix plus DID matches internal extensions, then the call is completed. Otherwise, a prompt can be played to the caller that an invalid number has been dialed.

- Host name validation--Use the "permit hostname" feature to validate initial SIP Invites that contain a fully qualified domain name (FQDN) host name in the Request Uniform Resource Identifier (Request URI) against a configured list of legitimate source hostnames.

- Dynamic Domain Name Service (DNS)--If you are using DNS as the "session target" on dial peers, the actual IP address destination of call connections can vary from one call to the next. Use voice source groups and ACLs to restrict the valid address ranges expected in DNS responses (which are used subsequently for call setup destinations).

For more configuration guidance, see the " Cisco IOS Unified Communications Toll Fraud Prevention " paper.

Additional References

The following sections provide references related to MGCP.

Related Documents

Related Topic

Document Title

Cisco IOS configuration examples

Cisco Systems Technologies website at http://cisco.com/en/US/tech/index.html . Select a technology category and subsequent hierarchy of subcategories. Click Technical Documentation > Configuration Examples.

Cisco IOS debug command reference

Cisco IOS Debug Command Reference, Release 12.3T at http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123tcr/123dbr/index.htm

Cisco IOS troubleshooting information

Cisco IOS Voice Troubleshooting and Monitoring Guide at http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/voipt_c/index.htm

Cisco IOS voice command reference

Cisco IOS Voice Command Reference, Release 12.3T at http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123tcr/123tvr/index.htm

Cisco IOS Voice Configuration Library preface and glossary

Cisco IOS Voice Configuration Library at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

MIBs

MIBs

MIBs Link

To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://www.cisco.com/go/mibs

Technical Assistance

Description

Link

The Cisco Support website provides extensive online resources, including documentation and tools for troubleshooting and resolving technical issues with Cisco products and technologies.

To receive security and technical information about your products, you can subscribe to various services, such as the Product Alert Tool (accessed from Field Notices), the Cisco Technical Services Newsletter, and Really Simple Syndication (RSS) Feeds.

Access to most tools on the Cisco Support website requires a Cisco.com user ID and password.

http://www.cisco.com/techsupport

Basic MGCP Configuration

This chapter provides basic configuration information for Media Gateway Control Protocol (MGCP) and related protocols.

For more information about related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

How to Configure MGCP and Related Protocols

Note

RGWs are configured only with MGCP.

Configuring a TGW for MGCP

To configure a trunking gateway (TGW) for MGCP, perform this task:

SUMMARY STEPS

1.   mgcp

2.   mgcp call-agent [ipaddr|hostname] [port] service-type mgcp

3.   controller t1 number

4.   ds0-group channel-number timeslots range type none service mgcp

5.   exit

6.   mgcp restart-delay value

7.   mgcp package-capability {s-package | dtmf-package | gm-package | lcs-package | rtp-package | trunk-package | script-package}

8.   mgcp default-package {as-package | dtmf-package | gm-package | rtp-package | trunk-package}

9.   mgcp dtmf-relay {codec | low-bit-rate} mode {cisco | out-of-band}

10.   mgcp modem passthru {cisco | ca}

11.   mgcp sdp simple

12.   exit

DETAILED STEPS

Step 1 mgcp

Initiates the MGCP application.

Example:

Router(config)# mgcp

Step 2 mgcp call-agent [ipaddr|hostname] [port] service-type mgcp

Specifies the call agent's IP address or domain name, the port, and gateway control service type.

Example:

Router(config)# mgcp call-agent [ipaddr |hostname ] [port ] service-type mgcp

Step 3 controller t1 number

Specifies the channel number of the T1 trunk to be used for analog calls and enters controller configuration mode.

Example:

Router(config)# controller t1 number

Step 4 ds0-group channel-number timeslots range type none service mgcp

Configures the channelized T1 time slots to accept the analog calls.

Example:

Router(config-controller)# ds0-group channel-number timeslots range type none service mgcp

Step 5 exit

Exits the current mode.

Example:

Router(config-controller)# exit

Step 6 mgcp restart-delay value

(Optional) Specifies the delay value sent in the RSIP graceful teardown method, in seconds. Range is from 0 to 600. Default is 0.

Example:

Router(config)# mgcp restart-delay value

Step 7 mgcp package-capability {s-package | dtmf-package | gm-package | lcs-package | rtp-package | trunk-package | script-package}

(Optional) Specifies the event packages that are supported on the trunking gateway. Default is trunk-package.

Example:

Router(config)# mgcp package-capability {trunk-package | dtmf-package | gm-package | lcs-package | rtp-package | as-package}

Step 8 mgcp default-package {as-package | dtmf-package | gm-package | rtp-package | trunk-package}

(Optional) Specifies the default event package. Overrides the mgcp package-capability default package.

Example:

Router(config)# mgcp default-package {as-package | dtmf-package | gm-package | rtp-package | trunk-package}

Step 9 mgcp dtmf-relay {codec | low-bit-rate} mode {cisco | out-of-band}

(Optional) Used for relaying digits through the IP network. Default is no mgcp dtmf-relay for all codecs.

Example:

Router(config)# mgcp dtmf-relay {codec | low-bit-rate} mode {cisco | out-of-band}

Step 10 mgcp modem passthru {cisco | ca}

(Optional) Configures the gateway for modem and fax data.

Example:

Router(config)# mgcp modem passthru {cisco | ca}

Step 11 mgcp sdp simple

(Optional) Specifies use of a subset of the session description protocol (SDP). Some call agents require this subset to send data through the network. Default is no mgcp sdp simple.

Example:

Router(config)# mgcp sdp simple

Step 12 exit

Exits the current mode.

Example:

Router(config)# exit

Configuring a TGW for SGCP

Perform this task to configure a trunking gateway (TGW) for Simple Gateway Control Protocol (SGCP):

SUMMARY STEPS

1.   mgcp

2.   mgcp call-agent [ipaddr | hostname] [port] service-type sgcp

3.   controller t1 number

4.   ds0-group channel-number timeslots range type {none | fgdos} [tone_type] [addr_info] service {sgcp | voice}

5.   exit

DETAILED STEPS

Step 1 mgcp

Initiates the MGCP application.

Example:

Router(config)# mgcp

Step 2 mgcp call-agent [ipaddr | hostname] [port] service-type sgcp

Specifies the call agent's IP address or domain name, the port, and gateway control service type.

Example:

Router(config)# mgcp call-agent [ipaddr | hostname ] [port ] service-type sgcp

Step 3 controller t1 number

Specifies the channel number of the T1 trunk to be used for analog calls and enters controller configuration mode.

Example:

Router(config)# controller t1 number

Step 4 ds0-group channel-number timeslots range type {none | fgdos} [tone_type] [addr_info] service {sgcp | voice}

Configures the channelized T1 time slots to accept the analog calls. For type none, use service sgcp. For type fgdos, use service voice.

Example:

Router(config-controller)# ds0-group channel-number timeslots range type {none | fgdos} [tone_type ] [addr_info ] service {sgcp | voice}

Step 5 exit

Exits the current mode.

Example:

Router(config-controller)# exit

Configuring an RGW

To configure a residential gateway (RGW), perform this task:

SUMMARY STEPS

1.   mgcp

2.   mgcp call-agent [ipaddr | hostname] [port] service-type sgcp

3.   dial-peer voice number pots

4.   application MGCPAPP

5.   exit

6.   mgcp package-capability {line-package | dtmf-package | gm-package | rtp-package}

7.   mgcp default-package [line-package | dtmf-package | gm-package]

8.   exit

DETAILED STEPS

Step 1 mgcp

Initiates the MGCP application.

Note

RGWs are configured only with MGCP.

Example:

Router(config)# mgcp

Step 2 mgcp call-agent [ipaddr | hostname] [port] service-type sgcp

Specifies the call-agent IP address or domain name, port, and gateway control service type.

Example:

Router(config)# mgcp call-agent [ipaddr | hostname ] [port ] service-type mgcp

Step 3 dial-peer voice number pots

Sets up the dial peer for a voice port.

Example:

Router(config)# dial-peer voice number pots

Step 4 application MGCPAPP

Selects the MGCP application to run on the voice port.

Example:

Router(config-dial-peer)# application MGCPAPP

Step 5 exit

Exits the current mode.

Example:

Router(config-dial-peer)# exit

Step 6 mgcp package-capability {line-package | dtmf-package | gm-package | rtp-package}

(Optional) Specifies event packages that are supported on the residential gateway. Default is line-package.

Example:

Router(config)# mgcp package-capability {line-package | dtmf-package | gm-package | rtp-package}

Step 7 mgcp default-package [line-package | dtmf-package | gm-package]

(Optional) Specifies the default event package. Overrides the mgcp package-capability command.

Example:

Router(config)# mgcp default-package [line-package | dtmf-package | gm-package]

Step 8 exit

Exits the current mode.

Example:

Router(config)# exit

Configuring a SDP Aware NSE Mode

The Cisco IOS MGCP gateway relies only on the local modem or fax configuration to determine whether Named Signaling Event (NSE) should be used or not for the current call. SDP-aware NSE mode enables the Cisco IOS MGCP gateway to negotiate NSE-based modem and fax features by considering both the local configuration and the remote support for NSE.

Note

Cisco Unified Call Manager (UCM) does not support modem or fax passthrough. This feature should not be enabled when Cisco UCM is the call agent.
>

SUMMARY STEPS

1.   mgcp

2.   mgcp behavior negotiate-nse enable

3.   exit

DETAILED STEPS

Step 1 mgcp

Initiates the MGCP application.

Example:

Router(config)# mgcp

Step 2 mgcp behavior negotiate-nse enable

Enables SDP-aware NSE mode.

Example:

Router(config)# mgcp behavior negotiate-nse enable

Step 3 exit

Exits global configuration mode and returns to privileged EXEC mode.

Example:

Router(config)# exit

Verifying NSE Mode Configuration

SUMMARY STEPS

1.   show mgcp

DETAILED STEPS

Verifying the TGW or RGW Configuration

SUMMARY STEPS

1.   show running-configuration

DETAILED STEPS

Step 1 show running-configuration

Displays the current configuration settings.

Example:

Router(config)# show running-configuration

Blocking New Calls

You can block all new MGCP calls to the router (Step 1) and terminate all existing active calls (Step 2), which means that an active call is not terminated until the caller hangs up.

To block all new calls, use the following commands in global configuration mode:

SUMMARY STEPS

1.   mgcp block-newcalls

2.   no mgcp block-newcalls

DETAILED STEPS

Step 1 mgcp block-newcalls

Prevents the gateway from accepting new calls.

Example:

Router(config)# mgcp block-newcalls

Step 2 no mgcp block-newcalls

Restarts normal MGCP call operation.

Example:

Router(config)# no mgcp block-newcalls

Configuration Examples for MGCP and Related Protocols

- Configuring a Cisco AS5300 as a TGW with MGCP Example

- Configuring a Cisco AS5300 as a TGW with SGCP Example

- Configuring a Cisco 3660 as a TGW with MGCP Example

- Configuring a Cisco uBR924 as an RGW Example

- Configuring a Cisco 2620 as an RGW Example

Configuring a Cisco AS5300 as a TGW with MGCP Example

The following example illustrates a configuration only for MGCP calls. FGD-OS calls are not supported.

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

resource-pool disable

!

ip subnet-zero

ip ftp username smith

ip host B 209.165.200.225

ip host C 209.165.200.226

ip domain-name cisco.com

ip name-server 209.165.202.129

!

mgcp

mgcp request timeout 10000

mgcp call-agent 192.168.10.10 2302

mgcp restart-delay 5

mgcp package-capability gm-package

mgcp package-capability dtmf-package

mgcp package-capability trunk-package

mgcp package-capability rtp-package

mgcp package-capability as-package

mgcp package-capability mf-package

mgcp package-capability script-package

mgcp default-package trunk-package

mta receive maximum-recipients 0

!

controller T1 0

framing esf

clock source line primary

linecode b8zs

ds0-group 0 timeslots 1-24 type none service mgcp

!

controller T1 1

framing esf

clock source line secondary 1

linecode b8zs

ds0-group 0 timeslots 1-24 type none service mgcp

!

controller T1 2

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type none service mgcp

!

controller T1 3

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type none service mgcp

!

voice-port 0:0

!

voice-port 1:0

!

voice-port 2:0

!

voice-port 3:0

!

interface Ethernet0

ip address 192.168.10.9 255.255.255.0

no ip directed-broadcast

!

interface FastEthernet0

ip address 172.22.91.73 255.255.255.0

no ip directed-broadcast

shutdown

duplex auto

speed auto

!

no ip classless

ip route 0.0.0.0 0.0.0.0 172.22.91.1

ip route 209.165.200.225 255.255.255.255 192.168.0.1

no ip http server

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line vty 0 4

login

!

end

Configuring a Cisco AS5300 as a TGW with SGCP Example

The following example illustrates a configuration that supports MGCP and FGD-OS calls:

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

resource-pool disable

!

ip subnet-zero

ip ftp username smith

ip host B 209.165.200.225

ip host C 209.165.200.226

ip domain-name cisco.com

ip name-server 209.165.202.129

!

mgcp

mgcp request timeout 10000

mgcp call-agent 192.168.10.10 2302 sgcp

mta receive maximum-recipients 0

!

controller T1 0

framing esf

clock source line primary

linecode b8zs

ds0-group 0 timeslots 1-24 type none service mgcp

!

controller T1 1

framing esf

clock source line secondary 1

linecode b8zs

ds0-group 0 timeslots 1-24 type fgd-os mf dnis-ani service voice

!

controller T1 2

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type none service mgcp

!

controller T1 3

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type none service mgcp

!

!voice-port 0:0

!

voice-port 1:0

!

voice-port 2:0

!

voice-port 3:0

!

interface Ethernet0

ip address 192.168.10.9 255.255.255.0

no ip directed-broadcast

!

interface FastEthernet0

ip address 172.22.91.73 255.255.255.0

no ip directed-broadcast

shutdown

duplex auto

speed auto

!

no ip classless

ip route 0.0.0.0 0.0.0.0 172.22.91.1

ip route 209.165.200.225 255.255.255.255 192.168.0.1

no ip http server

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line vty 0 4

login

!

end

Configuring a Cisco 3660 as a TGW with MGCP Example

The following example illustrates a platform that does not support FGD-OS calls.

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

memory-size iomem 40

voice-card 1

!

ip subnet-zero

!

mgcp 4000

mgcp call-agent 209.165.202.129 4000

mgcp package-capability gm-package

mgcp package-capability dtmf-package

mgcp package-capability rtp-package

mgcp package-capability as-package

isdn voice-call-failure 0

cns event-service server

!

controller T1 1/0

framing esf

clock source internal

ds0-group 1 timeslots 1-24 type none service mgcp

!

controller T1 1/1

framing esf

clock source internal

ds0-group 1 timeslots 1-24 type none service mgcp

!

voice-port 1/0:1

!

voice-port 1/1:1

!

interface FastEthernet0/0

ip address 209.165.202.140 255.255.255.0

no ip directed-broadcast

load-interval 30

duplex auto

speed auto

!

interface FastEthernet0/1

no ip address

no ip directed-broadcast

no ip mroute-cache

load-interval 30

shutdown

duplex auto

speed auto

!

ip default-gateway 209.165.202.130

ip classless

ip route 209.165.200.225 255.255.255.255 FastEthernet0/0

no ip http server

!

snmp-server engineID local 00000009020000107BD8CD80

snmp-server community public RO

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line vty 0 4

login

!

end

Configuring a Cisco uBR924 as an RGW Example

The following example illustrates a platform that does not support FGD-OS calls.

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

logging buffered 200000 debugging

!

clock timezone - -8

ip subnet-zero

no ip routing

no ip domain-lookup

ip host A 192.168.147.91

ip host C 209.165.200.224

ip host D 209.165.200.225

!

mgcp

mgcp call-agent 192.168.10.10 2490

mgcp package-capability gm-package

mgcp package-capability dtmf-package

mgcp package-capability line-package

mgcp default-package line-package

!

voice-port 0

input gain -3

!

voice-port 1

input gain -3

!

dial-peer voice 1 pots

application MGCPAPP

port 1

!

dial-peer voice 2 pots

application MGCPAPP

port 0

!

interface Ethernet0

ip address 192.168.147.91 255.255.255.0

no ip directed-broadcast

no ip route-cache

no ip mroute-cache

!

interface cable-modem0

ip address negotiated

no ip directed-broadcast

no ip route-cache

no ip mroute-cache

cable-modem downstream saved channel 459000000 20

cable-modem downstream saved channel 699000000 19 2

cable-modem mac-timer t2 100000

no cable-modem compliant bridge

bridge-group 59

bridge-group 59 spanning-disabled

!

ip default-gateway 10.1.1.1

ip classless

no ip http server

!

line con 0

exec-timeout 0 0

transport input none

line vty 0 4

login

!

end

Configuring a Cisco 2620 as an RGW Example

The following example illustrates a platform that does not support FGD-OS calls.

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname D

!

memory-size iomem 10

ip subnet-zero

!

mgcp

mgcp call-agent 172.20.5.20

mgcp package-capability gm-package

mgcp package-capability dtmf-package

mgcp package-capability line-package

mgcp package-capability rtp-package

mgcp default-package line-package

cns event-service server

!

voice-port 1/0/0

!

voice-port 1/0/1

!

dial-peer voice 1 pots

application MGCPAPP

port 1/0/0

!

dial-peer voice 2 pots

application MGCPAPP

port 1/0/1

!

interface Ethernet0/0

no ip address

no ip directed-broadcast

shutdown

!

interface Serial0/0

no ip address

no ip directed-broadcast

no ip mroute-cache

shutdown

no fair-queue

!

interface Ethernet0/1

ip address 172.20.5.25 255.255.255.0

no ip directed-broadcast

!

interface Serial0/1

no ip address

no ip directed-broadcast

shutdown

!

ip default-gateway 209.165.202.130

ip classless

ip route 209.165.200.225 255.255.255.224 Ethernet0/1

no ip http server

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line vty 0 4

login

!

end

Tip

See the "Additional References for MGCP and SGCP" section on page x for related documents, standards, and MIBs.

- See the " Glossary " for definitions of terms in this guide.

Additional References

Related Documents

Related Topic

Document Title

Cisco IOS Voice commands

Cisco IOS Voice Command Reference

Cisco IOS Voice Configuration Library

For more information about Cisco IOS voice features, including feature documents, and troubleshooting information--at

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Technical Assistance

Description

Link

The Cisco Support and Documentation website provides online resources to download documentation, software, and tools. Use these resources to install and configure the software and to troubleshoot and resolve technical issues with Cisco products and technologies. Access to most tools on the Cisco Support and Documentation website requires a Cisco.com user ID and password.

http://www.cisco.com/cisco/web/support/index.html

Feature Information for Basic MGCP Configuration

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Table 3

Feature Information for MGCP Basic Configuration

Feature Name

Releases

Feature Information

Configuring a TGW and RGW for MGCP

12.4(22)Y

SDP Aware NSE Mode

15.1(3)T

Support was added for negotiating remote NSE support by configuring modem pass through on the gateway.

Configuring MGCP 1.0

This chapter provides configuration information on configuring the MGCP 1.0 Including Network-based Call Signaling (NCS) 1.0 and Trunking Gateway Control Protocol (TGCP) 1.0 Profiles feature. The feature implements MGCP 1.0, NCS 1.0, and TGCP 1.0 support in existing MGCP stacks.

Feature benefits include the following:

- MGCP 1.0 provides flexible interoperability with a wide variety of call agents, thus enabling a wide range of solutions.

- MGCP 1.0 contains many improvements over its previous release.

- NCS 1.0 and TGCP 1.0 allow participation in packet cable solutions.

- The ability to interoperate with H.323 and Session Initiation Protocol (SIP) control agents allows leverage of the feature sets available in the different protocols and provides the ability to migrate smoothly from one protocol to another.

For more information about this and related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Feature History for MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles

Release

Modification

12.2(2)XA

This feature was introduced on the following platforms: Cisco CVA122, Cisco uBR924, and Cisco AS5300.

12.2(2)XA1

This feature was implemented on the following platforms: Cisco CVA122, Cisco uBR925, and Cisco AS5300

12.2(2)XB

This feature was implemented on the following platforms: Cisco AS5350 and Cisco AS5400.

12.2(4)T

This feature was implemented on the following platforms: Cisco CVA122, Cisco CVA122E, Cisco uBR925, Cisco 2600 series, Cisco 2650, Cisco 3660, and Cisco MC3810. AAL2 PVC support was introduced for MGCP 1.0 on the Cisco MC3810. Certain gateway features were integrated into MGCP 1.0.

Note

The Cisco AS5300 is not supported in this release.

12.2(8)T

The voice-port (MGCP profile) command was changed to port (MGCP profile) for all platforms supported in this release.

Note

The Cisco AS5300 is not supported in this release.

12.2(13)T

The fax keyword was added to the mgcp playout command.

12.4(24)T3

The maximum number of MGCP profiles that can be configured was increased from 13 (12 plus 1 default) to 29 (28 plus 1 default).

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for MGCP 1.0

Prerequisites are described in the "Prerequisites for Configuring MGCP and Related Protocols" section. In addition, the following apply:

- Ensure that the minimum software requirements are met.Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn.

- Configure Voice over ATM AAL2 PVC (optional step that applies to Cisco MC3810 only). The router that is intending to use the VoAAL2 features must have hardware support for VoAAL2.

- Set up the cable modems, if any. See the documentation for the cable product as listed in the Preface.

Note

IP addresses and host names in these examples are fictitious.

Information About MGCP 1.0

This feature implements the following MGCP protocols on supported Cisco media gateways:

- MGCP 1.0 (RFC 2705)

- Network-based Call Signaling (NCS) 1.0, the MGCP 1.0 profile for residential gateways (RGWs)

- Trunking Gateway Control Protocol (TGCP) 1.0, the MGCP 1.0 profile for trunking gateways (TGWs)

- VoIP--Includes signaling methods under VoIP.

- AAL2 PVC--Includes signaling methods under ATM adaptation layer 2 (AAL2) permanent virtual circuit (PVC).

- Basic/Extended RGW--Includes a collection of residential gateway features supporting channel-associated signaling (CAS). Digital CAS (recEive and transMit, or E&M) interfaces and analog (Foreign Exchange Office [FXO], Foreign Exchange Station [FXS], and E&M) interfaces are supported on platforms with the appropriate voice hardware.

- ISUP--Supports ISDN user part signaling for SS7 trunks.

- FGD-OS--Supports Feature Group D Operator Services signaling over T1 or E1 trunks.

- Incoming CAS--Supports digital CAS interfaces for digital incoming multifrequency tones (MF) CAS wink-start trunks in which an operator at an Operator Services Console can initiate the Operator Interrupt and Busy Line Verify (OI and BLV) functions.

- CAS PBX--Includes CAS private branch exchange (PBX) trunks, digit maps, CAS events, and quarantine buffer software. These features are supported on digital CAS interfaces.

MGCP1.0 is a protocol for the control of VoIP calls by external call-control elements known as media gateway controllers (MGCs) or call agents (CAs). It is described in the informational RFC 2705, published by the Internet Society.

PacketCable is an industry-wide initiative for developing interoperability standards for multimedia services over cable facilities using packet technology. PacketCable developed the NCS and TGCP protocols, which contain extensions and modifications to MGCP while preserving basic MGCP architecture and constructs. NCS is designed for use with analog, single-line user equipment on residential gateways, while TGCP is intended for use in VoIP-to-PSTN trunking gateways in a cable environment. To meet European cable requirements and equipment characteristics, the EuroPacketCable working group has adapted PacketCable standards under the name IP Cablecom .

MGCP Model

MGCP bases its call control and intelligence in centralized call agents, also called media gateway controllers. The call agents issue commands to simple, low-cost endpoints, which are housed in media gateways (MGs), and the call agents also receive event reports from the gateways. MGCP messages between call agents and media gateways are sent with Internet Protocol over User Datagram Protocol (IP/UDP).

The MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles feature provides protocols for RGWs and TGWs, which sit at the border of the packet network to provide an interface between traditional, circuit-based voice services and the packet network. Residential gateways offer a small number of analog line interfaces, while trunking gateways generally manage a large number of digital trunk circuits.

Two basic MGCP constructs are endpoints and connections. An endpoint is a source or sink for call data (RTP/IP) that is flowing through the gateway. A common type of endpoint is found at the physical interface between the POTS (plain old telephone service) or Public Switched Telephone Network (PSTN) service and the gateway; this type of endpoint might be an analog voice port or a digital DS0 group. There are other types of endpoints as well, and some are logical rather than physical. An endpoint is identified by a two-part endpoint name that contains the name of the entity on which it exists (for example, an access server or router) and the local name by which it is known (for example, a port identifier).

A connection is a temporary allocation of resources that enables a call to be completed. One or more connections is necessary to complete a call. Connections have names that identify them with the call to which they belong. Connections can be one-to-one or multipoint. Calls and connections are initiated, modified, and deleted on instructions from call agents.

Call agents manage call flow through standard MGCP commands that are sent to the endpoints under their control. The commands are delivered in standard ASCII text, and may contain session descriptions transmitted in Session Description Protocol (SDP), a text-based protocol. These messages are sent over IP/UDP.

Call agents keep track of endpoint and connection status through the gateway's reporting of standard events that are detected from endpoints and connections. Call agents also direct gateways to apply certain standard signals when a POTS or PSTN connection expects them. For example, when someone picks up a telephone handset, an off-hook event is detected on an endpoint on the residential gateway to which the telephone is connected. The gateway reports the event to a call agent, which orders the gateway to apply the dial-tone signal to the endpoint reporting the off-hook event. The person picking up the handset hears dial tone.

Related events and signals are grouped into standard packages that apply to particular types of endpoints. For instance, the off-hook event is found in the line package, which is associated with analog-line endpoints, which in turn are associated with residential gateways. Standard events, signals, and packages are defined in the NCS, TGCP, and MGCP standards and RFCs listed in the " Preface ."

The figure below shows a hypothetical MGCP network with both residential and trunking gateways. The residential gateway has telephone sets connected to the gateway's FXS voice ports. MGCP or NCS over IP/UDP is used for call control and reporting to the call agent, while Real-Time Transport Protocol (RTP) is used to transmit the actual voice data.

The figure below also shows two trunking gateways with T1 (or E1) connections to the PSTN. Incoming time-division multiplexing (TDM) data is sent through the gateway into the packet network using RTP. MGCP or TGCP over IP/UDP is used for call control and reporting to the call agent. Signaling System 7 (SS7) data travels a different route, however, bypassing the trunking gateway entirely in favor of a specialized signaling gateway, where the signaling data is transformed to ISUP/IP format and relayed to the call agent. Communication between two signaling gateways in the same packet network can be done with Integrated Services Digital Networks User Part over Internet Protocol (ISUP/IP), H.323, or Session Initiation Protocol (SIP).

Figure 3

MGCP Network Model

How to Configure MGCP 1.0

The three tasks listed below configure the MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles feature on a media gateway. The first task names the voice ports or DS1 groups that are serving as MGCP endpoints. This task also associates the ports with an MGCP service type or application and starts the MGCP daemon.

The last two tasks allow you to configure MGCP parameters to meet your requirements. Each MGCP parameter is either a global parameter or a profile-related parameter. When you configure a global MGCP parameter value, it applies to all the MGCP endpoints on the gateway. When you configure a profile-related MGCP parameter value, it applies only to the endpoints associated with the MGCP profile that you are configuring at that moment (an MGCP profile is a user-defined subset of all the MGCP endpoints on the gateway). There is also a predefined MGCP profile named default that you can use to configure profile-related parameters for endpoints that do not belong to a user-defined MGCP profile.

See the following sections for configuration tasks for the MGCP 1.0 including NCS 1.0 and TGCP 1.0 Profiles feature. Each task in the list is identified as either required or optional:

Identifying Endpoints and Configuring the MGCP Application

This task is required. Voice ports or DS0 groups that are acting as MGCP endpoints must be identified and associated with the MGCP application. The commands to identify MGCP endpoints depend on the type of endpoint that you are configuring.

To identify endpoints and configure the MGCP application, use the commands in the appropriate table, beginning in global configuration mode:

Analog CAS and POTS Lines

To identify endpoints and configure the MGCP application for use with analog CAS and POTS lines, use these commands, beginning in global configuration mode:

SUMMARY STEPS

1.   dial-peer voice tag pots

2.   application mgcpapp

3.   port port-number

4.   exit

5.   mgcp [gw-port]

DETAILED STEPS

Step 1 dial-peer voice tag pots

Enters dial-peer configuration mode and specifies the method of voice encapsulation.

Example:

Router(config)# dial-peer voice tag pots

Step 2 application mgcpapp

Enables the MGCP application on this dial peer.

Example:

Router(config-dial-peer)# application mgcpapp

Step 3 port port-number

Associates a dial peer with a specific voice port.

Example:

Router(config-dial-peer)# port port-number

Step 4 exit

Exits the current mode.

Example:

Router(config-dial-peer)# exit

Step 5 mgcp [gw-port]

Initiates the MGCP daemon. The optional argument is the UDP port over which the gateway receives messages from the call agent (the gateway MGCP port number).

Default is 2427.

Example:

Router(config)# mgcp [gw-port ]

Digital CAS Trunks

To identify endpoints and configure the MGCP application for use with digital CAS trunks, use these commands, beginning in global configuration mode:

SUMMARY STEPS

1.   controller {t1 | e1} cntlr-number

2.   mode cas

3.   Do one of the following:

- framing {sf | esf}

- for T1 lines

- or for E1 lines

- framing {crc4 | no-crc4} [australia]

4.   Do one of the following:

- linecode {ami | b8zs}

- for T1 lines

- or for E1 lines

- linecode {ami | hdb3}

5.   ds0-group channel-number timeslots range type type

6.   exit

7.   Do one of the following:

- voice-port slot/port:ds0-group-no

- for Cisco 2600 and Cisco 3600 series

- or for Cisco MC3810

- voice-port slot:ds0-group-no

8.   dial-type {dtmf | mf | pulse}

9.   exit

10.   dial peer voice tag pots

11.   application mgcpapp

12.   port port-number

13.   exit

14.   mgcp [gw-port]

DETAILED STEPS

Step 1 controller {t1 | e1} cntlr-number

Configures a T1 or E1 controller and enters controller configuration mode for the digital CAS port.

Example:

Router(config)# controller {t1 | e1} cntlr-number

Step 2 mode cas

(Required for Cisco MC3810 only) Configures the T1 or E1 controller to support CAS mode.

Example:

Router(config-controller)# mode cas

Step 3 Do one of the following:

Selects frame type for T1 or E1 line.

T1 default is sf. E1 default is crc4.

- framing {sf | esf}

- for T1 lines

- or for E1 lines

- framing {crc4 | no-crc4} [australia]

Example:

Router(config-controller)# framing {sf | esf}

Example:

Example:

Router(config-controller)# framing {crc4 | no-crc4} [australia]

Step 4 Do one of the following:

Specifies the line encoding to use.

T1 default is ami. E1 default is hdb3.

- linecode {ami | b8zs}

- for T1 lines

- or for E1 lines

- linecode {ami | hdb3}

Example:

Router(config-controller)# linecode {ami | b8zs}

Example:

Example:

Router(config-controller)# linecode {ami | hdb3}

Step 5 ds0-group channel-number timeslots range type type

Specifies the DS0 time slots that make up a logical voice port on a T1 or E1 controller and specifies the signaling type by which the router connects to the PBX or PSTN. Use command-line interface (CLI) help (enter ? after type) for valid signaling types.

Example:

Router(config-controller)# ds0-group channel-number timeslots range type type

Step 6 exit

Exits the current mode.

Example:

Router(config-controller)# exit

Step 7 Do one of the following:

Enters voice-port configuration mode.

- voice-port slot/port:ds0-group-no

- for Cisco 2600 and Cisco 3600 series

- or for Cisco MC3810

- voice-port slot:ds0-group-no

Example:

Router(config)# voice-port slot /port :ds0-group-no

Example:

Example:

Router(config)# voice-port slot :ds0-group-no

Step 8 dial-type {dtmf | mf | pulse}

(Required for MF trunks) Specifies the type of out-dialing for voice port interfaces.

Default is dtmf.

Example:

Router(config-voiceport)# dial-type {dtmf | mf | pulse}

Step 9 exit

Exits the current mode.

Example:

Router(config-voiceport)# exit

Step 10 dial peer voice tag pots

Enters dial-peer configuration mode and specifies the method of voice encapsulation.

Example:

Router(config)# dial peer voice tag pots

Step 11 application mgcpapp

Enables the MGCP application on this dial peer.

Example:

Router(config-dial-peer)# application mgcpapp

Step 12 port port-number

Associates a dial peer with a specific voice port.

Example:

Router(config-dial-peer)# port port-number

Step 13 exit

Exits the current mode.

Example:

Router(config-dial-peer)# exit

Step 14 mgcp [gw-port]

Initiates the MGCP daemon. The optional port-number argument is the UDP port over which the gateway receives messages from the call agent (the gateway MGCP port number).

Default is 2427.

Example:

Router(config)# mgcp [gw-port ]

ISUP Signaling Trunks

To identify endpoints and configure the MGCP application for use with Integrated Services Digital Network Upper Part (ISUP) signaling trunks, use these commands, beginning in global configuration mode:

SUMMARY STEPS

1.   controller {t1 | e1} cntlr-number

2.   ds0-group channel-number timeslots range type none service mgcp

3.   exit

4.   mgcp [gw-port]

DETAILED STEPS

Step 1 controller {t1 | e1} cntlr-number

Configures a T1 or E1 controller and enters controller configuration mode for the ISUP trunk port.

Example:

Router(config)# controller {t1 | e1} cntlr-number

Step 2 ds0-group channel-number timeslots range type none service mgcp

Specifies the DS0 time slots that make up a logical voice port on a T1 or E1 controller and specifies the signaling type by which the router connects to the PBX or PSTN.

Specify the type none and service mgcp options to identify this voice port as an MGCP endpoint.

Example:

Router(config-controller)# ds0-group channel-number timeslots range type none service mgcp

Step 3 exit

Exits the current mode.

Example:

Router(config-controller)# exit

Step 4 mgcp [gw-port]

Initiates the MGCP daemon. The optional port number argument allows you to specify the UDP port over which the gateway receives messages from the call agent (the gateway MGCP port number).

Default UDP port number for gateways is 2427.

Example:

Router(config)# mgcp [gw-port ]

FGD-OS Trunks

To identify endpoints and configure the MGCP application for use with Feature Group D Operator Services (FGD-OS) signaling over T1 or E1 trunks, use these commands, beginning in global configuration mode:

SUMMARY STEPS

1.   controller {t1 | e1} cntlr-number

2.   ds0-group channel-number timeslots range type fgd-os service mgcp

3.   exit

4.   mgcp [gw-port]

DETAILED STEPS

Step 1 controller {t1 | e1} cntlr-number

Configures a T1 or E1 controller and enters controller configuration mode for the FGD-OS trunk port.

Example:

Router(config)# controller {t1 | e1} cntlr-number

Step 2 ds0-group channel-number timeslots range type fgd-os service mgcp

Specifies the DS0 time slots that make up a logical voice port on a T1 or E1 controller and specifies the signaling type by which the router connects to the PBX or PSTN.

Specify the type fgd-os option for FGD-OS signaling, and the service mgcp option to identify this voice port as an MGCP endpoint.

Example:

Router(config-controller)# ds0-group channel-number timeslots range type fgd-os service mgcp

Step 3 exit

Exits the current mode.

Example:

Router(config-controller)# exit

Step 4 mgcp [gw-port]

Initiates the MGCP daemon. The optional argument is the UDP port over which the gateway receives messages from the call agent (the gateway MGCP port number).

Default is 2427.

Example:

Router(config)# mgcp [gw-port ]

Digital VoATM with AAL2 PVC

To identify endpoints and configure the MGCP application for use with digital Voice over Asynchronous Transfer Mode (VoATM) with ATM Adaptation Layer 2 (AAL2) Permanent Virtual Circuit (PVC), use these commands, beginning in global configuration mode:

SUMMARY STEPS

1.   controller {t1 | e1} cntlr-number

2.   mode atm

3.   Do one of the following:

- framing {sf | esf}

- for T1 lines

- or for E1 lines

- framing {crc4 | no-crc4} [australia]

4.   Do one of the following:

- linecode {ami | b8zs}

- for T1 lines

- or for E1 lines

- linecode {ami | hdb3}

5.   exit

6.   dial peer voice tag pots

7.   application mgcpapp

8.   port port-number

9.   exit

10.   mgcp [gw-port]

DETAILED STEPS

Step 1 controller {t1 | e1} cntlr-number

Enters dial-peer configuration mode and specifies the method of voice encapsulation.

Example:

Router(config)# controller {t1 | e1} cntlr-number

Step 2 mode atm

Specifies that the controller supports ATM encapsulation and create ATM interface 0.

When the controller is set to ATM mode, the following takes place:

- Controller framing is automatically set to Extended Superframe (ESF).

- The line code is automatically set to B8ZS.

Example:

Router(config-controller)# mode atm

Step 3 Do one of the following:

Selects frame type for T1 or E1 line.

T1 default is sf. E1 default is crc4.

- framing {sf | esf}

- for T1 lines

- or for E1 lines

- framing {crc4 | no-crc4} [australia]

Example:

Router(config-controller)# framing {sf | esf}

Example:

Example:

Router(config-controller)# framing {crc4 | no-crc4} [australia]

Step 4 Do one of the following:

Specifies the line encoding to use.

T1 default is ami. E1 default is hdb3.

- linecode {ami | b8zs}

- for T1 lines

- or for E1 lines

- linecode {ami | hdb3}

Example:

Router(config-controller)# linecode {ami | b8zs}

Example:

Example:

Router(config-controller)# linecode {ami | hdb3}

Step 5 exit

Exits the current mode.

Example:

Router(config-controller)# exit

Step 6 dial peer voice tag pots

Enters dial-peer configuration mode and specifies the method of voice encapsulation.

Example:

Router(config)# dial peer voice tag pots

Step 7 application mgcpapp

Enables the MGCP application on this dial peer.

Example:

Router(config-dial-peer)# application mgcpapp

Step 8 port port-number

Associates a dial peer with a specific voice port.

Example:

Router(config-dial-peer)# port port-number

Step 9 exit

Exits the current mode.

Example:

Router(config-dial-peer)# exit

Step 10 mgcp [gw-port]

Initiates the MGCP daemon. The optional argument is the UDP port over which the gateway receives messages from the call agent (the gateway MGCP port number).

Default is 2427.

Example:

Router(config)# mgcp [gw-port ]

Configuring Global MGCP Parameters

This optional task configures global MGCP parameters on the gateway so that you can set these values to conform to the requirements of the call agent, trunks, or lines that are being used with this gateway. The global parameter values that you configure are associated with every MGCP endpoint that you have identified on this gateway.

In addition to the global MGCP parameters, there are other MGCP parameters that apply only to MGCP profiles on the gateway. For configuration of profile-related parameters, see the Configuring an MGCP Profile and Profile-Related MGCP Parameters.

Note

The only parameter that is common to both profile and global configurations is the call-agent parameter, which is configured with the call-agent command for MGCP profile configuration and with the mgcp call-agent command for the global configuration. These commands are mutually exclusive; whichever command you configure first blocks configuration of the other. For example, if the MGCP profile call-agent command is configured on an endpoint, then you are not allowed to configure the global mgcp call-agentcommand.

To configure global MGCP parameters, complete these steps as needed, beginning in global configuration mode:

SUMMARY STEPS

1.   mgcp call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version]

2.   mgcp behavior {auep | signal} v0.1

3.   mgcp sdp simple

4.   mgcp sdp xpc-codec

5.   mgcp codec type [packetization-period value]

6.   no mgcp timer receive-rtcp

7.   no mgcp piggyback message

8.   mgcp endpoint offset

9.   mgcp persistent {hookflash | offhook | onhook}

10.   mgcp request timeout {timeout-value | max maxtimeout-value}

11.   mgcp dtmf-relay voip codec {all | low-bit-rate} mode {cisco | nse | out-of-band}

12.   mgcp max-waiting-delay value

13.   mgcp restart-delay value

14.   mgcp vad

15.   mgcp ip-tos {high-reliability | high-throughput | low-cost | low-delay | rtp precedence value | signaling precedence value}

16.   mgcp quality-threshold {hwm-cell-loss value | hwm-jitter-buffer value | hwm-latency value | hwm-packet-loss value | lwm-cell-loss value | lwm-jitter-buffer value | lwm-latency value | lwm-packet-loss value}

17.   mgcp playout { adaptive init-value min-value max-value | fax value | fixed init-value

18.   mgcp package-capability [package-type]

19.   mgcp default package [package-type]

DETAILED STEPS

Step 1 mgcp call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version]

Configures parameters for communicating with the call agent (media gateway controller).

Note

You can define a call agent globally with the mgcp call-agent command, or locally for each MGCP profile with the call-agent command, but not both. Whichever command you configure first blocks configuration of the other.

Example:

Router(config)# mgcp call-agent {dns-name | ip-address } [port ] [service-type type ] [version protocol-version ]

Step 2 mgcp behavior {auep | signal} v0.1

(Optional) Forces a gateway to follow the MGCP Version 0.1 protocol for a specified behavior. All other MGCP functionality continues to behave according to the version of MGCP that is specified in the mgcp call-agent command.

- auep --Forces the gateway to reply to an Audit Endpoint (AUEP) command according to the MGCP Version 0.1 specification. If this keyword is used, an AUEP command on an out-of-service endpoint results in a return code of 501. Use this keyword with Cisco IOS Release 12.3(2)T1 or a later release.

- signal --Forces the gateway to handle signaling tones according to the MGCP Version 0.1 specification. The MGCP 0.1 specification treats call signaling tones as on-off tones, which terminate only after a specific MGCP message has been sent to stop the signal. The specifications for MGCP 1.0 and later versions treat call signaling tones as timeout tones, which terminate when the appropriate timeout timer expires. Use this keyword with Cisco IOS Release 12.3(4)T or a later release.

- v0.1 --Selects MGCP Version 0.1.

Example:

Router(config)# mgcp behavior {auep | signal} v0.1

Step 3 mgcp sdp simple

Specifies that a subset of the SDP fields should be used.

Example:

Router(config)# mgcp sdp simple

Step 4 mgcp sdp xpc-codec

Enables codec negotiation in the SDP.

Example:

Router(config)# mgcp sdp xpc-codec

Step 5 mgcp codec type [packetization-period value]

Selects the default codec type and its optional packetization period value.

Example:

Router(config)# mgcp codec type [packetization-period value

Step 6 no mgcp timer receive-rtcp

Disables the timer used by a gateway to disconnect a VoIP call when the IP connectivity is lost with the remote gateway. The timer is known as the RTP Control Protocol (RTCP) transmission interval timer.

Example:

Router(config)# no mgcp timer receive-rtcp

Step 7 no mgcp piggyback message

Disables piggyback messages.

Example:

Router(config)# no mgcp piggyback message

Step 8 mgcp endpoint offset

Increments the voice-port or DS0-group portion of the endpoint name for NCS 1.0.

Example:

Router(config)# mgcp endpoint offset

Step 9 mgcp persistent {hookflash | offhook | onhook}

Enables call-agent notification of the specified type of event.

Example:

Router(config)# mgcp persistent {hookflash | offhook | onhook}

Step 10 mgcp request timeout {timeout-value | max maxtimeout-value}

Specifies how long the gateway waits for a call-agent response to a request before retransmitting the request.

Example:

Router(config)# mgcp request timeout {timeout-value | max maxtimeout-value }

Step 11 mgcp dtmf-relay voip codec {all | low-bit-rate} mode {cisco | nse | out-of-band}

Ensures accurate forwarding of digits with a compressed codec.

Example:

Router(config)# mgcp dtmf-relay voip codec {all | low-bit-rate} mode {cisco | nse | out-of-band}

Step 12 mgcp max-waiting-delay value

Specifies the number of milliseconds to wait after a restart before connecting with the call agent. Range is from 0 to 600,000 (600 seconds). Default is 3000.

If used, these values should be staggered among gateways to avoid having large numbers of gateways connecting with the call agent at the same time after a mass restart.

Example:

Router(config)# mgcp max-waiting-delay value

Step 13 mgcp restart-delay value

Sets the delay value sent in the RestartInProgress (RSIP) graceful teardown, in seconds. Range is from 0 to 600. Default is 0.

Example:

Router(config)# mgcp restart-delay value

Step 14 mgcp vad

Enables voice activity detection (VAD) as a default for MGCP calls. Default is disabled.

Example:

Router(config)# mgcp vad

Step 15 mgcp ip-tos {high-reliability | high-throughput | low-cost | low-delay | rtp precedence value | signaling precedence value}

Enables the IP type of service (ToS) for MGCP-controlled connections.

Example:

Router(config)# mgcp ip-tos {high-reliability | high-throughput | low-cost | low-delay | rtp precedence value | signaling precedence value }

Step 16 mgcp quality-threshold {hwm-cell-loss value | hwm-jitter-buffer value | hwm-latency value | hwm-packet-loss value | lwm-cell-loss value | lwm-jitter-buffer value | lwm-latency value | lwm-packet-loss value}

Sets the jitter buffer size threshold, latency threshold, and packet-loss threshold parameters.

Example:

Router(config)# mgcp quality-threshold {hwm-cell-loss value | hwm-jitter-buffer value | hwm-latency value | hwm-packet-loss value | lwm-cell-loss value | lwm-jitter-buffer value | lwm-latency value | lwm-packet-loss value }

Step 17 mgcp playout { adaptive init-value min-value max-value | fax value | fixed init-value

Configures the jitter buffer packet size in milliseconds for MGCP calls. The default is adaptive 60 4 200

- adaptive init-value min-value max-value --Defines the range for the jitter-buffer packet size. The range for each value is 4 to 250. Default is 60 4 200. Note that init-value must be between min-valueand max-value.

- fax value --Defines the fax playout buffer size. The range is 0 to 700. The default value is 300. The range and default value might vary with different platforms. See the platform digital signal processor (DSP) specifications before setting this value.

- fixed init-value --Defines the fixed size for the jitter-buffer packet size. The range is 4 to 250. There is no default value.

Example:

Router(config)# mgcp playout {adaptive init-value min-value max-value | fax value | fixed init-value }

Step 18 mgcp package-capability [package-type]

Specifies an MGCP package to be supported on this gateway. Configure one package at a time and repeat this command to configure support for more than one package. Available package types vary with the type of gateway.

Example:

Router(config)# mgcp package-capability [package-type ]

Step 19 mgcp default package [package-type]

Defines the package to be used as the default when no package is named with an event. Available package types vary with the type of gateway.

Example:

Router(config)# mgcp default package [package-type ]

Configuring an MGCP Profile and Profile-Related MGCP Parameters

This optional task creates a named, user-defined MGCP profile that consists of a subset of all the MGCP endpoints on this gateway. More than one MGCP profile can be configured on a gateway. Each MGCP profile is associated with a call agent and one or more endpoints. When multiple MGCP profiles are configured, endpoints on a single media gateway can be controlled by different call agents. When each endpoint comes on line, an RSIP (RestartInProgress) message notifies the appropriate call agent of the endpoint's presence.

Note

When partitioning a gateway for multiple call-agent control, the call agents must be coordinated so that there are no overlapping transaction identification numbers.

In addition, this task allows you to configure profile-related MGCP parameters to conform to the requirements of the call agent, trunks, or lines that are being used with the profile's endpoints. These parameters are called profile-related MGCP parameters because they are associated with a particular MGCP profile, or subset of endpoints, and they are configured in MGCP profile configuration mode. Other parameters are considered global MGCP parameters; when they are configured, they apply to all the endpoints on a gateway. Global MGCP parameters are discussed in the Configuring Global MGCP Parameters.

The parameters for an MGCP profile are configured in a special MGCP profile configuration mode that you enter with the mgcp profile command. One or more endpoints are associated with the profile by using the voice-port command in MGCP profile configuration mode.

Note

The only parameter that can be configured in both profile configuration mode and in global configuration mode is call agent, which is configured with the call-agent command for MGCP profiles, and with the mgcp call-agent command for global configurations. These commands are mutually exclusive; whichever command you configure first blocks configuration of the other. For example, if the MGCP profile call-agent command is configured on an endpoint, then you are not allowed to configure the global mgcp call-agent command.

You do not have to define MGCP profiles to configure profile-related parameters. For endpoints that are not associated with a user-defined MGCP profile, the values for profile-related parameters are provided by a predefined profile with the name default . The default profile is configured in the same way that a user-defined MGCP profile is configured, except that the keyword default is used in place of a profile name in the mgcp profile command. The default profile has no association with voice ports or a call agent (the call agent for these endpoints is defined by the global mgcp call-agent command).

In the excerpt below from a show running-config command output, two MGCP profiles are defined: MAX1 and MAX2. Each profile is associated with a different call agent and a different voice port. The MAX1 profile is configured with a value of 3 for the max1 retries parameter and 5 for max2 retries. The MAX2 profile uses the values in the default profile for those parameters. In the MAX2 profile, the MT package is configured as a persistent package. The max1 retries parameter for the default profile is configured with a value of 2. The max2 retries parameter is not configured, so the value used is the default value, which is 7. The MAX2 profile has a value of 2 for the max1 retries parameter and 7 for max2 retries.

!

mgcp profile MAX1

call agent ca1.example.com 4022 service-type mgcp version 1.0

max1 retries 3

max2 retries 5

voice-port 2/1:1

!

mgcp profile MAX2

call-agent ca2.example.com 50031 service-type mgcp version 0.1

package persistent mt-package

voice-port 2/0:1

!

mgcp profile default

max1 retries 2

To configure parameters for a user-defined MGCP profile or for the default profile, use the following commands as appropriate, beginning in global configuration mode:

SUMMARY STEPS

1.   mgcp profile {profile-name | default}

2.   description text

3.   call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version]

4.   voice-port port-number

5.   default command

6.   package persistent package-name

7.   timeout tsmax tsmax-value

8.   timeout tdmax tdmax-value

9.   timeout tdinit tdinit-value

10.   timeout tcrit tcrit-value

11.   timeout tpar tpar-value

12.   timeout thist thist-value

13.   timeout tone mwi mwitone-value

14.   timeout tone ringback ringbacktone-value

15.   timeout tone ringback connection connectiontone-value

16.   timeout tone network congestion congestiontone-value

17.   timeout tone busy busytone-value

18.   timeout tone dial dialtone-value

19.   timeout tone dial stutter stuttertone-value

20.   timeout tone ringing ringingtone-value

21.   timeout tone ringing distinctive distinctivetone-value

22.   timeout tone reorder reordertone-value

23.   timeout tone cot1 continuity1tone-value

24.   timeout tone cot2 continuity2tone-value

25.   max1 lookup

26.   max1 retries value

27.   max2 lookup

28.   max2 retries value

29.   exit

DETAILED STEPS

Step 1 mgcp profile {profile-name | default}

Initiates MGCP profile mode, in which you create and configure a named MGCP profile associated with one or more endpoints, or configure the default profile. Effective with Cisco IOS Release 12.4(24)T3, the maximum number of MGCP profiles that can be configured is increased from 13 (12 plus 1 default) to 29 (28 plus 1 default).

Example:

Router(config)# mgcp profile {profile-name | default}

Step 2 description text

Provides a description for the profile.

Example:

Router(config-mgcp-profile)# description text

Step 3 call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version]

Defines the call agent's DNS name or IP address, UDP port number, service type, and protocol version. (Not used when configuring the default profile.)

Note

You can define a call agent globally with the mgcp call-agentcommand, or locally for each MGCP profile with the call-agentcommand, but not both. Whichever command you configure first blocks configuration of the other.

Example:

Router(config-mgcp-profile)# call-agent {dns-name | ip-address } [port ] [service-type type ] [version protocol-version ]

Step 4 voice-port port-number

Provides the voice port number or DS0 group number for the endpoint to be associated with this MGCP profile. Repeat this command to add more than one endpoint to the profile. (Not used when configuring the default profile.)

Example:

Router(config-mgcp-profile)# voice-port port-number

Step 5 default command

Restores the parameter represented by command to its default value.

Example:

Router(config-mgcp-profile)# default command

Step 6 package persistent package-name

Configures the package type used when reporting persistent events for an MF CAS endpoint type. Valid types are ms-package and mt-package.Default is ms-package.

Example:

Router(config-mgcp-profile)# package persistent package-name

Step 7 timeout tsmax tsmax-value

Configures the maximum timeout value after which MGCP messages are removed from the retransmission queue, in seconds. Range is from 1 to 1000. Default is 20.

Example:

Example:

Router(config-mgcp-profile)# timeout tsmax tsmax-value

Step 8 timeout tdmax tdmax-value

Configures the maximum timeout value for the disconnected procedure (Tdmax), in seconds. Range is from 300 to 600. Default is 600.

Example:

Router(config-mgcp-profile)# timeout tdmax tdmax-value

Step 9 timeout tdinit tdinit-value

Configures the initial waiting delay value (Tdinit) used as the timer for the disconnect procedure, in seconds. Range is from 1 to 30. Default is 15.

Example:

Router(config-mgcp-profile)# timeout tdinit tdinit-value

Step 10 timeout tcrit tcrit-value

Configures the critical timeout value (Tcritical) for the interdigit timer used in digit map matching, in seconds. Range is from 1 to 600. Default is 4.

Example:

Router(config-mgcp-profile)# timeout tcrit tcrit-value

Step 11 timeout tpar tpar-value

Configures the partial timeout value (Tpartial) for the interdigit timer used in digit map matching, in seconds. Range is from 1 to 60. Default is 16.

Example:

Router(config-mgcp-profile)# timeout tpar tpar-value

Step 12 timeout thist thist-value

Configures the packet storage timeout value, in seconds. Range is from 1 to 1100. Default is 30.

Example:

Router(config-mgcp-profile)# timeout thist thist-value

Step 13 timeout tone mwi mwitone-value

Configures the message waiting indicator timeout value, in seconds. Range is from 1 to 600. Default is 16.

Example:

Router(config-mgcp-profile)# timeout tone mwi mwitone-value

Step 14 timeout tone ringback ringbacktone-value

Configures the ringback tone timeout value, in seconds. Range is from 1 to 600. Default is 180.

Example:

Router(config-mgcp-profile)# timeout tone ringback ringbacktone-value

Step 15 timeout tone ringback connection connectiontone-value

Configures the timeout value for ringback tone on connection, in seconds. Range is from 1 to 600. Default is 180.

Example:

Router(config-mgcp-profile)# timeout tone ringback connection connectiontone-value

Step 16 timeout tone network congestion congestiontone-value

Configures the network congestion tone timeout value, in seconds. Range is from 1 to 600. Default is 180.

Example:

Router(config-mgcp-profile)# timeout tone network congestion congestiontone-value

Step 17 timeout tone busy busytone-value

Configures the busy tone timeout value, in seconds. Range is from 1 to 600. Default is 3.

Example:

Router(config-mgcp-profile)# timeout tone busy busytone-value

Step 18 timeout tone dial dialtone-value

Configures the dial tone timeout value, in seconds. Range is from 1 to 600. Default is 16.

Example:

Router(config-mgcp-profile)# timeout tone dial dialtone-value

Step 19 timeout tone dial stutter stuttertone-value

Configures the stutter dial tone timeout value, in seconds. Range is from 1 to 600. Default is 16.

Example:

Router(config-mgcp-profile)# timeout tone dial stutter stuttertone-value

Step 20 timeout tone ringing ringingtone-value

Configures the ringing tone timeout value, in seconds. Range is from 1 to 600. Default is 180.

Example:

Router(config-mgcp-profile)# timeout tone ringing ringingtone-value

Step 21 timeout tone ringing distinctive distinctivetone-value

Configures the distinctive ringing tone timeout value, in seconds. Range is from 1 to 600. Default is 180.

Example:

Router(config-mgcp-profile)# timeout tone ringing distinctive distinctivetone-value

Step 22 timeout tone reorder reordertone-value

Configures the reorder tone timeout value, in seconds. Range is from 1 to 600. Default is 30.

Example:

Router(config-mgcp-profile)# timeout tone reorder reordertone-value

Step 23 timeout tone cot1 continuity1tone-value

Configures the continuity1 tone timeout value, in seconds. Range is from 1 to 600. Default is 3.

Example:

Router(config-mgcp-profile)# timeout tone cot1 continuity1tone-value

Step 24 timeout tone cot2 continuity2tone-value

Configures the continuity2 tone timeout value, in seconds. Range is from 1 to 600. Default is 3.

Example:

Router(config-mgcp-profile)# timeout tone cot2 continuity2tone-value

Step 25 max1 lookup

Enables the DNS lookup procedure after the suspicion threshold is reached. Default is enabled.

Example:

Router(config-mgcp-profile)# max1 lookup

Step 26 max1 retries value

Sets the suspicion threshold number of retries. Range is from 3 to 30. Default is 5.

Example:

Router(config-mgcp-profile)# max1 retries value

Step 27 max2 lookup

Enables the DNS lookup procedure after the disconnect threshold is reached. Default is enabled.

Example:

Router(config-mgcp-profile)# max2 lookup

Step 28 max2 retries value

Sets the disconnect threshold number of retries. Range is from 3 to 30. Default is 7.

Example:

Router(config-mgcp-profile)# max2 retries value

Step 29 exit

Exits the current mode.

Example:

Router(config-mgcp-profile)# exit

Verifying the Configuration

SUMMARY STEPS

1.   show running-configuration

2.   show mgcp [connection | endpoint | profile [profile-name] | statistics

DETAILED STEPS

Step 1 show running-configuration

Displays the current configuration settings.

Example:

Router# show running-configuration

Step 2 show mgcp [connection | endpoint | profile [profile-name] | statistics

Displays the current MGCP settings.

Example:

Router# show mgcp [connection | endpoint | profile [profile-name ] | statistics ]

Troubleshooting Tips

The following suggestions help with troubleshooting:

- Use the show running-config command to verify that the following are properly configured:

- For CAS and POTS endpoints, POTS dial peers are configured with the mgcpapp application.

- The correct packages are enabled in the mgcp package-capability command.

- The mgcp call-agent or call-agent command defines the call agent and service type correctly.

- Reset the MGCP statistical counters with the clear mgcp statisticscommand.

- If RTP traffic is not getting through, make sure that IP routing is enabled. Use the show rtp statisticscommand, then use the debug ip udp command and track down the MGCP RTP packets.

Router# show rtp statistics

RTP Statistics info:

No. CallId Xmit-pkts Xmit-bytes Rcvd-pkts  Rcvd-bytes Lost pkts  Jitter Latenc

1   17492  0x8A      0x5640     0x8A       0x5640     0x0        0x0      0x0

Router# show rtp statistics

RTP Statistics info:

No. CallId Xmit-pkts Xmit-bytes Rcvd-pkts  Rcvd-bytes Lost pkts  Jitter Latenc

1   17492  0xDA      0x8840     0xDB       0x88E0     0x0        0x160    0x0

- If an RSIP message is not received by the call agent, make sure that the mgcp call-agentcommand or the MGCP profile call-agent command is configured with the correct call agent name or IP address and UDP port. Use the show mgcp command or the show mgcp profile command to display this information:

Router# show mgcp

MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE

MGCP call-agent: 172.29.248.51  Initial protocol service is MGCP, v. 1.0

...

MGCP gateway port: 2727, MGCP maximum waiting delay 3000

...

Router# show mgcp profile

MGCP Profile nycprofile

Description: NY branch office configuration

Call-agent: 10.14.2.200 Initial protocol service is MGCP, v. 1.0

- To verify connections and endpoints, use the show mgcp command:

Router# show mgcp connection

Endpoint  Call_ID(C) Conn_ID(I) (P)ort (M)ode (S)tate (C)odec (E)vent[SIFL] (R)esult[EA]

1. S0/DS1-1/5      C=F123AB,5,6  I=0x3  P=16506,16602  M=3  S=4  C=1 E=2,0,0,2  R=0,0

2. S0/DS1-1/6      C=F123AB,7,8  I=0x4  P=16602,16506  M=3  S=4  C=1 E=0,0,0,0  R=0,0

Router# show mgcp endpoint

T1/0 ds0-group 0 timeslots 1-24

T1/1 ds0-group 0 timeslots 1-24

T1/2 ds0-group 0 timeslots 1-24

T1/3 ds0-group 0 timeslots 1-24

- If an MGCP message is rejected, it may be because the remote media gateway does not support SDP mandatory parameters (the o= , s= , and t= lines). If this is the case, configure the mgcp sdp simple command to send SDP messages without those parameters.

- If you notice problems with voice quality, make sure that the cptone (voice-port configuration) command is set for the correct country code. Capturing RTP packets from the sniffer may help to debug the problem, such as whether the payload type or timestamps are set correctly, and so forth.

- To check operation of interfaces, use the show interfacecommand.

- To view information about activity on the T1 or E1 line, use the show controllers command. Alarms, line conditions, and other errors are displayed. The data is updated every 10 seconds; and every 15 minutes, the cumulative data is stored and retained for 24 hours.

- When necessary, you can enable debug traces for errors, events, media, packets, and parser. The command debug mgcp packets can be used to verify that your packets are arriving at the gateway and to monitor message flow in general. Note that there is always a performance penalty when using debug commands. The sample output below shows the use of the optional input-hex keyword to enable display of hexadecimal values.

Router# debug mgcp packets input-hex

Media Gateway Control Protocol input packets in hex value debugging is on

MGCP Packet received -

DLCX 49993 * MGCP 0.1

MGCP Packet received in hex -

44 4C 43 58 20 34 39 39 39 33 20 2A 20 4D 47 43 50 20 30 2E 31 A

send_mgcp_msg, MGCP Packet sent --->

250 49993

Configuration Examples for MGCP 1.0

- Cisco uBR925 Using Radio Frequency Interface Example

- Cisco uBR925 Using Ethernet0 Interface Example

- Cisco CVA122 Using Radio Frequency Interface Example

- Cisco 2600 Series as a Residential Gateway Example

- Cisco 3660 Platform as a Trunking Gateway Example

- Cisco MC3810 as a Residential Gateway Example

- Cisco MC3810 as a VoAAL2 Gateway using AAL2 PVCs Example

Cisco uBR925 Using Radio Frequency Interface Example

This example shows how to set up a Cisco uBR925 as an MGCP residential gateway. The call agent is specified to the cable router (Cisco uBR925, Cisco CVA122, or Cisco CVA122E) by a Dynamic Host Configuration Protocol (DHCP) offer on a cable radio frequency (RF) network. On completion of the DHCP offer, the call agent is set in the MGCP profile on the cable modem. This setting is displayed with the show mgcp profile command. The router does not show the call agent in the CLI.

version 12.2

no service single-slot-reload-enable

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname hydepark

!

logging rate-limit console 10 except errors

!

clock timezone - 0 6

ip subnet-zero

no ip routing

ip domain-name example.com

ip name-server 10.0.0.229

!

ip ssh time-out 120

ip ssh authentication-retries 3

no ip dhcp-client network-discovery

!

interface Ethernet0

ip address 192.168.0.11 255.255.0.0

no ip route-cache

no ip mroute-cache

bridge-group 59

bridge-group 59 spanning-disabled

!

interface cable-modem0

no ip route-cache

no ip mroute-cache

cable-modem boot admin 2

cable-modem boot oper 5

bridge-group 59

bridge-group 59 spanning-disabled

!

ip classless

no ip http server

no ip http cable-monitor

!

snmp-server manager

!

voice-port 0

input gain -2

output attenuation 0

!

voice-port 1

input gain -2

output attenuation 0

!

mgcp

! Use this CLI with NCS 1.0

mgcp endpoint offset

!

mgcp profile default

!

dial-peer voice 100 pots

application MGCPAPP

port 0

!

dial-peer voice 101 pots

application MGCPAPP

port 1

!

line con 0

line vty 0 4

login

!

end

Cisco uBR925 Using Ethernet0 Interface Example

This example shows how to set up a Cisco uBR925 as a residential gateway:

version 12.2

no service single-slot-reload-enable

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname monticello

!

logging rate-limit console 10 except errors

!

clock timezone - 0 6

ip subnet-zero

ip domain-name example.com

ip name-server 10.0.0.229

!

ip ssh time-out 120

ip ssh authentication-retries 3

no ip dhcp-client network-discovery

!

interface Ethernet0

ip address 192.168.0.11 255.255.0.0

no ip route-cache

no ip mroute-cache

bridge-group 59

bridge-group 59 spanning-disabled

!

interface cable-modem0

no ip route-cache

no ip mroute-cache

shutdown

cable-modem boot admin 2

cable-modem boot oper 5

no cable-modem compliant bridge

cable-modem voip clock-internal

bridge-group 59

bridge-group 59 spanning-disabled

!

ip classless

no ip http server

no ip http cable-monitor

!

ip default-gateway 172.16.1.1

!

! We are using the cable modem without its RF interface. So

! route IP traffic out the Ethernet0 interface.

!

ip route 0.0.0.0 0.0.0.0 Ethernet0

!

snmp-server manager

!

voice-port 0

input gain -2

output attenuation 0

!

voice-port 1

input gain -2

output attenuation 0

!

mgcp

!

! The ip address of call agent below can be a FQDN as well.

mgcp call-agent 10.0.0.224 service-type ncs version 1.0

! Use this CLI with NCS 1.0

mgcp endpoint offset

!

mgcp profile default

!

dial-peer voice 100 pots

application MGCPAPP

port 0

!

dial-peer voice 101 pots

application MGCPAPP

port 1

!

line con 0

line vty 0 4

login

!

end

Cisco CVA122 Using Radio Frequency Interface Example

The call agent is specified to the cable router (Cisco uBR925, Cisco CVA122, or Cisco CVA122E) by a DHCP offer on a cable RF network. On completion of the DHCP offer, the call agent is set in the MGCP profile on the cable modem. This setting is displayed with the show mgcp profile command. The router does not show the call agent in the CLI.

version 12.2

no service single-slot-reload-enable

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

service internal

!

hostname mtvernon

!

no logging buffered

logging rate-limit console 10 except errors

!

clock timezone - -5

ip subnet-zero

no ip routing

ip domain-name example.com

ip name-server 10.0.0.229

!

no ip dhcp-client network-discovery

!

interface Ethernet0

ip address 10.20.0.59 255.255.0.0

no ip route-cache

no ip mroute-cache

shutdown

bridge-group 59

bridge-group 59 spanning-disabled

!

interface cable-modem0

no ip route-cache

no ip mroute-cache

cable-modem boot admin 2

cable-modem boot oper 5

bridge-group 59

bridge-group 59 spanning-disabled

!

interface usb0

ip address 10.20.0.59 255.255.0.0

no ip route-cache

no ip mroute-cache

arp timeout 0

bridge-group 59

bridge-group 59 spanning-disabled

!

ip classless

no ip http server

no ip http cable-monitor

!

access-list 1 deny   10.0.0.254

access-list 1 permit any

snmp-server packetsize 4096

snmp-server manager

call rsvp-sync

!

voice-port 0

input gain -2

output attenuation 0

timeouts interdigit 2

!

voice-port 1

input gain -2

output attenuation 0

timeouts interdigit 2

!

mgcp

!

mgcp profile default

!

mgcp profile test

call-agent test service-type ncs version 1.0

!

dial-peer voice 100 pots

application MGCPAPP

port 0

!

dial-peer voice 101 pots

application MGCPAPP

port 1

!

line con 0

exec-timeout 0 0

line vty 0 4

exec-timeout 0 0

login

!

end

Cisco 2600 Series as a Residential Gateway Example

This example shows a Cisco 2620 router being configured as an analog residential gateway:

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname betty-2620

!

voice-port 1/0/0

!

voice-port 1/0/1

!

dial-peer voice 1 pots

application mgcpapp

destination-pattern 100

port 1/0/0

!

dial-peer voice 2 pots

application mgcpapp

destination-pattern 101

port 1/0/1

!

process-max-time 200

!

mgcp 4000

mgcp call-agent 10.14.2.200 4000 service-type mgcp version 1.0

mgcp sdp simple

no mgcp timer receive-rtcp

mgcp sdp xpc-codec

no mgcp piggyback message

mgcp endpoint offset

no mgcp persistent hook on

no mgcp persistent hook flash

mgcp request timeout 1000

mgcp dtmf-relay codec all mode cisco

mgcp max-waiting-delay 600000

mgcp restart-delay 500

mgcp codec g711ulaw packetization-period 10

mgcp ip-tos rtp precedence 7

mgcp quality-threshold lwm-jitter-buffer 59

mgcp quality-threshold lwm-latency 199

mgcp quality-threshold lwm-packet-loss 2

mgcp playout adaptive 100 50 150

mgcp package-capability dtmf-package

mgcp package-capability mf-package

mgcp package-capability rtp-package

mgcp package-capability as-package

isdn voice-call-failure 0

srcp 2428

cns event-service server

!

mgcp profile cisco

call-agent 10.14.2.200 4000 service-type mgcp version 1.0

voice-port 0:1

package persistent mt-package

timeout tsmax 100

timeout tdinit 30

timeout tcrit 600

timeout tpar 600

timeout thist 60

timeout tone mwi 600

timeout tone ringback 600

timeout tone ringback connection 600

timeout tone network congestion 600

timeout tone busy 600

timeout tone dial 600

timeout tone dial stutter 600

timeout tone ringing 600

timeout tone ringing distinctive 600

timeout tone reorder 600

timeout tone cot1 600

timeout tone cot2 600

max1 retries 10

no max2 lookup

max2 retries 10

!

interface Ethernet0/0

ip address 10.14.12.9 255.0.0.0

!

interface Ethernet0/1

no ip address

shutdown

!

ip classless

ip route 0.0.0.0 0.0.0.0 10.14.0.1

no cdp run

!

line con 0

exec-timeout 0 0

line aux 0

line vty 0 4

exec-timeout 0 0

password test

login

!

end

Cisco 3660 Platform as a Trunking Gateway Example

This example shows a Cisco 3660 that is being configured for CAS trunks. The association of endpoints with the MGCP application is made in the dial-peer configuration.

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname charley-3660

!

controller t1 1/0

linecode b8zs

clock source line secondary 1

ds0-group 0 timeslots 1-24 type e&m-winkstart

!

controller t1 1/1

linecode b8zs

clock source line secondary 1

ds0-group 0 timeslots 1-24 type e&m-winkstart

!

ip subnet-zero

!

voice-port 1/0:0

dial-type mf

!

voice-port 1/1:0

dial-type mf

!

dial-peer voice 1 pots

application mgcpapp

destination-pattern 100

port 1/0:0

!

dial-peer voice 2 pots

application mgcpapp

destination-pattern 101

port 1/1:0

!

mgcp 4000

mgcp call-agent 10.14.2.200 4000 service-type mgcp version 1.0

mgcp sdp simple

no mgcp timer receive-rtcp

mgcp sdp xpc-codec

no mgcp piggyback message

mgcp endpoint offset

mgcp persistent hook on

mgcp persistent hook flash

mgcp request timeout 1000

mgcp dtmf-relay codec all mode cisco

mgcp max-waiting-delay 600000

mgcp restart-delay 500

mgcp codec g711ulaw packetization-period 10

mgcp ip-tos rtp precedence 7

mgcp quality-threshold lwm-jitter-buffer 59

mgcp quality-threshold lwm-latency 199

mgcp quality-threshold lwm-packet-loss 2

mgcp playout adaptive 100 50 150

mgcp package-capability dtmf-package

mgcp package-capability mf-package

mgcp package-capability rtp-package

mgcp package-capability as-package

isdn voice-call-failure 0

srcp 2428

cns event-service server

!

mgcp profile cisco

call-agent 10.14.2.200 4000 service-type mgcp version 1.0

voice-port 1/0:0

package persistent mt-package

timeout tsmax 100

timeout tdinit 30

timeout tcrit 600

timeout tpar 600

timeout thist 60

timeout tone mwi 600

timeout tone ringback 600

timeout tone ringback connection 600

timeout tone network congestion 600

timeout tone busy 600

timeout tone dial 600

timeout tone dial stutter 600

timeout tone ringing 600

timeout tone ringing distinctive 600

timeout tone reorder 600

timeout tone cot1 600

timeout tone cot2 600

max1 retries 10

no max2 lookup

max2 retries 10

!

interface FastEthernet0/0

ip address 10.14.12.12 255.0.0.0

speed auto

duplex auto

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

ip classless

ip route 0.0.0.0 0.0.0.0 10.14.0.1

no ip http server

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line vty 0 4

exec-timeout 0 0

password trial

login

!

end

Cisco MC3810 as a Residential Gateway Example

The following example shows a Cisco MC3810 being configured as a residential gateway:

version 12.2

no service pad

service timestamps debug datetime msec

service timestamps log uptime

!

hostname harry

!

logging buffered

!

ip subnet-zero

ip host buffalo 192.168.254.254

!

mgcp

mgcp call-agent 10.14.90.1

!

voice-card 0

codec complexity high

!

controller T1 0

framing esf

linecode b8zs

!

interface Ethernet0

ip address 10.14.92.3 255.255.0.0

!

interface Serial0

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface FR-ATM20

no ip address

shutdown

!

ip default-gateway 10.14.0.1

ip route 192.168.254.0 255.255.255.0 10.14.0.1

!

voice-port 1/1

!

dial-peer voice 1 pots

application mgcpapp

port 1/1

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Cisco MC3810 as a VoAAL2 Gateway using AAL2 PVCs Example

This example shows a Cisco MC3810 being configured as a VoAAL2 gateway using AAL2 PVCs:

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname stella-mc3810

!

network-clock base-rate 56k

ip subnet-zero

no ip domain-lookup

ip host camel 192.168.254.254

ip host buffalo 192.168.254.253

!

mgcp

mgcp call-agent 10.14.117.4 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode nse

mgcp dtmf-relay voaal2 codec all

mgcp modem passthrough nse

mgcp package-capability rtp-package

mgcp tse payload 100

mgcp timer receive-rtcp 100

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 10.14.121.1 255.255.0.0

!

interface Serial0

no ip address

no ip mroute-cache

shutdown

no fair-queue

!

interface Serial1

no ip address

shutdown

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.2 point-to-point

pvc 2/200

vbr-rt 760 760 100

encapsulation aal2

vcci 2

!

interface FR-ATM20

no ip address

shutdown

!

router igrp 1

redistribute connected

network 1.0.0.0

!

ip default-gateway 10.14.0.1

no ip http server

ip classless

ip route 192.168.254.0 255.255.255.0 10.14.0.1

!

dialer-list 1 protocol ip permit

dialer-list 1 protocol ipx permit

voice-port 1/1

!

voice-port 1/2

shutdown

!

voice-port 1/6

shutdown

!

dial-peer voice 1 pots

application mgcpapp

port 1/1

!

line con 0

transport input none

line aux 0

line 2 3

line vty 0 4

password lab

login

!

end

Note

See the "Additional References for MGCP and SGCP" section in the Preface for related documents, standards, and MIBs. See " Glossary " for definitions of terms in this guide.

Configuring MGCP Basic CLASS and Operator Services

This chapter provides information on configuring and troubleshooting the MGCP Basic (CLASS) and Operation Services feature. The feature provides xGCP support for three-way calling on residential and trunking gateways.

Feature benefits include the following:

- The merged SGCP/MGCP software for RGWs, BGWs, and TGWs enables easier development and growth of Cisco and customer solutions.

- MGCP BCOS satisfies the requirements for providing basic CLASS services on Cisco IOS gateways that enable multiple xGCP solutions, particularly residential gateway and IP Centrex.

For more information about this and related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Feature History for MGCP Basic (CLASS) and Operation Services

Release

Modification

12.2(2)T

This feature was introduced.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for MGCP Basic CLASS and Operator Services

Prerequisites are described in the "Prerequisites for Configuring MGCP and Related Protocols" section on page 3 .

Restrictions for MGCP Basic CLASS and Operator Services

- For the Cisco MC3810 series platform, the MGCP BCOS software is supported on an HCM version of an DSP card; it is not supported on an VCM version.

To check the type of DSP card in your Cisco MC3810 series platform, enter a show version command at the EXEC prompt.

- If you have an HCM card, the following line appears as part of the show version information:

1 6-DSP (slot 2) High Performance Compression Module(v01.A0)

- If you have an VCM card, the following line appears as part of the show version information:

1 6-DSP (slot 2) Voice Compression Module(v255.V7)

If you have an HCM card, the MGCP BCOS features will function properly. If you have an VCM card, the feature is not supported.

- The G.728 and G.723 codecs do not support three-way calling.

Information About MGCP Basic CLASS and Operator Services

The MGCP BCOS are a set of calling features, sometimes called "custom calling" features, that use MGCP to transmit voice, video, and data over the IP network. These features are usually found in circuit-based networks. MGCP BCOS brings them to the Cisco IOS gateways on packet-based networks.

The MGCP BCOS software is built on the MGCP CAS PBX and AAL2 software package, and supports MGCP 0.1 and the earlier protocol version Simple Gateway Control Protocol (SGCP) 1.1 and 1.5.

The following MGCP BCOS features are available on residential gateways (RGWs) and business gateways (BGWs):

The following two features can be run as residential gateway (RGW) or trunking gateway (TGW) features:

Distinctive Power Ring

A telephone rings in a distinctive pattern when a call comes in from a predefined telephone number. The following patterns are available:

- R1: One long ring

- R2: Long ring -long ring

- R3: Short ring-short ring-long ring

- R4: Short ring - long ring - short ring

- R5: One short ring

Visual Message Waiting Indicator

A light goes on when a message is waiting for the line.

Caller ID

The calling party's telephone number, date, and time of the call appear on the receiving telephone's display between the first and second rings. A maximum of 18 digits are shown., and private and unlisted numbers are displayed. If the called party answers the phone on the first ring, the calling party's number does not appear.

If the called party has an appropriate name display unit, the calling party's name and telephone number appear on the display. The name and number appear between the first and second rings.

If the calling party has blocked Caller ID from displaying the telephone number, the called party sees "P" for private or "Anonymous" on the display unit.

Caller ID with Call Waiting

If the called party has Caller ID and has enabled the Call Waiting feature, then the calling party's name (if an appropriate display unit is available) and telephone number appear while the called party is on the line with another call.

If the calling party has blocked Caller ID from displaying the name and telephone number, the called party will see "P" for private or "Anonymous" on the display unit.

Call Forwarding

The following scenarios are available:

- The call agent transfers all incoming calls to a designated telephone number when the called number does not answer after a predetermined interval.

- The call agent transfers all incoming calls to a designated telephone number when the called number is busy.

- The call agent transfers all incoming calls to a specific destination when the user enters a code and a destination telephone number that receives the calls. The user is responsible for all charges between the original called number and the receiving telephone number.

- A user can activate Call Forwarding remotely using a touch-tone telephone and a user-defined personal identification number (PIN), which, by default, is the last four digits of the user's telephone number. The original destination telephone emits a Ring Splash when a call is forwarded.

Ring Splash

Also known as Reminder Ring, Ring Splash is activated when the user enables Call Forwarding on the telephone. The user hears Distinctive Power Ring R5 when the line is idle and a call has been forwarded. This reminds the user that Call Forwarding is active.

Distinctive Call-Waiting Tone

The called party hears four audible tone patterns (waiting tones, or WTs) when a call is waiting on the called party's line. The call agent provides the following tone patterns in sequence as the incoming call continues to wait:

- WT1: One short tone

- WT2: Short tone-short tone

- WT3: Short tone-short tone-short tone

- WT4: Short tone-long tone-short tone

Message-Waiting Tone

For users with an active voice mail system, a special dial tone is heard when the user goes off-hook and a message is waiting. The dial tone is a sequence of 10 short tones followed by a steady tone. If the user has a telephone with a visual message indicator, the indicator light goes on when a message is waiting.

Stutter Dial Tone

This tone is used in place of the dial tone to indicate that a message is waiting. When the user goes off-hook, a sequence of three short tones followed by a steady tone is heard.

Off-Hook Warning Tone

The user hears this tone when the telephone is off-hook. The tone is repeated bursts of sound of rising pitch.

911 Calls

The user can make a 911 call to an Emergency Service Bureau (ESB), and the call is maintained as long as the ESB does not hang up. If the user hangs up, the call is maintained. If the user hangs up and picks up the phone again, the call resumes. If the user hangs up and does not pick up the phone again, the ESB can ring the user and resume the call.

This feature is available in SGCP mode on the Cisco 3660 platforms and in MGCP mode on all supported platforms.

Three-Way Calling

The user can create a 3-way call by pressing the switchhook quickly to put the first call on hold, dial a third party, and press the switchhook again quickly to join all parties to the call. This feature is supported on all five platforms.

Considerations for Three-way Calling

- The user who sets up the 3-way call must be connected to a residential gateway, which handles the call setup. TWC is transparent to a trunking gateway.

- Only the G.711u and G.711a codecs support TWC. If any part of a 3-way call is made on a codec other than the G.711u, that codec must be switched to G.711u mode before the second switchhook flash in order for the 3-way call to be set up.

- TWC supports calls originating as Voice over IP or Voice over AAL2 calls, not Voice over ATM or Voice over Frame Relay calls. However, if the network has ATM or Frame Relay as a transport protocol, the VoIP call is completed.

- The user originating the 3-way call is the controller. Each of the two other users on the call can add another person onto the call, which is referred to as call chaining. Those new users can also add another person to the call. However, when five people in total are on the call, adding more users causes voice quality to degrade.

- If the controller of the call hangs up, all the users are disconnected from the call. If one of the non-controller users hangs up, the remaining users are still connected to the call.

- If the controller presses the switchhook quickly for a third time, the last user connected to the call is dropped from the call.

- If two users are on a call and a third user calls one of them, that third user cannot be joined (bridged) into the two-party call.

Examples of Service-Provider Solutions

The Basic CLASS and Operator Services features support MGCP solutions in the following areas:

- Residential cable access

A CLEC can use residential cable access to provide residential customers with basic telephony and data services. CLASS features and Three-way calling, Caller ID with Call Waiting, and Distinctive Call Waiting Tone are features that support these customers. The figure below illustrates a possible residential cable access solution.

Figure 4

Residential Cable Access Solution

Note that, in the figure above, the residential gateway must support the CLASS features and 911 capability.

- IP Centrex and IP PBX

In these solutions, a call agent provides business voice services that are traditionally offered by a circuit-based PBX. CLASS features and Three-way calling, Caller ID with Call Waiting, Distinctive Call Waiting Tone, and Visual Message Waiting Indicator are features suitable for these customers. The figure below illustrates an IP Centrex solution:

Figure 5

IP Centrex Solution

In the figure above, the residential gateway (the Cisco 2600 series platforms) must support the CLASS features.

- Integrated access

A CLEC or IXC can provide small, medium, and large businesses with integrated voice and data access services. The integrated access device can be located at the central office or on the customer's premises. Access to the subscriber can be analog or digital, and transport of voice and data can be over IP, Frame Relay, or ATM. CLASS features and Three-way calling, Caller ID with Call Waiting, Distinctive Call Waiting Tone, and Visual Message Waiting Indicator are features suitable for these customers. The figure below illustrates an integrated access solution.

Figure 6

Integrated Access Solution

In the figure above, the residential gateway (the Cisco 2600 series and Cisco MC3810 series platforms) must support the CLASS features.

- Telecommuter or small-office-home-office solution

The figure below illustrates a telecommuter or small-office-home-office solution:

Figure 7

Telecommuter or Small Office-Home Office Solution

In the figure above, the residential gateway must support the CLASS features.

Other solutions are possible using the MGCP open protocol.

Troubleshooting MGCP Basic CLASS and Operator Services

No new or modified configuration tasks are required to initiate MGCP Basic CLASS and Operator Services. MGCP BCOS co resides with MGCP CAS PBX and AAL2 PVC software, for which configuration activities are required. These are discussed in " Appendix A: Configuring MGCP CAS PBX and AAL2 PVC .

The following MGCP BCOS features do not work on telephones from all manufacturers when the telephones are connected to a Cisco MC3810 series platform:

- CID - Caller ID

- VMWI - Visual Message Waiting Indicator

- CIDCW - Caller ID with Call Waiting

The table below summarizes the findings for the models tested.

Table 4

Telephones and Feature Capabilities

Telephone

CID

VMWI

CIDCW

Casio TI-345

Y

--

N

Casio TI-360

Y

--

N

Dial Digital CP-2892C

Y

Y

Y

GE 29299GE1-A

Y

--

Y

Panasonic KX-TSC7

Y

N

N

Panasonic KX-TSC55-b

Y

Y

Y

Sony IT-ID80

Y

--

Y

To correct this operation, change the idle voltage in the voice port from low to high.

To change the voice port idle voltage, perform these additional steps:

- If the phone is already connected to the voice port, lift the phone's handset.

- If the phone is not connected to the voice port, do the following:

1. Attach the phone to the voice port.

2. Do a "shut" to the voice port.

3. Do a "no shut" to the voice port.

4. Lift the phone's handset.

Configuration Examples for MGCP Basic CLASS and Operator Services

No new or modified configuration settings are needed to implement MGCP Basic CLASS and Operator Services. See the MGCP CAS PBX and AAl2 PVC setup in "Appendix A: Configuring MGCP CAS PBX and AAL2 PVC" for sample configurations.

Tip

See the "Additional References for MGCP and SGCP" section on page x for related documents, standards, and MIBs and the " Glossary " for definitions of terms in this guide.

Configuring NAS Package for MGCP

This chapter provides information on configuring the Network Access Server (NAS) Package for MGCP feature. The feature adds support for the MGCP NAS package on universal gateways. Data calls can be terminated on a trunking media gateway that is serving as a NAS. Trunks on the NAS are controlled and managed by a call agent supporting MGCP for both voice and data calls. The call agent must support the MGCP NAS package.

Key feature benefits derive from the presence of universal ports that are able to terminate both voice and data calls under control of the MGCP call agent. These benefits include the following:

- Cost savings

- Sharing of trunks (T1 or E1) for dial and voice services

- Collapsed IP backbone infrastructure

- Simplified operations and management

- Increased revenue

- Optimized utilization of trunk (T1 or E1) resources

- Flexibility in deploying new services

- Flexibility in access network engineering

For more information about this and related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Feature History for NAS Package for MGCP

Release

Modification

12.2(2)XB

This feature was introduced on the Cisco AS5350 and Cisco AS5400.

12.2(11)T

This feature was implemented on the Cisco AS5850.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for NAS Package for MGCP

Prerequisites are described in the "Prerequisites for Configuring MGCP and Related Protocols" section on page 3 . In addition, the following apply:

- Configure a data network.

- Configure MGCP.

Information About NAS Package for MGCP

This feature adds support for the Network Access Server Package for Media Gateway Control Protocol package on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateways. With this implementation, data calls can be terminated on a trunking media gateway that is serving as a network access server (NAS). Trunks on the NAS are controlled and managed by a call agent that supports Media Gateway Control Protocol (MGCP) for both voice and data calls. The call agent must support the MGCP NAS package.

These capabilities are enabled by the universal port functionality of the Cisco AS5350, Cisco AS5400, and Cisco AS5850, which allows these platforms to operate simultaneously as network access servers and voice gateways to deliver universal services on any port at any time. These universal services include dial access, real-time voice and fax, wireless data access, and unified communications.

The MGCP NAS package implements signals and events to create, modify, and tear down data calls. The events include signaling the arrival of an outbound call (IP to Public Switched Telephone Network [PSTN]) to the media gateway controller (call agent), reporting carrier loss and call authorization status, and receiving callback requests. The following types of calls can be terminated as data calls:

- Data within the voice band (analog modem)

- ISDN data (digital modem)

- Data over voice when using a call agent that recognizes this call type and delivers these calls as digital data to the NAS

The NAS package provides MGCP capabilities for data calls on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 that support all the dial-in and dial-out services, including the following:

- Virtual Private Network (VPN) with Layer 2 Tunneling Protocol (L2TP)

- Scalable Multichassis Multilink PPP (MMP) across multiple channels

- MGCP 1.0 and MGCP 0.1

- Call preauthentication with MGCP dial calls

Resource pool management can be used to manage dial ports when dialed number identification service (DNIS) preauthentication is enabled. The NAS returns an error with a preauthentication failure code to the call agent, which releases the call gracefully with a busy cause. Refer to the Cisco IOS Release 12.3 Configuration Guides and Command References , for more information about dial-pool management, and for more information about authentication, authorization, and accounting (AAA) preauthentication services.

The figure below shows a typical network topology for universal port media gateways.

Figure 8

Media Gateways Operating As Network Access Servers

How to Configure NAS Package for MGCP

With the Network Access Server Package for Media Gateway Control Protocol feature, the NAS supports both data and voice calls, which can be managed from a single call agent that supports MGCP with the NAS package. The NAS package provides the interface to a call agent (media gateway controller) for handling modem calls that terminate on the NAS and that originate from the PSTN, including callback requests. Results of AAA authorization and preauthorization requests from the NAS are reported to the call agent as notifications.

See the following sections for configuration tasks for the Network Access Server Package for Media Gateway Control Protocol feature. Each task in the list is identified as either required or optional.

Configuring the NAS for MGCP

In this task, MGCP is configured on the trunking gateway (NAS), and the NAS package is set as the default package. The steps that are listed are the minimum needed to configure MGCP on the NAS. For more commands and optional settings for MGCP, see the documents listed in the "Related Documents" section on page xi .

To configure the NAS Package for MGCP feature, use the following commands in global configuration mode:

SUMMARY STEPS

1.   mgcp [gw-port]

2.   mgcp call-agent {dns-name | ip-address} [ca-port] [service-type type] [version protocol-version]

3.   mgcp default-package nas-package

DETAILED STEPS

Step 1 mgcp [gw-port]

Allocates resources for MGCP and starts the MGCP daemon.

If no port is specified, the command defaults to port 2427.

Example:

Router(config)# mgcp [gw-port ]

Step 2 mgcp call-agent {dns-name | ip-address} [ca-port] [service-type type] [version protocol-version]

Configures the gateway with the address and protocol of the call agent (media gateway controller). Make sure to specify a call agent that supports the NAS package.

Example:

Router(config)# mgcp call-agent {dns-name | ip-address } [ca-port ] [service-type type ] [version protocol-version ]

Step 3 mgcp default-package nas-package

(Optional) Defines the default package to be used for MGCP signaling. For this feature, specify the NAS-Package. Default generally used on trunking gateways is Trunk-Package and can be left unchanged.

Example:

Router(config)# mgcp default-package nas-package

Configuring Controllers

In this task, in addition to the standard controller commands, you configure a T1 or E1 controller for external signaling control by MGCP. You can also set the AAA preauthentication timer to expire after a certain number of milliseconds have elapsed without a response from the AAA server and indicate whether the call should be accepted or rejected if no response occurs before the timer expires.

To configure a controller to use the Network Access Server Package for Media Gateway Control Protocol feature, use the following commands beginning in global configuration mode:

SUMMARY STEPS

1.   controller {t1 | e1} slot/port

2.   Do one of the following:

- framing {sf | esf}

- for TI lines

- or for E1 lines

- framing {crc4 | no-crc4} [australia]

3.   extsig mgcp

4.   guard-timer milliseconds [on-expiry {accept | reject}]

5.   Do one of the following:

- linecode {ami | b8zs}

- for T1 lines

- or for E1 lines

- linecode {ami | hdb3}

6.   ds0-group channel-number timeslots range type none service mgcp

7.   exit

DETAILED STEPS

Step 1 controller {t1 | e1} slot/port

Configures a T1 or E1 controller and enters controller configuration mode.

Example:

Router(config)# controller {t1 | e1} slot/port

Step 2 Do one of the following:

Selects the frame type for the T1 or E1 trunk.

T1 default is sf. E1 default is crc4.

- framing {sf | esf}

- for TI lines

- or for E1 lines

- framing {crc4 | no-crc4} [australia]

Example:

Router(config-controller)# framing {sf | esf}

Example:

Example:

Router(config-controller)# framing {crc4 | no-crc4} [australia]

Step 3 extsig mgcp

Configures external signaling control by MGCP for this controller. For T3 trunks, each logical T1 must be configured with the extsig mgcpcommand.

Example:

Router(config-controller)# extsig mgcp

Step 4 guard-timer milliseconds [on-expiry {accept | reject}]

(Optional) Sets a guard timer for the number of milliseconds to wait for a AAA server to respond to a preauthentication request before expiring. Also specifies the default action to take when the timer expires without a response from AAA.

Example:

Router(config-controller)# guard-timer milliseconds [on-expiry {accept | reject}]

Step 5 Do one of the following:

Specifies the line encoding to use.

T1 default is ami. E1 default is hdb3.

- linecode {ami | b8zs}

- for T1 lines

- or for E1 lines

- linecode {ami | hdb3}

Example:

Router(config-controller)# linecode {ami | b8zs}

Example:

Example:

Router(config-controller)# linecode {ami | hdb3}

Step 6 ds0-group channel-number timeslots range type none service mgcp

Specifies the DS0 time slots that make up a logical voice port on a T1 or E1 controller and specifies the signaling type by which the router connects to the PBX or PSTN.

Example:

Router(config-controller)# ds0-group channel-number

Example:

timeslots range type none service mgcp

Step 7 exit

Exits the current mode.

Example:

Router(config-controller)# exit

Configuring Dialer Interfaces and Routing

This set of tasks configures dial-on-demand routing (DDR) on a dialer interface that is under external call control by MGCP.

DDR refers to a collection of Cisco features that allows two or more Cisco routers to establish a dynamic connection over simple dial-up facilities to route packets and exchange routing updates on an as-needed basis. DDR is used for low-volume, periodic network connections over the PSTN or an ISDN. A connection is automatically established whenever interesting traffic is detected; during configuration you define what constitutes interesting traffic.

ISDN B channels, synchronous serial interfaces, and asynchronous interfaces can all be converted to dialer interfaces using dialer interface configuration commands.

DDR provides several functions. First, DDR spoofs, or pretends, that there are established configured routes to provide the image of full-time connectivity using the dialer interfaces. When the routing table forwards a packet to a dialer interface, DDR filters out the interesting packets for establishing, maintaining, and releasing switched connections. Internetworking is achieved over the DDR-maintained connection using PPP or other WAN encapsulation techniques.

The encapsulation methods available depend on the physical interface being used. Cisco supports PPP, High-Level Data Link Control (HDLC), Serial Line Internet Protocol (SLIP), and X.25 data-link encapsulations for DDR. PPP is the recommended encapsulation method because it supports multiple protocols and is used for synchronous, asynchronous, or ISDN connections. In addition, PPP performs address negotiation and authentication, and it is interoperable with different vendors.

There are two ways of setting up addressing on dialer interfaces:

- Applying a subnet to the dialer interfaces--Each site with a dialer interface is given a unique node address on a shared subnet for use on its dialer interface. This method is similar to numbering a LAN or multipoint WAN, and it simplifies the addressing scheme and creation of static routes.

- Using unnumbered interfaces--Similar to using unnumbered addressing on leased-line point-to-point interfaces, the address of another interface on the router is borrowed for use on the dialer interface. Unnumbered addressing takes advantage of the fact that there are only two devices on the point-to-point link.

DDR uses manually entered static network protocol routes. This eliminates the use of a routing protocol that broadcasts routing updates across the DDR connection, causing unnecessary connections.

Similar to the function provided by an Address Resolution Protocol (ARP) table, dialer map statements translate next-hop protocol addresses to telephone numbers. Without statically configured dialer maps, DDR call initiation cannot occur. When the routing table points at a dialer interface, and the next-hop address is not found in a dialer map, the packet is dropped.

Authentication in DDR network design provides two functions: security and dialer state. As most DDR networks connect to the PSTN, it is imperative that a strong security model be implemented to prevent unauthorized access to sensitive resources. Authentication also allows the DDR code to keep track of what sites are currently connected and provides for building of Multilink PPP bundles.

In summary, the following main tasks are involved in configuring the dialer interface and routing:

- Specification of interesting traffic--What traffic type should enable the link?

- Definition of static routes--What route do you take to get to the destination?

- Configuration of dialer information--What number do you call to get to the next-hop router, and what service parameters do you use for the call?

For MGCP NAS, configuration of dialer interfaces entails the use of the dialer extsig command in interface configuration mode, which enables the External Call Service Provider (XCSP) subsystem to provide an interface between the Cisco IOS software and the MGCP protocol. The XCSP subsystem enables services such as modem call setup and teardown for the dialer interface.

To configure the dialer interface and routing, use the following commands beginning in global configuration mode:

SUMMARY STEPS

1.   interface dialer-name

2.   Do one of the following:

- ip unnumbered interface-number

- ip address ip-address subnet-mask [secondary]

3.   encapsulation ppp

4.   dialer in-band [no-parity | odd-parity]

5.   dialer idle-timeout seconds [inbound | either]

6.   dialer map protocol next-hop-address [name host-name] [dial-string[: isdn-subaddress]]

7.   dialer extsig

8.   dialer-group number

9.   no cdp enable

10.   ppp authentication chap

11.   exit

12.   dialer list number protocol protocol-name {permit | deny [list access-list-number | access-group]}

13.   ip route prefix mask {ip-address | interface-type interface-number} [distance] [tag tag] [permanent]

DETAILED STEPS

Step 1 interface dialer-name

Enters interface mode for the dialer interface.

Example:

Router(config)# interface dialer-name

Step 2 Do one of the following:

Enables IP processing on the dialer interface, configures the dialer interface not to have an explicit IP address, and assigns the IP address of the loopback interface instead. This command helps conserve IP addresses.

- ip unnumbered interface-number

- ip address ip-address subnet-mask [secondary]

Example:

Router(config-if)# ip unnumbered interface-number

Example:

Example:

Router(config-if)# ip address ip-address subnet-mask [secondary]

Step 3 encapsulation ppp

Sets encapsulation type for PPP.

Example:

Router(config-if)# encapsulation ppp

Step 4 dialer in-band [no-parity | odd-parity]

Specifies that dial-on-demand routing (DDR) is to be supported. The in-band keyword specifies that the same interface that sends the data performs call setup and teardown operations between the router and an external dialing device such as a modem.

By default, no parity is applied to the dialer string.

Example:

Router(config-if)# dialer in-band [no-parity | odd-parity]

Step 5 dialer idle-timeout seconds [inbound | either]

Specifies the duration of idle time before a line is disconnected.

Default direction is outbound. Default idle time is 120 seconds.

Example:

Router(config-if)# dialer idle-timeout seconds [inbound | either]

Step 6 dialer map protocol next-hop-address [name host-name] [dial-string[: isdn-subaddress]]

Configures a serial interface to make digital calls or to accept incoming calls from a specified location and to authenticate if so configured.

Example:

Router(config-if)# dialer map protocol next-hop-address [name host-name ] [dial-string [:isdn-subaddress ]]

Step 7 dialer extsig

Specifies an interface for the initiation and termination of digital calls for external signaling protocols. Only one dialer with external signaling per NAS is permitted.

Example:

Router(config-if)# dialer extsig

Step 8 dialer-group number

Controls access by configuring an interface to belong to a specific dialing group.

Example:

Router(config-if)# dialer-group number

Step 9 no cdp enable

Disables Cisco Discovery Protocol (CDP) on the interface.

Example:

Router(config-if)# no cdp enable

Step 10 ppp authentication chap

Enables Challenge Handshake Authentication Protocol (CHAP) authentication on the interface.

Example:

Router(config-if)# ppp authentication chap

Step 11 exit

Exits the current mode.

Example:

Router(config-if)# exit

Step 12 dialer list number protocol protocol-name {permit | deny [list access-list-number | access-group]}

Defines a DDR dialer list for dialing by protocol or by a combination of a protocol and a previously defined access list. Each dialer interface can have only one dialer group, but the same dialer list can be assigned to multiple interfaces (using the dialer-group command).

Example:

Router(config)# dialer list number protocol protocol-name {permit | deny [list access-list-number | access-group ]}

Step 13 ip route prefix mask {ip-address | interface-type interface-number} [distance] [tag tag] [permanent]

Establishes a static route. Because you do not want dynamic routing protocols running across the DDR links, you manually configure static routes.

Example:

Router(config)# ip route prefix mask {ip-address | interface-type interface-number } [distance ] [tag tag ] [permanent]

Verifying the NAS Package for MGCP

To verify configuration, use the following commands.

SUMMARY STEPS

1.    Use the following command to display the running configuration to verify configured parameters for MGCP, controllers, dialer interfaces, and routing:

2.    Use the following command to display MGCP configurations for NAS:

DETAILED STEPS

Step 1 Use the following command to display the running configuration to verify configured parameters for MGCP, controllers, dialer interfaces, and routing:

Example:

Router# show running-configuration

The following example shows the configuration for serial interface 1:

Example:

Router# show running-configuration interface serial 1

Building configuration...

Current configuration:

!

interface Serial1

no ip address

no ip directed-broadcast

no ip route-cache

no ip mroute-cache

shutdown

!

end

Step 2 Use the following command to display MGCP configurations for NAS:

Example:

Router# show mgcp nas

{dump

slot

port

channel |

info

The following is sample output from the show mgcp nas dump command:

Example:

Router# show mgcp nas dump 7 0 23

Slot 7 state= Up

Port 0 state= Up

State  Idle  PortCb=0x630DE864 ss_id=0x0 handle=0x0

bearer cap=Modem call_id= conn_id=

Events req-

4d21h:

callp=0x62D137D4 - state=MGCP_CALL_IDLE - data_call No

Endpt name=S7/DS1-0/23

The following is sample output from the show mgcp nas info command:

Example:

Router# show mgcp nas info

Slot 7 state= Up

Port 0 state= Up

ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID XX XX XX

XX XX XX XX XX

Channel State Legend

NP=Not Present, OO=Out Of Service, ID=Idle, US=In Use

CI=Connection in progress, RI=In Release in progress

RO=Out Release in progress, DN=Down, SH=Shutdown

XX=Unconfigurable

Troubleshooting Tips

In addition, a number of show and debug commands are useful for troubleshooting the Network Access Server Package for Media Gateway Control Protocol feature. These commands are listed in the following sections:

MGCP Troubleshooting

To display detailed information on the MGCP application and operations, use the following commands in privileged EXEC mode:

Command

Purpose

show mgcp nas info

Router# show mgcp nas info

Displays status of the MGCP data channels.

See Example Output for show mgcp nas info Command.

show mgcp nas dump

slot port chan

Router# show mgcp nas dump slot port chan

Displays status and details about the specified MGCP data slot, port, and channel.

See Example Output for show mgcp nas dump Command.

show mgcp connection

Router# show mgcp connection

Displays active MGCP connections on the router.

See Example Output for show mgcp connection Command.

show xcsp slot slot-num

Router# show xcsp slot slot-num

Displays the status of a router slot under the control of the External Call Service Provider (XCSP) subsystem.

See Example Output for show xcsp slot Command.

show xcsp port  slot port

Router# show xcsp port slot

port

Displays the status of a port under the control of the External Call Service Provider (XCSP) subsystem.

See Example Output for show xcsp port Command.

show cdapi

Router# show cdapi

Displays information about the call distributor application programming interface (CDAPI), which is the internal API that provides an interface between the MGCP signaling stacks and applications.

See Example Output for show cdapi Command.

Example Output for show mgcp nas info Command

The following is sample output from the show mgcp nas info command:

Router# show mgcp nas info

Slot 7 state= Up

Port 0 state= Up

ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID ID XX XX XX

XX XX XX XX XX

Channel State Legend

NP=Not Present, OO=Out Of Service, ID=Idle, US=In Use

CI=Connection in progress, RI=In Release in progress

RO=Out Release in progress, DN=Down, SH=Shutdown

XX=Unconfigurable

Example Output for show mgcp nas dump Command

The following is sample output from the show mgcp nas dump command:

Router# show mgcp nas dump 7 0 23

Slot 7 state= Up

Port 0 state= Up

State  Idle  PortCb=0x630DE864 ss_id=0x0 handle=0x0

bearer cap=Modem call_id= conn_id=

Events req-

4d21h:

callp=0x62D137D4 - state=MGCP_CALL_IDLE - data_call No

Endpt name=S7/DS1-0/23

Example Output for show mgcp connection Command

The following is sample output from the show mgcp connection command for Voice over IP (VoIP) connections:

Router# show mgcp connection

Endpoint   Call_ID(C) Conn_ID(I) (P)ort (M)ode (S)tate (C)odec (E)vent[SIFL] (R)esult[EA]

1. S0/DS1-0/1 C=103,23,24 I=0x8 P=16586,16634 M=3 S=4,4 C=5 E=2,0,0,2 R=0,0

2. S0/DS1-0/2 C=103,25,26 I=0x9 P=16634,16586 M=3 S=4,4 C=5 E=0,0,0,0 R=0,0

3. S0/DS1-0/3 C=101,15,16 I=0x4 P=16506,16544 M=3 S=4,4 C=5 E=2,0,0,2 R=0,0

4. S0/DS1-0/4 C=101,17,18 I=0x5 P=16544,16506 M=3 S=4,4 C=5 E=0,0,0,0 R=0,0

5. S0/DS1-0/5 C=102,19,20 I=0,6 P=16572,16600 M=3 S=4,4 C=5 E=2,0,0,2 R=0,0

6. S0/DS1-0/6 C=102,21,22 I=0x7 P=16600,16572 M=3 S=4,4 C=5 E=0,0,0,0 R=0,0

Total number of active calls 6

The following is sample output from the show mgcp connection command for VoAAL2 connections:

Router# show mgcp connection

Endpoint  Call_ID(C) Conn_ID(I)(V)cci/cid (M)ode (S)tate (C)odec (E)vent[SIFL] (R)esult[EA]

1.aaln/S1/1   C=1,11,12    I=0x2    V=2/10     M=3    S=4,4    C=1    E=3,0,0,3     R=0,0

Total number of active calls 1

Example Output for show xcsp slot Command

The following is sample output from the show xcsp slot command:

Router# show xcsp slot 1

Slot 1 configured

Number of ports configured=1 slot state= Up

Example Output for show xcsp port Command

The following is sample output for the show xcsp port command:

Router# show xcsp port 1 0

Slot 1 configured

Number of ports configured=1 slot state= Up

===================================================

Port 0 State= Up type = 5850 24 port T1

Channel states

0  Idle

1  Idle

2  Idle

3  Idle

4  Idle

5  Idle

6  Idle

7  Idle

8  Idle

9  Idle

10  Idle

11  Idle

12  Idle

13  Idle

14  Idle

15  Idle

16  Idle

17  Idle

18  Idle

19  Idle

20  Idle

21  Idle

22  Idle

23  Idle

Example Output for show cdapi Command

The following is output for the show cdapi command:

Router# show cdapi

Registered CDAPI Applications/Stacks

====================================

Application TSP CDAPI Application

Application Type(s) Voice Facility Signaling

Application Level Tunnel

Application Mode Enbloc

Signaling Stack ISDN

Interface Se023

Signaling Stack ISDN

Interface Se123

Active CDAPI Calls

==================

Interface Se023

No active calls.

Interface Se123

Call ID = 0x39, Call Type = VOICE, Application = TSP CDAPI Application

CDAPI Message Buffers

=====================

Used Msg Buffers 0, Free Msg Buffers 1600

Used Raw Buffers 1, Free Raw Buffers 799

Used Large-Raw Buffers 0, Free Large-Raw Buffers 80

scarlatti1#

MGCP Debugging

To debug MGCP calls, events, and operations, use the following commands in privileged EXEC mode:

Command

Purpose

debug mgcp all

Router# debug mgcp all

Enables all MGCP debugs.

See Example Output for debug mgcp all Command.

debug mgcp events

Router# debug mgcp events

Enables MGCP events debugging, which shows information such as the following: whether the router is detected, the MGCP event that initiates a call, and the reset of an controller that is being serviced by MGCP.

See Example Output for debug mgcp events Command.

debug mgcp packets

Router# debug mgcp packets

Enables debugging of MGCP packets. Useful for displaying contents of NTFY, CRCX, DLCX, and other packets.

See Example Output for debug mgcp packets Command.

debug mgcp parser

Router# debug mgcp parser

Enables debugging of MGCP parser and builder. Useful to determine whether NTFY, CRCX, and other packets have the format that the router expects.

See Example Output for debug mgcp parser Command.

debug mgcp nas

Router# debug mgcp nas

Enables debugging for MGCP data channels and events.

See Example Output for debug mgcp nas Command.

debug xcsp {all | cot | event }

Router# debug xcsp {all | cot | event}

Enables reporting of the exchange of signaling information between the MGCP protocol stack and end applications, such as call switching module (CSM) and dialer.

See Example Output for debug xcsp Command.

debug cdapi {detail | events}

Router# debug cdapi {detail | events}

Displays real-time information about the call distributor application programming interface (CDAPI).

See Example Output for debug cdapi Command.

Example Output for debug mgcp all Command

The debug mgcp all command and keyword would show a compilation of all this output, including the debug mgcp voipcac command and keyword output. Note that using the debug mgcp all command and keyword may severely impact network performance.

Example Output for debug mgcp events Command

The following example illustrates the output from the debug mgcp events command and keyword:

Router# debug mgcp events

Media Gateway Control Protocol events debugging is on

Router#

1w1d: MGC stat - 172.19.184.65, total=44, succ=7, failed=21

1w1d: MGCP msg 1

1w1d: remove_old_under_specified_ack:

1w1d: MGC stat - 172.19.184.65, total=44, succ=8, failed=21

1w1d: updating lport with 2427setup_ipsocket: laddr=172.29.248.193, lport=2427,

faddr=172.19.184.65, fport=2427

1w1d: enqueue_ack: ackqhead=0, ackqtail=0, ackp=1DC1D38, msg=21A037C

Example Output for debug mgcp packets Command

The following example illustrates the output from the debug mgcp packetscommand and keyword:

Router# debug mgcp packets

Media Gateway Control Protocol packets debugging is on

Router#

1w1d: MGCP Packet received -

DLCX 408631346 * MGCP 0.1

1w1d: send_mgcp_msg, MGCP Packet sent --->

1w1d: 250 408631346

<---

Example Output for debug mgcp parser Command

The following example illustrates the output from the debug mgcp parser command and keyword:

Router# debug mgcp parser

Media Gateway Control Protocol parser debugging is on

Router#

1w1d: -- mgcp_parse_packet() - call mgcp_parse_header

- mgcp_parse_header()- Request Verb FOUND DLCX

- mgcp_parse_packet() - out mgcp_parse_header

- SUCCESS: mgcp_parse_packet()- MGCP Header parsing was OK

- mgcp_val_mandatory_parms()

- SUCCESS: mgcp_parse_packet()- END of Parsing

1w1d: -- mgcp_build_packet()-

1w1d: - mgcp_estimate_msg_buf_length() - 87 bytes needed for header

- mgcp_estimate_msg_buf_length() - 87 bytes needed after checking parameter lines

- mgcp_estimate_msg_buf_length() - 87 bytes needed after checking SDP lines

- SUCCESS: MGCP message building OK

- SUCCESS: END of building

Example Output for debug mgcp nas Command

The following example displays output for the debug mgcp nas command and keyword, with the debug mgcp packetscommand and keyword enabled as well:

Router# debug mgcp nas

Media Gateway Control Protocol nas pkg events debugging is on

Router# debug mgcp packets

Media Gateway Control Protocol packets debugging is on

Router#

01:49:14:MGCP Packet received -

CRCX 58 S7/DS1-0/23 MGCP 1.0

X:57

M:nas/data

C:3

L:b:64, nas/bt:modem, nas/cdn:3000, nas/cgn:1000

mgcp_parse_conn_mode :string past nas = data

mgcp_chq_nas_pkg:Full string:nas/bt:modem

mgcp_chq_nas_pkg:string past slash:bt

mgcp_chq_nas_pkg:string past colon:modem

mgcp_chq_nas_pkg:Full string:nas/cdn:3000

mgcp_chq_nas_pkg:string past slash:cdn

mgcp_chq_nas_pkg:string past colon:3000

mgcp_chq_nas_pkg:Full string:nas/cgn:1000

c5400#

mgcp_chq_nas_pkg:string past slash:cgn

mgcp_chq_nas_pkg:string past colon:1000

CHECK DATA CALL for S7/DS1-0/23

mgcpapp_xcsp_get_chan_cb -Found - Channel state  Idle

CRCX Recv

mgcpapp_endpt_is_data:endpt S7/DS1-0/23, slot 7, port 0 chan 23

mgcpapp_data_call_hnd:mgcpapp_xcsp_get_chan_cb -Found - Channel state  Idle

bw=64, bearer=E1,cdn=3000,cgn=1000

Example Output for debug xcsp Command

The following examples show output for the debug xcsp all command and keyword and the debug xcsp event command and keyword:

Router# debug xcsp all

xcsp all debugging is on

Router# debug xcsp event

xcsp events debugging is on

01:49:14:xcsp_call_msg:Event  Call Indication , channel state =  Idle  for slot port channel 7

c5400# 0 23

01:49:14:xcsp_process_sig_fsm:state/event  Idle / Call Indication

01:49:14:xcsp_incall:

01:49:14:xcsp_incall CONNECT_IND:cdn=3000 cgn=1000

01:49:14:xcsp:START guard TIMER

01:49:14:xcsp_fsm:slot 7 port 0 chan 23 oldstate =  Idle  newstate= Connection

in progress mgcpapp_process_mgcp_msg PROCESSED NAS PACKAGE EVENT

01:49:14:Received message on XCSP_CDAPI

01:49:14:process_cdapi_msg :slot/port/channel 7/0/23

01:49:14: process_cdapi_msg:new  slot/port/channel 7/0/23

01:49:14:

c5400#Received CONN_RESP:callid=0x7016

01:49:14:process_cdapi:Event CONN_RESP, channel state = 8 for slot port channel 7 0 23

01:49:14:xcsp_process_sig_fsm:state/event  Connection in progress / In Call accept

mgcpapp_xcsp_alert:

mgcpapp_xcsp_get_chan_cb -Found - Channel state  Connection in progress

200 58  Alert

I:630AED90

<---:Ack send SUCCESSFUL

01:49:14:xcsp_fsm:slot 7 p

c5400#ort 0 chan 23 oldstate =  Connection in progress  newstate= Connection in progress

01:49:14:Received message on XCSP_CDAPI

01:49:14:process_cdapi_msg :slot/port/channel 7/0/23

01:49:14: process_cdapi_msg:new  slot/port/channel 7/0/23

01:49:14: Received CALL_CONN:callid=0x7016

01:49:14:process_cdapi:Event CONN_, channel state = 8 for slot port channel 7

0 23

01:49:14:xcsp_process_sig_fsm:state/event  Connection in progress / in call connect

mgcpapp_xcsp_connect:

mgcpapp_xc

c5400#sp_get_chan_cb -Found - Channel state In Use

01:49:14:STOP TIMER

01:49:14:xcsp_fsm:slot 7 port 0 chan 23 oldstate =  Connection in progress

newstate=In Use

c5400#

01:50:23:Received message on XCSP_CDAPI

01:50:23:process_cdapi_msg :slot/port/channel 7/0/23

01:50:23: process_cdapi_msg:new  slot/port/channel 7/0/23

01:50:23: Received CALL_DISC_REQ:callid=0x7016

01:50:23:process_cdapi:Event DISC_CONN_REQ, channel state = 7 for slot port

channel 7 0 23

01:50:23:xcsp_process_sig_fsm:state/event In Use / release Request

mgcpapp_xcsp_disconnect

mgcpapp_xcsp_get_chan_cb -Fou

c5400#nd - Channel state In Use

01:50:23:send_mgcp_msg, MGCP Packet sent --->

01:50:23:RSIP 1 *@c5400 MGCP 1.0

RM:restart

.

DLCX 4 S7/DS1-0/23 MGCP 1.0

C:3

I:630AED90

E:801 /NAS User request

<---

01:50:23:xcsp_fsm:slot 7 port 0 chan 23 oldstate = In Use  newstate=Out

Release in progress

xcsp_restart Serial7/0:22 vc = 22

xcsp_restart Put idb Serial7/0:22 in down state

01:50:23:MGCP Packet received -

200 4 bye

Data call ack received callp=0x62AEEA70mgcpapp_xcsp

c5400#_ack_recv:mgcpapp_xcsp_get_chan_cb -Found - Channel state Out Release in progress

mgcpapp_xcsp_ack_recv ACK 200 rcvd:transaction id = 4 endpt=S7/DS1-0/23

01:50:23:xcsp_call_msg:Event  Release confirm , channel state = Out Release in progress for slot port channel 7 0 23

01:50:23:xcsp_process_sig_fsm:state/event Out Release in progress/ Release confirm

01:50:23:STOP TIMER

01:50:23:xcsp_fsm:slot 7 port 0 chan 23 oldstate = Out Release in progress

newstate= Idle

Example Output for debug cdapi Command

The following example shows output for the debug cdapi command:

003909 ISDN Se123 RX <- SETUP pd = 8 callref = 0x06BB

003909 Bearer Capability i = 0x9090A2

003909 Channel ID i = 0xA18381

003909 Facility i = 0x9FAA068001008201008B0100A1180202274C020100800F534341524C415454492D3530303733

003909 Progress Ind i = 0x8183 - Origination address is non-ISDN

003909 Calling Party Number i = 0xA1, '50073'

003909 Called Party Number i = 0xC1, '3450070'

003909 CDAPI Se123 TX -> CDAPI_MSG_CONNECT_IND to TSP CDAPI Application call = 0x24

003909 From Appl/Stack = ISDN

003909 Call Type = VOICE

003909 B Channel = 0

003909 Cause = 0

003909 Calling Party Number = 50073

003909 Called Party Number = 3450070

003909 CDAPI Se123 TX -> CDAPI_MSG_CONNECT_RESP to ISDN call = 0x24

003909 From Appl/Stack = TSP CDAPI Application

003909 Call Type = VOICE

003909 B Channel = 0

003909 Cause = 0

003909 CDAPI-ISDN Se123 RX <- CDAPI_MSG_CONNECT_RESP from TSP CDAPI Application call = 0x24

003909 Call Type = VOICE

003909 B Channel = 0

003909 Cause = 0

003909 CDAPI Se123 TX -> CDAPI_MSG_SUBTYPE_CALL_PROC_REQ to ISDN call = 0x24

003909 From Appl/Stack = TSP CDAPI Application

003909 Call Type = VOICE

003909 B Channel = 0

003909 Cause = 0

003909 CDAPI-ISDN Se123 RX <- CDAPI_MSG_SUBTYPE_CALL_PROC_REQ from TSP CDAPI Application call = 0x24

003909 Call Type = VOICE

003909 B Channel = 0

003909 Cause = 0

003909 ISDN Se123 TX -> CALL_PROC pd = 8 callref = 0x86BB

003909 Channel ID i = 0xA98381

Controller Troubleshooting

The commands in this section can be helpful in finding sources of problems with call connections and switching. The call switching module (CSM) associated with a controller contains digit collection logic that processes incoming calls for automatic number information (ANI) and dialed number identification service (DNIS) digits.

To display information on controller and CSM configuration and operation, use the following commands in privileged EXEC mode.

Command

Purpose

show controllers t1 | e1 [slot / port]

Router# show controllers t1 | e1 [slot/port

]

Displays whether the T1 or E1 connection between the router and switch (central office [CO] or PBX) is up or down and whether the connection is functioning properly.

See Example Output for show controllers e1 or t1 Command.

show voice port [slot / port]

Router# show voice port [slot

/port

]

Displays the port state and the parameters configured on the voice ports of Cisco voice interface cards. Voice-port defaults, like all command-line interface default parameters, do not display in the output for the show running-configcommand, but they can be seen with the show voice port command.

See Example Output for show voice port Command.

show csm modem [slot/port | modem-group-number]

Router# show csm modem [slot

/port

| modem-group-number

]

Displays the CSM call statistics for a specific modem, for a group of modems, or for all modems.

debug csm modem [slot/port | group modem-group-number]

Router# debug csm modem [slot

/port

| group modem-group-number

]

Traces the complete sequence of switching of incoming and outgoing modem call.

Example Output for show controllers e1 or t1 Command

The following is an output example from the show controllers e1 command on the Cisco 7500 series:

Router# show controllers e1

e1 0/0 is up.

Applique type is Channelized E1 - unbalanced

Framing is CRC4, Line Code is HDB3

No alarms detected.

Data in current interval (725 seconds elapsed):

0 Line Code Violations, 0 Path Code Violations

0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs

Total Data (last 24 hours)

0 Line Code Violations, 0 Path Code Violations,

0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,

0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs

The following is an example of the show controllers e1 display including the board identifier type:

Router# show controllers e1

E1 4/1 is up.

No alarms detected.

Framing is CRC4, Line Code is hdb3

Data in current interval (0 seconds elapsed):

0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs,

0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs,

0 Severely Err Secs, 0 Unavail Secs

Total Data (last 79 15 minute intervals):

0 Line Code Violations, 0 Path Code Violations, 0 Slip Secs, 0 Fr Loss Secs,

0 Line Err Secs, 0 Degraded Mins, 0 Errored Secs, 0 Bursty Err Secs,

0 Severely Err Secs, 0 Unavail Secs

The following is an example from the show controllers t1 command on the Cisco 7500 series routers:

Router# show controllers t1

T1 4/1 is up.

No alarms detected.

Framing is ESF, Line Code is AMI, Clock Source is line

Data in current interval (0 seconds elapsed):

0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs,

0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs,

0 Severely Err Secs, 0 Unavail Secs

Total Data (last 79 15 minute intervals):

0 Line Code Violations, 0 Path Code Violations, 0 Slip Secs, 0 Fr Loss Secs,

0 Line Err Secs, 0 Degraded Mins, 0 Errored Secs, 0 Bursty Err Secs,

0 Severely Err Secs, 0 Unavail Secs

The following example shows the status of the T1 controllers connected to the Cisco AS5800 access servers:

Router# show controller T1

T1 1/0/0:1 is up.

No alarms detected.

Framing is ESF, Line Code is AMI, Clock Source is Line.

Data in current interval (770 seconds elapsed):

5 Line Code Violations, 8 Path Code Violations

0 Slip Secs, 0 Fr Loss Secs, 7 Line Err Secs, 0 Degraded Mins

0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 7 Unavail Secs

Total Data (last 81 15 minute intervals):

7 Line Code Violations, 4 Path Code Violations,

6 Slip Secs, 20 Fr Loss Secs, 2 Line Err Secs, 0 Degraded Mins,

0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 2 Unavail Secs

T1 1/0/1:5 is down.

Transmitter is sending remote alarm.

Receiver has loss of frame.

Framing is SF, Line Code is AMI, Clock Source is Line.

Data in current interval (770 seconds elapsed):

50 Line Code Violations, 5 Path Code Violations

0 Slip Secs, 7 Fr Loss Secs, 7 Line Err Secs, 0 Degraded Mins

0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 7 Unavail Secs

Total Data (last 81 15 minute intervals):

27 Line Code Violations, 22 Path Code Violations,

0 Slip Secs, 13 Fr Loss Secs, 13 Line Err Secs, 0 Degraded Mins,

0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 13 Unavail Secs

Router#

Example Output for show voice port Command

The following is sample output from the Cisco AS5800 for the show voice port command:

ISDN 1/0/0:D

Type of VoicePort is ISDN

Operation State is DORMANT

Administrative State is UP

No Interface Down Failure

Description is ""

Noise Regeneration is enabled

Non Linear Processing is enabled

Music On Hold Threshold is Set to -38 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 0 dB

Echo Cancellation is enabled

Echo Cancel Coverage is set to 16 ms

Connection Mode is normal

Connection Number is not set

Initial Time Out is set to 10 s

Interdigit Time Out is set to 10 s

Region Tone is set for US

The following example displays voice port configuration information for the digital voice port 0 located in slot 1, DS0 group 1:

receEive and transMit Slot is 1, Sub-unit is 0, Port is 1

Type of VoicePort is E&M

Operation State is DORMANT

Administrative State is UP

No Interface Down Failure

Description is not set

Noise Regeneration is enabled

Non Linear Processing is enabled

Music On Hold Threshold is Set to -38 DBMS

In Gain is Set to 0 dBm

Out Attenuation is Set to 0 dB

Echo Cancellation is enabled

Echo Cancel Coverage is set to 8 ms

Connection Mode is normal

Connection Number is not set

Initial Time Out is set to 10 s

Interdigit Time Out is set to 10 s

Region Tone is set for US

The following is sample output from the show voice port command for an E&M digital voice port on a Cisco 3600 series:

receEive and transMit Slot is 1, Sub-unit is 0, Port is 1

Type of VoicePort is E&M

Operation State is DORMANT

Administrative State is UP

No Interface Down Failure

Description is not set

Noise Regeneration is enabled

Non Linear Processing is enabled

Music On Hold Threshold is Set to -38 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 0 dB

Echo Cancellation is enabled

Echo Cancel Coverage is set to 8 ms

Connection Mode is normal

Connection Number is not set

Initial Time Out is set to 10 s

Interdigit Time Out is set to 10 s

Region Tone is set for US

The following is sample output from the show voice port command for an FXS analog voice port on a Cisco MC3810 multiservice concentrator:

Voice port 1/2 Slot is 1, Port is 2

Type of VoicePort is FXS

Operation State is UP

Administrative State is UP

No Interface Down Failure

Description is not set

Noise Regeneration is enabled

Non Linear Processing is enabled

In Gain is Set to 0 dB

Out Attenuation is Set to 0 dB

Echo Cancellation is enabled

Echo Cancel Coverage is set to 8 ms

Connection Mode is normal

Connection Number is not set

Initial Time Out is set to 10 s

Interdigit Time Out is set to 10 s

Coder Type is g729ar8

Companding Type is u-law

Voice Activity Detection is disabled

Ringing Time Out is 180 s

Wait Release Time Out is 30 s

Nominal Playout Delay is 80 milliseconds

Maximum Playout Delay is 160 milliseconds

Analog Info Follows:

Region Tone is set for northamerica

Currently processing Voice

Maintenance Mode Set to None (not in mtc mode)

Number of signaling protocol errors are 0

Impedance is set to 600r Ohm

Analog interface A-D gain offset = -3 dB

Analog interface D-A gain offset = -3 dB

Voice card specific Info Follows:

Signal Type is loopStart

Ring Frequency is 20 Hz

Hook Status is On Hook

Ring Active Status is inactive

Ring Ground Status is inactive

Tip Ground Status is active

Digit Duration Timing is set to 100 ms

InterDigit Duration Timing is set to 100 ms

Ring Cadence are [20 40] * 100 msec

InterDigit Pulse Duration Timing is set to 500 ms

The following is sample output from the show voice port command for a Foreign Exchange Station (FXS) analog voice port on a Cisco 3600 series:

Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0

Type of VoicePort is FXS

Operation State is DORMANT

Administrative State is UP

The Interface Down Failure Cause is 0

Alias is NULL

Noise Regeneration is enabled

Non Linear Processing is enabled

Music On Hold Threshold is Set to 0 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 0 dB

Echo Cancellation is enabled

Echo Cancel Coverage is set to 16ms

Connection Mode is Normal

Connection Number is

Initial Time Out is set to 10 s

Interdigit Time Out is set to 10 s

Analog Info Follows:

Region Tone is set for northamerica

Currently processing none

Maintenance Mode Set to None (not in mtc mode)

Number of signaling protocol errors are 0

Voice card specific Info Follows:

Signal Type is loopStart

Ring Frequency is 25 Hz

Hook Status is On Hook

Ring Active Status is inactive

Ring Ground Status is inactive

Tip Ground Status is inactive

Digit Duration Timing is set to 100 ms

InterDigit Duration Timing is set to 100 ms

Hook Flash Duration Timing is set to 600 ms

The following is sample output from the show voice port command for an E&M analog voice port on a Cisco 3600 series:

E&M Slot is 1, Sub-unit is 0, Port is 0

Type of VoicePort is E&M

Operation State is unknown

Administrative State is unknown

The Interface Down Failure Cause is 0

Alias is NULL

Noise Regeneration is disabled

Non Linear Processing is disabled

Music On Hold Threshold is Set to 0 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 0 dB

Echo Cancellation is disabled

Echo Cancel Coverage is set to 16ms

Connection Mode is Normal

Connection Number is

Initial Time Out is set to 0 s

Interdigit Time Out is set to 0 s

Analog Info Follows:

Region Tone is set for northamerica

Currently processing none

Maintenance Mode Set to None (not in mtc mode)

Number of signaling protocol errors are 0

Voice card specific Info Follows:

Signal Type is wink-start

Operation Type is 2-wire

Impedance is set to 600r Ohm

E&M Type is unknown

Dial Type is dtmf

In Seizure is inactive

Out Seizure is inactive

Digit Duration Timing is set to 0 ms

InterDigit Duration Timing is set to 0 ms

Pulse Rate Timing is set to 0 pulses/second

InterDigit Pulse Duration Timing is set to 0 ms

Clear Wait Duration Timing is set to 0 ms

Wink Wait Duration Timing is set to 0 ms

Wink Duration Timing is set to 0 ms

Delay Start Timing is set to 0 ms

Delay Duration Timing is set to 0 ms

Dialer Interface and Routing Troubleshooting

To obtain information on dialer interfaces, routing configuration, and routing operations, use the following commands in privileged EXEC mode.

Command

Purpose

show dialer map

Router# show dialer map

Displays configured dynamic and static dialer maps.

See Example Output for show dialer map Command.

show dialer

Router# show dialer

Displays general diagnostic information about an interface configured for DDR, such as the number of times the dialer string has been successfully reached, and the idle timer and the fast idle timer values for each B channel. Current call-specific information is also provided, such as the length of a call and the number and name of the device to which the interface is currently connected. When external signaling is configured, the output also displays the CDAPI state.

See Example Output for show dialer Command.

show interface Dialer-num

Router# show interface Dialer-num

Shows whether the interface and protocol are up (spoofing), a state in which the dialer interface pretends to be up/up so that associated routes remain in force and packets can be routed to the interface.

See Example Output for show interface Command.

show ip route

Router# show ip route

Displays the routes known to the router, including static and dynamically learned routes.

See Example Output for show ip route Command.

Example Output for show dialer map Command

The following is sample output from the show dialer map command.

Router# show dialer map

Static dialer map ip 10.1.1.1 name peer_1 on Dialer1

Static dialer map ip 10.1.1.2 name peer_2 on Dialer1

BAP dialer map ip 10.1.1.2 name peer_2 on Dialer1

Dynamic dialer map ip 10.1.1.3 name peer_3 on Dialer1

BAP dialer map ip 10.1.1.3 name peer_3 on Dialer1

Example Output for show dialer Command

The following is sample output from the show dialer command for a BRI interface when dialer profiles are configured:

Router# show dialer interface bri 0

BRI0 - dialer type = ISDN

Dial String Successes Failures Last called Last status

0 incoming call(s) have been screened.

BRI0: B-Channel 1

Idle timer (120 secs), Fast idle timer (20 secs)

Wait for carrier (30 secs), Re-enable (15 secs)

Dialer state is data link layer up

Dial reason: ip (s=10.1.1.8, d=10.1.1.1)

Interface bound to profile Dialer0

Time until disconnect 102 secs

Current call connected 00:00:19

Connected to 5773872 (wolfman)

BRI0: B-Channel 2

Idle timer (120 secs), Fast idle timer (20 secs)

Wait for carrier (30 secs), Re-enable (15 secs)

Dialer state is idle

The following is sample output from the show dialer command for a dialer under external signaling control:

Router# show dialer

Se7/0:0 - dialer type = IN-BAND SYNC NO-PARITY

Rotary group 1, priority 0

Idle timer (222222 secs), Fast idle timer (20 secs)

Wait for carrier (30 secs), Re-enable (15 secs)

Dialer state is idle

Dialer cdapi state is idle <<<<<<<<<==================

Se7/0:1 - dialer type = IN-BAND SYNC NO-PARITY

Rotary group 1, priority 0

Idle timer (222222 secs), Fast idle timer (20 secs)

Wait for carrier (30 secs), Re-enable (15 secs)

Dialer state is idle

Dialer cdapi state is idle <<<<<<<<<<=================

The following is sample output from the show dialer command for an asynchronous interface:

Router# show dialer interface async 1

Async1 - dialer type = IN-BAND NO-PARITY

Idle timer (900 secs), Fast idle timer (20 secs)

Wait for carrier (30 secs), Re-enable (15 secs)

Time until disconnect 838 secs

Current call connected 0:02:16

Connected to 8986

Dial String Successes Failures Last called Last status

8986 0 0 never Defaults

8986 8 3 0:02:16 Success Defaults

When the show dialer EXEC command is issued for a synchronous serial interface configured for DTR dialing, output similar to the following is displayed:

Serial 0 - dialer type = DTR SYNC

Idle timer (120 secs), Fst idle timer (20 secs)

Wait for carrier (30 secs), Re-enable (15 secs)

Dial String Successes Failures Last called Last status

---- 1 0 1:04:47 Success DTR dialer

8986 0 0 never Defaults

Example Output for show interface Command

The following is sample output from the show interface Dialer0command:

Router# show interface Dialer0

Dialer0 is up (spoofing), line protocol is up (spoofing)

Hardware is Unknown

Internet address is 60.0.0.2/24

MTU 1500 bytes, BW 56 Kbit, DLY 20000 usec,

reliability 255/255, txload 1/255, rxload 1/255

Encapsulation PPP, loopback not set

DTR is pulsed for 1 seconds on reset

Last input never, output never, output hang never

Last clearing of "show interface" counters 1d17h

Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0

Queueing strategy: weighted fair

Output queue: 0/1000/64/0 (size/max total/threshold/drops)

Conversations  0/0/16 (active/max active/max total)

Reserved Conversations 0/0 (allocated/max allocated)

Available Bandwidth 42 kilobits/sec

5 minute input rate 0 bits/sec, 0 packets/sec

5 minute output rate 0 bits/sec, 0 packets/sec

0 packets input, 0 bytes

0 packets output, 0 bytes

Example Output for show ip route Command

The following examples display all downloaded static routes. A P designates which route was installed using AAA route download.

Router# show ip route

Codes: C - connected, S - static, I - IGRP, R - RIP, M - mobile, B - BGP

D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter area

N1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2

E1 - OSPF external type 1, E2 - OSPF external type 2, E - EGP

i - IS-IS, L1 - IS-IS level-1, L2 - IS-IS level-2, * - candidate default

U - per-user static route, o - ODR, P - periodic downloaded static route

T - traffic engineered route

Gateway of last resort is 172.21.17.1 to network 0.0.0.0

172.31.0.0/32 is subnetted, 1 subnets

P 172.31.229.41 is directly connected, Dialer1 20.0.0.0/24 is subnetted, 3 subnets

P 10.1.1.0 [200/0] via 172.31.229.41, Dialer1

P 10.1.3.0 [200/0] via 172.31.229.41, Dialer1

P 10.1.2.0 [200/0] via 172.31.229.41, Dialer1

Router# show ip route static

172.27.4.0/8 is variably subnetted, 2 subnets, 2 masks

P 172.1.1.1/32 is directly connected, BRI0

P 172.27.4.0/8 [1/0] via 103.1.1.1, BRI0

S 172.31.0.0/16 [1/0] via 172.21.114.65, Ethernet0

S 10.0.0.0/8 is directly connected, BRI0

P 10.0.0.0/8 is directly connected, BRI0

172.21.0.0/16 is variably subnetted, 5 subnets, 2 masks

S 172.21.114.201/32 is directly connected, BRI0

S 172.21.114.205/32 is directly connected, BRI0

S 172.21.114.174/32 is directly connected, BRI0

S 172.21.114.12/32 is directly connected, BRI0

P 10.0.0.0/8 is directly connected, BRI0

P 10.1.0.0/8 is directly connected, BRI0

P 10.2.2.0/8 is directly connected, BRI0

S* 0.0.0.0/0 [1/0] via 172.21.114.65, Ethernet0

S 172.29.0.0/16 [1/0] via 172.21.114.65, Ethernet0

To debug dialer and authorization or to clear in-progress calls, use the following commands in privileged EXEC mode.

Command

Purpose

debug dialer

Router# debug dialer

Displays the activity that triggers a dial attempt.

See Example Output for show dialer Command.

clear interface

Router# clear interface

Clears a call that is in progress. In a troubleshooting situation, it is sometimes useful to clear historical statistics to track the current number of successful calls relative to failures. Use this command with care. It sometimes requires that you clear both the local and remote routers.

See Example Output for clear interface Command.

debug ppp negotiation

Router# debug ppp negotiation

Displays negotiation of PPP options and Network Control Protocol (NCP) parameters.

See Example Output for debug ppp negotiation Command.

debug ppp authentication

Router# debug ppp authentication

Displays exchange of Challenge Handshake Authentication Protocol (CHAP) and Password Authentication Protocol (PAP) packets.

See Example Output for debug ppp authentication Command.

Example Output for debug dialer Command

Displays the activity that triggers a dial attempt.

Dialing cause: Async1: ip (s=172.16.1.111 d=172.16.2.22)

Example Output for clear interface Command

The following example demonstrates the use of the clear interface command with the RLM feature:

Router# clear interface loopback 1

02:48:52: rlm 1: [State_Up, rx ACTIVE_LINK_BROKEN] over link [10.1.1.1(Loopback1), 10.1.4.1]

02:48:52: rlm 1: link [10.1.1.2(Loopback2), 10.1.4.2] requests activation

02:48:52: rlm 1: link [10.1.1.1(Loopback1), 10.1.4.1] is deactivated

02:48:52: rlm 1: link [10.1.1.1(Loopback1), 10.1.4.1] = socket[10.1.1.1, 10.1.4.1]

02:48:52: rlm 1: [State_Recover, rx USER_SOCKET_OPENED] over link [10.1.1.1(Loopback1), 10.1.4.1] for user RLM_MGR

02:48:52: rlm 1: link [10.1.1.1(Loopback1), 10.1.4.1] is opened

02:48:52: rlm 1: link [10.1.1.1(Loopback1), 10.1.5.1] = socket[10.1.1.1, 10.1.5.1]

02:48:52: rlm 1: [State_Recover, rx USER_SOCKET_OPENED] over link [10.1.1.1(Loopback1), 10.1.5.1] for user RLM_MGR

02:48:52: rlm 1: link [10.1.1.1(Loopback1), 10.1.5.1] is opened

02:48:52: rlm 1: [State_Recover, rx START_ACK] over link [10.1.1.2(Loopback2), 10.1.4.2]

02:48:52: rlm 1: link [10.1.1.2(Loopback2), 10.1.4.2] is activated

02:48:52: rlm 1: [State_Up, rx LINK_OPENED] over link [10.1.1.1(Loopback1), 10.1.4.1]

Example Output for debug ppp negotiation Command

The following is sample output from the debug ppp negotiation command. This is a normal negotiation, where both sides agree on Network Control Program (NCP) parameters. In this case, protocol type IP is proposed and acknowledged.

Router# debug ppp negotiation

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 3D56CAC

ppp: received config for type = 4 (QUALITYTYPE) acked

ppp: received config for type = 5 (MAGICNUMBER) value = 3D567F8 acked (ok)

PPP Serial4: state = ACKSENT fsm_rconfack(C021): rcvd id 5

ppp: config ACK received, type = 4 (CI_QUALITYTYPE), value = C025

ppp: config ACK received, type = 5 (CI_MAGICNUMBER), value = 3D56CAC

ppp: ipcp_reqci: returning CONFACK.

(ok)

PPP Serial4: state = ACKSENT fsm_rconfack(8021): rcvd id 4

The following is sample output from the debug ppp negotiation command when the remote side of the connection is unable to respond to LQM requests:

Router# debug ppp negotiation

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44B7010

ppp: sending CONFREQ, type = 4 (CI_QUALITYTYPE), value = C025/3E8

ppp: sending CONFREQ, type = 5 (CI_MAGICNUMBER), value = 44C1488

Example Output for debug ppp authentication Command

The following is sample output from the debug ppp authentication command. Use this debug command to determine why an authentication fails.

Router# debug ppp authentication

Serial0: Unable to authenticate. No name received from peer

Serial0: Unable to validate CHAP response. USERNAME pioneer not found.

Serial0: Unable to validate CHAP response. No password defined for USERNAME pioneer

Serial0: Failed CHAP authentication with remote.

Remote message is Unknown name

Serial0: remote passed CHAP authentication.

Serial0: Passed CHAP authentication with remote.

Serial0: CHAP input code = 4 id = 3 len = 48

Configuration Examples for NAC Package for MGCP

- NAS Package for MGCP Example

NAS Package for MGCP Example

This example configures the Network Access Server Package for Media Gateway Control Protocol Feature on a Cisco AS5400:

version 12.2

no service single-slot-reload-enable

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname 54iwo

!

no boot startup-test

logging rate-limit console 10 except errors

!

resource-pool disable

!

resource-pool profile service user1sample

!

voice-fastpath enable

ip subnet-zero

ip host 54ccxv 172.18.16.25

!

no ip dhcp-client network-discovery

isdn switch-type primary-ni

!

fax interface-type modem

mta receive maximum-recipients 0

!

controller T1 7/0

framing esf

extsig mgcp

guard-timer 10 on-expiry reject

linecode b8zs

ds0-group 1 timeslots 1-24 type none service mgcp

!

controller T1 7/1

framing esf

linecode ami

pri-group timeslots 1-24

!

controller T1 7/2

framing sf

linecode ami

!

controller T1 7/3

framing sf

linecode ami

!

controller T1 7/4

framing sf

linecode ami

!

controller T1 7/5

framing sf

linecode ami

!

controller T1 7/6

framing sf

linecode ami

!

controller T1 7/7

framing sf

linecode ami

!

interface Loopback0

ip address 172.16.0.3 255.255.255.0

!

interface FastEthernet0/0

ip address 172.18.184.183 255.255.255.0

duplex auto

speed auto

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface Serial0/0

no ip address

shutdown

clockrate 2000000

!

interface Serial0/1

no ip address

shutdown

clockrate 2000000

!

interface Serial7/1:23

no ip address

encapsulation ppp

dialer rotary-group 9

dialer-group 1

isdn switch-type primary-ni

isdn incoming-voice modem

no cdp enable

!

interface Async1/00

ip unnumbered Loopback0

dialer in-band

dialer map ip 172.23.0.1 234567

dialer-group 1

!

interface Async1/01

ip address 10.17.1.1 255.255.255.0

encapsulation ppp

dialer in-band

dialer map ip 10.17.1.2 22222

dialer-group 1

!

interface Async1/02

no ip address

!

interface Async1/03

no ip address

!

interface Async1/04

no ip address

!

interface Async1/05

no ip address

!

interface Async3/102

no ip address

!

interface Async3/103

no ip address

!

interface Async3/104

no ip address

!

interface Async3/105

no ip address

!

interface Async3/106

no ip address

!

interface Async3/107

no ip address

!

interface Group-Async0

no ip address

no group-range

!

interface Dialer1

ip unnumbered Loopback0

encapsulation ppp

dialer in-band

dialer idle-timeout 222222

dialer map ip 172.16.0.1 name 53bxbv 1000

dialer extsig

dialer-group 1

no cdp enable

ppp authentication chap

ppp direction dedicated

!

interface Dialer9

ip address 10.1.1.1 255.255.255.0

encapsulation ppp

dialer in-band

dialer map ip 10.1.1.2 23456

dialer-group 1

no cdp enable

!

ip classless

ip route 0.0.0.0 0.0.0.0 172.18.184.1

ip route 172.16.0.1 255.255.255.255 Dialer1

ip route 172.23.0.1 255.255.255.255 Async1/00

no ip http server

!

dialer-list 1 protocol ip permit

!

call rsvp-sync

!

voice-port 7/0:1

!

voice-port 7/1:D

!

mgcp

mgcp call-agent 172.18.64.242 service-type mgcp version 1.0

no mgcp timer receive-rtcp

!

mgcp profile default

max2 retries 5

!

line con 0

exec-timeout 0 0

logging synchronous

line aux 0

logging synchronous

line vty 0 4

password mango

login

line 1/00 1/107

no flush-at-activation

modem InOut

line 3/00 3/107

no flush-at-activation

modem InOut

!

scheduler allocate 10000 400

end

Note

See the "Additional References for MGCP and SGCP" section on page x for related documents, standards, and MIBs and see the " Glossary " for definitions of terms in this guide.

Configuring SGCP RSIP and AUEP Enhancements

This section provides information on configuring the Simple Gateway Control Protocol (SGCP) Restart In Progress (RSIP) and Audit Endpoint (AUEP) Enhancements feature. The feature provides enhancements to SGCP for disconnected RSIP and audit endpoints requested by call agents.

Feature benefits include the following:

- Provides SGCP 1.5 gateways with the ability to synchronize endpoints with call agents after the disconnected procedure has occurred.

For more information about this and related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Feature History for SGCP RSIP and AUEP Enhancements

Release

Modification

12.2(11)T

This feature was introduced on the following platforms: Cisco IAD2420 series, Cisco 2600 series, and Cisco MC3810.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for SGCP RSIP and AUEP Enhancements

- Configure SGCP 1.5 on the gateway.

Restrictions for SGCP RSIP and AUEP Enhancements

- This feature applies only to SGCP 1.5 gateways.

- This feature does not apply to MGCP gateways.

Information About SGCP RSIP and AUEP Enhancements

The SGCP RSIP and AUEP Enhancements feature provides additional messaging capabilities that allow an endpoint on a Simple Gateway Control Protocol (SGCP) 1.5 gateway to synchronize with a call agent after the endpoint returns to service from the disconnected procedure. The additional messaging capabilities provide the following:

- A special disconnected Restart In Progress (RSIP) message that the gateway sends to the call agent as a result of the disconnected procedure.

- Additional fields in the Audit Endpoint (AUEP) command that the call agent uses to query the endpoint's status when contact is reestablished.

Media Gateway Control Protocol (MGCP) provides this ability automatically, but it must be explicitly configured for SGCP networks, as described in the How to Configure SGCP RSIP and AUEP Enhancements.

An endpoint may lose contact with its call agent because the call agent is temporarily off line or because of faults in the network. When a gateway recognizes that an endpoint has lost its communication with the call agent, it initiates the disconnected procedure. The disconnected procedure requires the endpoint to send RSIPs to the call agent and also to guarantee that the first message that the call agent sees from the endpoint is an RSIP command. The endpoint continues to attempt to send RSIPs at the intervals prescribed by the disconnected procedure until an attempt is successful. The RSIP identifies itself as an RSIP that was generated from a disconnected procedure rather than from a restart. The following output is seen on the gateway:

Disconnected RSIP sent from gateway

00:04:27:RSIP 7 ds1-3/2@RouterA SGCP 1.5

RM:disconnected

On receipt of a disconnected RSIP message, the call agent may decide to send an AUEP command to query the status of endpoints and synchronize endpoints. The SGCP RSIP and AUEP Enhancements feature provides the following additional fields of information in the AUEP:

- I--List of connection identifiers for current connections on the endpoint

- ES--Event state of the endpoint (off-hook or on-hook)

- RM--Restart method for the endpoint, which is one of the following:

- Graceful--Endpoints are being taken out of service after a delay; the call agent should not make new connections.

- Forced--Endpoints were abruptly taken out of service; connections were lost.

- Restart--Endpoints with no connections will be returned to service after a delay.

- Disconnected--Endpoints are being returned to service after the disconnected procedure.

How to Configure SGCP RSIP and AUEP Enhancements

- Configuring SGCP RSIP and AUEP Enhancements

- Verifying SGCP RSIP Configuration

Configuring SGCP RSIP and AUEP Enhancements

To configure enhanced restart and endpoint audit messaging capabilities on an SGCP gateway, use the following command in global configuration mode:

Command

Purpose

Router(config)# mgcp sgcp disconnect notify

Enables enhanced endpoint synchronization with a call agent after a disconnected procedure. The command is disabled by default.

Verifying SGCP RSIP Configuration

To verify your configuration, enter the show mgcp command. The following example shows that disconnected RSIP is enabled.

Router# show mgcp

MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE

MGCP call-agent:172.16.193.148 Initial protocol service is SGCP 1.5

MGCP block-newcalls DISABLED

MGCP send SGCP RSIP:forced/restart/graceful DISABLED, disconnected ENABLED

MGCP quarantine mode discard/step

MGCP quarantine of persistent events is ENABLED

MGCP dtmf-relay for VoIP disabled for all codec types

MGCP dtmf-relay for VoAAL2 disabled for all codec types

MGCP voip modem passthrough mode:NSE, codec:g711ulaw, redundancy:DISABLED,

MGCP voaal2 modem passthrough mode:NSE, codec:g711ulaw

MGCP TSE payload:0

MGCP Named Signalling Event (NSE) response timer:200

MGCP Network (IP/AAL2) Continuity Test timer:200

MGCP 'RTP stream loss' timer:5

MGCP request timeout 500

MGCP maximum exponential request timeout 4000

MGCP gateway port:2427, MGCP maximum waiting delay 3000

MGCP restart delay 0, MGCP vad DISABLED

MGCP rtrcac DISABLED

MGCP system resource check DISABLED

MGCP xpc-codec:DISABLED, MGCP persistent hookflash:DISABLED

MGCP persistent offhook:ENABLED, MGCP persistent onhook:DISABLED

MGCP piggyback msg ENABLED, MGCP endpoint offset DISABLED

MGCP simple-sdp DISABLED

MGCP undotted-notation DISABLED

MGCP codec type g711ulaw, MGCP packetization period 20

MGCP JB threshold lwm 30, MGCP JB threshold hwm 150

MGCP LAT threshold lmw 150, MGCP LAT threshold hwm 300

MGCP PL threshold lwm 1000, MGCP PL threshold hwm 10000

MGCP CL threshold lwm 1000, MGCP CL threshold hwm 10000

MGCP playout mode is adaptive 60, 4, 200 in msec

MGCP IP ToS low delay disabled, MGCP IP ToS high throughput disabled

MGCP IP ToS high reliability disabled, MGCP IP ToS low cost disabled

MGCP IP RTP precedence 5, MGCP signaling precedence:3

MGCP default package:line-package

MGCP supported packages:gm-package dtmf-package trunk-package line-package

hs-package atm-package ms-package dt-package res-package

mt-package

MGCP Digit Map matching order:shortest match

SGCP Digit Map matching order:always left-to-right

MGCP VoAAL2 ignore-lco-codec DISABLED

Configuration Examples for SGCP RSIP and AUEP Enhancements

- Disconnected RSIP Messaging Example

Disconnected RSIP Messaging Example

The following example shows the configuration of disconnected RSIP messaging on a Cisco MC3810.

version 12.2

no parser cache

no service pad

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname Router2

!

boot system tftp smithj/mc3810-r3jjl 172.16.206.10

logging buffered 2000000 debugging

no logging console

enable password lab

!

network-clock base-rate 56k

ip subnet-zero

!

no ip domain-lookup

ip host corona 172.16.206.10

ip host redlands 172.31.140.33

ip host rialto 172.16.193.147

!

voice service voip

fax protocol t38 ls-redundancy 0 hs-redundancy 0

!

no voice confirmation-tone

voice-card 0

!

controller T1 0

mode cas

framing esf

clock source internal

linecode ami

ds0-group 0 timeslots 1-24 type fxs-ground-start

!

interface Ethernet0

ip address 172.16.193.162 255.255.255.0

no ip mroute-cache

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface FR-ATM20

no ip address

shutdown

ip classless

ip route 10.0.0.0 10.0.0.0 172.16.193.1

ip route 172.16.0.0 255.255.0.0 172.16.193.1

no ip http server

!

!

call rsvp-sync

!

voice-port 0:0

!

voice-port 1/1

!

voice-port 1/2

description package

!

mgcp

mgcp call-agent 172.16.193.148 service-type sgcp version 1.5

mgcp sgcp disconnect notify

!

mgcp profile default

!

dial-peer cor custom

!

dial-peer voice 1 pots

application mgcpapp

port 1/1

!

dial-peer voice 2 pots

application mgcpapp

port 1/2

!

dial-peer voice 3 pots

application mgcpapp

port 0:0

!

gatekeeper

shutdown

!

line con 0

exec-timeout 0 0

line aux 0

line 2 3

line vty 0 4

exec-timeout 0 0

password hemet

login

!

end

Configuring MGCP Gateway Support

This section provides information on configuring the MGCP Gateway Support for the mgcp bind command feature.

Feature benefits include the following:

- Media gateway controller-to-media gateway ( MGC-to-MG) signaling and identification

The command allows you to use a loopback interface IP address for sourcing MGCP packets, which is transparent to any interface failure.

- Security of the media gateway

The command allows you to obtain a predefined interface for both MGCP and media control, which can be used for security configuration.

- Possible clash of voice and dial addressing

This feature allows you to assign a media bind interface other than loopback 0, which allows dial calls to conserve IP addresses.

- No interface diversity using routing and reduced MGCP voice diversity

You can use routing capability more efficiently if you configure the loopback interface for control. Using the command to configure the loopback interface helps in creating redundant MGCP control or media interface.

- MGCP backward compatibility

This feature is backward compatible with earlier MGCP features.

For more information about this and related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Feature History for MGCP Gateway Support for the mgcp bind Command

Release

Modification

12.2(13)T

This feature was introduced.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for Configuring MGCP Gateway Support

The current Media Gateway Control Protocol (MGCP) implementation does not allow the assignment of particular IP addresses for sourcing MGCP commands and media packets, which can cause firewall and security problems. This feature allows you to configure interfaces on which control and media packets can be exchanged. This new functionality allows you to separate signaling from voice by binding control (MGCP signaling) and media (Real-Time Transport Protocol, or RTP voice, fax, and modem) to specific gateway interfaces.

This feature includes new commands that can be used to configure the required interface for MGCP control and control of the required media packets.

Information About MGCP Gateway Support

If the media gateway (MG) uses an IP address, which is the outgoing interface of the MG, the media gateway controller (MGC) identifies the MG entity with that address. If that interface fails, MG sources MGCP from another interface, which is not known to the MGC. Some form of name lookup (host or Domain Name System, or DNS) needs to occur on the MGC at this time. Using the mgcp bindcommand, a loopback interface IP address can be used for sourcing MGCP packets, which is transparent to any interface failure.

Present implementation of MGCP media uses the "loopback 0" or best available IP address in the order indicated for media. A fixed default loopback 0 address for media streams breaks the dial address pool convention used for most configurations, where dial IP addresses are assigned from the loopback 0 address range. With this feature, it is possible to assign a media bind interface other than loopback 0, which helps dial calls conserve IP addresses.

If you configure the loopback interface for control, you can use routing capability more efficiently. Using themgcp bind command to configure the loopback interface helps in creating redundant MGCP control or media interface.

In the current implementation of MGCP, the source address of MGCP and media control is given by the IP layer. Because of this inconsistency, it is not possible to include a reliable access list or firewall configuration. Using the mgcp bind command for both MGCP and media control, you can get a predefined interface or IP address that can be used for security configuration.

The figure below shows a typical configuration flow using themgcp bind command.

Figure 9

Bind Configuration Flowchart

The figure below shows how the mgcp bind command takes effect for MGCP control. When themgcp bind command is configured for MGCP control, the MGCP_SYS_SOCKET_CHANG system event is posted to MGCPAPP. This event is processed by opening a new socket based on the configured interface.

Figure 10

Bind Configuration for Control Flowchart

The time frame for execution of themgcp bind command for media is different from that for control. The figure below shows how themgcp bindcommand is used for media. In this case, the IP address used for media Session Description Protocol (SDP) negotiation is taken from the configured interface. This flow is not active until an MGCP call is created.

The function call to get an IP address for the media returns a configured interface IP address, a loopback interface IP address, or a best available IP address in the order specified in the figure.

Figure 11

Bind Configuration for Media Flowchart

How to Configure MGCP Gateway Support

- Configuring the MGCP Application

- Configuring the bind Command

- Verifying MGCP Gateway Support

Configuring the MGCP Application

SUMMARY STEPS

1.   enable

2.   configure terminal

3.   mgcp call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version]

4.   mgcp

5.   exit

DETAILED STEPS

Step 1 enable

Enables higher privilege levels, such as privileged EXEC mode. Enter your password if prompted.

Example:

Router> enable

Step 2 configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

Step 3 mgcp call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version]

Configures the MGCP protocol and corresponding call agent.

Example:

Router(config)# mgcp call-agent 209.165.200.225 service-type mgcp version 1.0

Step 4 mgcp

Enables MGCP on the gateway.

Example:

Router(config)# mgcp

Step 5 exit

Exits the current mode.

Example:

Router(config)# exit

Configuring the bind Command

SUMMARY STEPS

1.   enable

2.   configure terminal

3.   mgcp bind {control | media} source-interface interface-id

4.   mgcp

5.   exit

DETAILED STEPS

Step 1 enable

Enables higher privilege levels, such as privileged EXEC mode. Enter your password if prompted.

Example:

Router> enable

Step 2 configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

Step 3 mgcp bind {control | media} source-interface interface-id

Sets a source interface for signaling and media packets.

Example:

Router(config)# mgcp bind {control} source- interface FastEthernet

Step 4 mgcp

Enables MGCP on the gateway.

Example:

Router(config)# mgcp

Step 5 exit

Exits the current mode.

Example:

Router(config)# exit

Troubleshooting Tips

To troubleshoot the MGCP Gateway Support for the Bind Command feature, use the debug mgcp command to enable debug traces for MGCP errors, events, media, packets, parser, and call admission control (CAC).

The following example illustrates the output for the debug mgcp command with the all keyword:

Router# debug mgcp all

20:54:13: MGC stat - 192.168.10.10, total=37, succ=28, failed=8

20:54:13: MGCP Packet received -

CRCX 55560 s0/ds1-0/1 SGCP 1.1

C: 78980

M: sendrecv

L: a:G.726-16

20:54:13: -- mgcp_parse_packet() - call mgcp_parse_header

- mgcp_parse_header()- Request Verb FOUND CRCX

- mgcp_parse_packet() - out mgcp_parse_header

- SUCCESS: mgcp_parse_packet()-MGCP Header parsing was OK

- mgcp_parse_parameter_lines(), code_str:: 78980, code_len:2, str:1640150312

- mgcp_parse_parameter_lines(str:C: 78980) -num_toks: 19

- mgcp_parse_parameter_lines() check NULL str(78980), in_ptr(C: 78980)

- mgcp_parse_parameter_lines() return Parse function in

mgcp_parm_rules_array[1]

- mgcp_parse_call_id(in_ptr: 78980)

- SUCCESS: mgcp_parse_call_id()-Call ID string(78980) parsing is OK

- mgcp_parse_parameter_lines(), code_str:: sendrecv, code_len:2, str:1640150312

- mgcp_parse_parameter_lines(str:M: sendrecv) -num_toks: 19

- mgcp_parse_parameter_lines() check NULL str(sendrecv), in_ptr(M: sendrecv)

- mgcp_parse_parameter_lines() return Parse function in

mgcp_parm_rules_array[6]

- mgcp_parse_conn_mode(in_ptr: sendrecv)

- mgcp_parse_conn_mode()- tmp_ptr:(sendrecv)

- mgcp_parse_conn_mode(match sendrecv sendrecv

- mgcp_parse_conn_mode(case MODE_SENDRECV)

- SUCCESS: Connection Mode parsing is OK

- mgcp_parse_parameter_lines(), code_str:: a:G.726-16, code_len:2,

str:1640150312

- mgcp_parse_parameter_lines(str:L: a:G.726-16) -num_toks: 19

- mgcp_parse_parameter_lines() check NULL str(a:G.726-16), in_ptr(L:

a:G.726-16)

- mgcp_parse_parameter_lines() return Parse function in mgcp_parm_rules_array[5]

- mgcp_parse_con_opts()

- mgcp_parse_codecs()

- SUCCESS: CODEC strings parsing is OK- SUCCESS: Local Connection option

parsing is OK- mgcp_val_mandatory_parms()

20:54:13: - SUCCESS: mgcp_parse_packet()- END of Parsing

20:54:13: MGCP msg 1

20:54:13: mgcp_search_call_by_endpt: endpt = s0/ds1-0/1, new_call = 1

20:54:13: slot=0,ds1=0,ds0=1

20:54:13: search endpoint - New call=1, callp 61C28130

20:54:13: callp: 61C28130, vdbptr: 0, state: 0

20:54:13: mgcp_remove_old_ack:

20:54:13: mgcp_idle_crcx: get capability

passthru is 3

20:54:13: process_request_ev- callp 61C28130, voice_if 61C281A4

20:54:13: process_detect_ev- callp 61C28130, voice_if 61C281A4

process_signal_ev- callp 61C28130, voice_ifp 61C281A4

20:54:13: mgcp_process_quarantine_mode- callp 61C28130, voice_if 61C281A4

20:54:13: mgcp_process_quarantine_mode- new q mode: process=0, loop=0

20:54:13: mgcp_xlat_ccapi_error_code - ack_code_tab_index = 0,

20:54:13: No SDP connection info

20:54:13: mgcp_select_codec - LC option, num codec=1, 1st codec=5

20:54:13: mgcp_select_codec - num supprt codec=11

20:54:13: mgcp_select_codec - LC codec list only

20:54:13: codec index=0, bw=16000, codec=5

20:54:13: selected codec=5mgcp_get_pkt_period: voip_codec=2, pkt_period=0, call

adjust_packetization_period

mgcp_get_pkt_period: voip_codec=2, pkt_period=10, after calling

adjust_packetization_period

20:54:13: selected codec 5

20:54:13: IP Precedence=60

20:54:13: MGCP msg qos value=0mgcp_get_pkt_period: voip_codec=2, pkt_period=0,

call adjust_packetization_period

mgcp_get_pkt_period: voip_codec=2, pkt_period=10, after calling

adjust_packetization_period

mgcp_new_codec_bytes: voip_codec=2, pkt_period=10, codec_bytes=20

20:54:13: callp : 61C28AE8, state : 2, call ID : 40, event : 5, minor evt:

1640137008

20:54:13: MGCPAPP state machine: state = 2, event = 5

20:54:13: mgcp_call_connect: call_id=40, ack will be sent later.

20:54:13: callp : 61C28AE8, new state : 3, call ID : 40

20:54:14: xlate_ccapi_ev - Protocol is SGCP, change pkg=2

20:54:14: MGCP Session Appl: ignore CCAPI event 22, callp 61C28130

20:54:14: xlate_ccapi_ev - Protocol is SGCP, change pkg=2

20:54:14: callp : 61C28130, state : 2, call ID : 39, event : 5, minor evt: 20

20:54:14: MGCPAPP state machine: state = 2, event = 5

20:54:14: callp : 61C28130, new state : 3, call ID : 39

20:54:14: xlate_ccapi_ev - Protocol is SGCP, change pkg=2

20:54:14: callp : 61C28130, state : 3, call ID : 39, event : 6, minor evt: 20

20:54:14: MGCPAPP state machine: state = 3, event = 6

20:54:14: call_id=39, mgcp_ignore_ccapi_ev: ignore 6 for state 3

20:54:14: callp : 61C28130, new state : 3, call ID : 39

20:54:14: MGCP voice mode event

20:54:14: xlate_ccapi_ev - Protocol is SGCP, change pkg=2

20:54:14: callp : 61C28130, state : 3, call ID : 39, event : 17, minor evt: 0

20:54:14: MGCPAPP state machine: state = 3, event = 17

20:54:14: mgcp_voice_mode_done(): callp 61C28130, major ev 17,

minor ev 0mgcp_start_ld_timer: timer already initialized

20:54:14: send_mgcp_create_ack

20:54:14: map_mgcp_error_code_to_string error_tab_index = 0, protocol version:

2

20:54:14: MGC stat - 1.13.89.3, total=37, succ=29, failed=8

20:54:14: Codec Cnt, 1, first codec 5

20:54:14: First Audio codec, 5, local encoding, 96

20:54:14: -- mgcp_build_packet()-

20:54:14: - mgcp_estimate_msg_buf_length() - 87 bytes needed for header

- mgcp_estimate_msg_buf_length() - 125 bytes needed after checking parameter lines

- mgcp_estimate_msg_buf_length() - 505 bytes needed after cheking SDP lines

20:54:14: --- mgcp_build_parameter_lines() ---

- mgcp_build_conn_id()

- SUCCESS: Conn ID string building is OK

- SUCCESS: Building MGCP Parameter lines is OK

- SUCCESS: building sdp owner id (o=) line

- SUCCESS: building sdp session name (s=) line

- SUCCESS: MGCP message building OK

- SUCCESS: END of building

updating lport with 2427

20:54:14: send_mgcp_msg, MGCP Packet sent --->

200 55560

I: 10

v=0

o=- 78980 0 IN IP4 192.168.10.9

s=Cisco SDP 0

c=IN IP4 192.168.10.9

t=0 0

m=audio 16444 RTP/AVP 96

a=rtpmap:96 G.726-16/8000/1

<---

20:54:14: enqueue_ack: voice_if=61C281A4, ackqhead=0, ackqtail=0,

ackp=61D753E8, msg=61D00010

20:54:14:

mgcp_process_quarantine_after_ack:ack_code=200mgcp_delete_qb_evt_q:cleanup QB

evt q

20:54:14: callp : 61C28130, new state : 4, call ID : 39

Verifying MGCP Gateway Support

SUMMARY STEPS

1.    Router# show mgcp

2.    Router# show ip socket

3.    Router# show running-configuration

DETAILED STEPS

Step 1 Router# show mgcp

Checks your configuration.

Step 2 Router# show ip socket

Displays IP socket information.

Step 3 Router# show running-configuration

Verifies bind functionality.

Configuration Examples for MGCP Gateway Support

The following is partial output from theshow running-configuration command indicating that bind is functional on receiving router 172.18.192.204. Updated output for MGCP binding is highlighted under the voice service VoIP indicator.

ip subnet-zero

ip ftp source-interface Ethernet0

!

voice service voip

mgcp bind control source-interface FastEthernet0

mgcp bind media source-interface FastEthernet0

!

interface FastEthernet0

ip address 172.18.192.204 255.255.255.0

duplex auto

speed auto

fair-queue 64 256 1000

ip rsvp bandwidth 75000 100

!

Configuring MGCP CAS MD Package

This chapter provides information on configuring the MGCP channel-associated signaling (CAS) MD Package feature. This feature introduces support for Feature Group D (FGD) Exchange Access North American (EANA) protocol signaling. The CAS MD package adds support for the reporting of automatic number identification (ANI) and dialed number identification service (DNIS) digits to enable the MGCP call agent to better handle customer billing.

For more information about this and related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm .

Feature History for MGCP CAS MD Package

Release

Modification

12.4(4)T

This feature was introduced on the Cisco AS5850.

12.4(15)T

Support was added for the Cisco AS5350, Cisco AS5350XM, Cisco AS5400XM, and Cisco AS5400HPX platforms.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for MGCP CAS MD Package

Prerequisites are described in "Prerequisites for Configuring MGCP and Related Protocols" on page 3.

Restrictions for MGCP CAS MD Package

FGD Exchange Access International (EAIN) signaling is not supported.

Information About MGCP CAS MD Package

- MD Package

MD Package

The MD package supports the FGD EANA protocol for T1 CAS interfaces as defined in RFC 3064. It includes support for ANI and DNIS reporting that enables the MGCP call agent to improve its handling of customer billing. The MD package is enabled automatically when a T1 interface is configured using the ds0-group command with the fgd-eana keyword. The order in which the voice gateway sends the ANI and DNIS digits can be controlled by using the notify command in the MGCP profile.

How to Configure the MGCP CAS MD Package

Note

You do not have to enable the CAS MD package with the mgcp package-capability command. The CAS MD package is enabled automatically when a T1 controller is configured for FGD EANA signaling using the ds0-group command.

Configuring the Incoming Called Number in the MGCP Dial Peer

Perform this procedure to specify the dial string to use for matching incoming calls to the MGCP dial peer.

SUMMARY STEPS

1.   enable

2.   configure terminal

3.   dial-peer voice tag pots

4.   service mgcpapp

5.   incoming called number string

6.   port port

7.   end

DETAILED STEPS

Step 1 enable

Enables privileged EXEC mode.

- Enter your password if prompted.

Example:

Router> enable

Step 2 configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

Step 3 dial-peer voice tag pots

Defines a dial peer as a POTS device and enters dial-peer configuration mode.

Example:

Router(config)# dial-peer voice 1003 pots

Step 4 service mgcpapp

Enables MGCP on the dial peer.

Note

Do not use this command in dial peers that support PRI backhaul or BRI backhaul.

Example:

Router(config-dial-peer)# service mgcpapp

Step 5 incoming called number string

Specifies the digit string that is used to match incoming calls to the dial peer.

Example:

Router(config-dial-peer)# incoming called number .

Step 6 port port

Binds the MGCP application to the specified voice port.

Example:

Router(config-dial-peer)# port 0/0:3:0

Step 7 end

Exits to privileged EXEC mode.

Example:

Router(config-dial-peer)# end

Modifying ANI and DNIS Order when Using CAS MD Package

Perform this procedure to specify the order in which ANI and DNIS digits are sent in notify messages to the call agent when using the CAS MD package.

SUMMARY STEPS

1.   enable

2.   configure terminal

3.   mgcp profile {profile-name | default}

4.   notify {ani-dnis | dnis-ani}

5.   end

6.   show mgcp profile [profile-name]

DETAILED STEPS

Step 1 enable

Enables privileged EXEC mode.

- Enter your password if prompted.

Example:

Router> enable

Step 2 configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

Step 3 mgcp profile {profile-name | default}

Defines an MGCP profile to be associated with one or more MGCP endpoints.

Example:

Router(config)# mgcp profile default

Step 4 notify {ani-dnis | dnis-ani}

Specifies the order in which ANI and DNIS digits are reported to the MGCP call agent.

- ani-dnis --ANI digits are sent in the first notify message. This is the default order.

- dnis-ani --DNIS digits are sent in the first notify message.

Example:

Router(config-mgcp-profile)# notify dnis-ani

Step 5 end

Exits to privileged EXEC mode.

Example:

Router(config-mgcp-profile)# end

Step 6 show mgcp profile [profile-name]

Displays configuration information for MGCP profiles including the setting of the notify command.

Example:

Router# show mgcp profile default

Configuration Examples for MGCP CAS MD Package

- CAS MD Package Configuration Example

- Cisco AS5850 Configuration Example

CAS MD Package Configuration Example

The following example shows the significant portions of a configuration for the CAS MD package.

...

controller T1 0/0:3

framing esf

ds0-group 0 timeslots 1 type fgd-eana mf ani-dnis

!

controller T1 0/0:4

framing esf

ds0-group 0 timeslots 1 type fgd-eana mf ani-dnis

...

mgcp profile default

notify dnis-ani

!

!

dial-peer voice 1003 pots

service mgcpapp

incoming called-number .

port 0/0:3:0

!

dial-peer voice 1004 pots

service mgcpapp

incoming called-number .

port 0/0:4:0

...

Cisco AS5850 Configuration Example

The following example shows a complete running configuration for a Cisco AS5850 universal gateway that is using the CAS MD package.

Current configuration : 2636 bytes

!

version 12.4

no service pad

service timestamps debug datetime msec localtime

service timestamps log datetime msec localtime

no service password-encryption

service internal

!

hostname Sample

!

boot-start-marker

boot system flash:c5850-p9-mz

boot-end-marker

!

!

redundancy

mode classic-split

logging buffered 20000000 debugging

no logging console

enable password temp

!

no aaa new-model

!

resource policy

!

!

!

resource-pool disable

dial-tdm-clock priority 1 external t1 120ohm

spe default-firmware spe-firmware-1

!

!

!

!

ip subnet-zero

ip cef distributed

!

!

!

isdn switch-type primary-5ess

!

!

!

!

!

!

controller T3 0/0

framing c-bit

cablelength 224

t1 1-7 controller

!

controller T1 0/0:1

shutdown

framing sf

ds0-group 0 timeslots 1 type fgd-eana mf ani-dnis

!

controller T1 0/0:2

shutdown

framing sf

ds0-group 0 timeslots 1 type fgd-eana mf ani-dnis

!

controller T1 0/0:3

framing esf

ds0-group 0 timeslots 1 type fgd-eana mf ani-dnis

!

controller T1 0/0:4

framing esf

ds0-group 0 timeslots 1 type fgd-eana mf ani-dnis

!

controller T1 0/0:5

framing esf

!

controller T1 0/0:6

shutdown

framing esf

!

controller T1 0/0:7

framing esf

!

!

!

interface Loopback0

no ip address

no ip route-cache cef

no ip route-cache distributed

no ip route-cache

!

interface FastEthernet6/0

ip address 172.16.0.46 255.255.255.0

no ip proxy-arp

logging event link-status

speed 100

full-duplex

no keepalive

!

interface GigabitEthernet6/0

no ip address

logging event link-status

shutdown

negotiation auto

!

interface GigabitEthernet6/1

no ip address

logging event link-status

shutdown

negotiation auto

!

interface Group-Async0

no ip address

encapsulation ppp

group-range 0/00 3/323

!

!

ip classless

ip route 0.0.0.0 0.0.0.0 172.16.0.200

no ip http server

!

!

!

!

!

!

voice-port 0/0:1:0

!

voice-port 0/0:2:0

!

voice-port 0/0:3:0

!

voice-port 0/0:4:0

!

mgcp

mgcp call-agent 172.16.0.200 18384 service-type mgcp version 0.1

mgcp package-capability dtmf-package

mgcp package-capability mf-package

mgcp package-capability rtp-package

no mgcp piggyback message

mgcp persistent onhook

mgcp fax t38 inhibit

!

mgcp profile default

!

!

dial-peer voice 1003 pots

service mgcpapp

incoming called-number .

port 0/0:3:0

!

dial-peer voice 1004 pots

service mgcpapp

incoming called-number .

port 0/0:4:0

!

!

!

line con 0

exec-timeout 0 0

transport output all

line aux 0

exec-timeout 0 0

transport output all

line vty 0 4

exec-timeout 0 0

privilege level 15

no login

transport input all

transport output all

line 0/00 0/215

modem InOut

transport input all

line 3/00 3/323

modem InOut

transport input all

!

end

Media and Signaling Authentication and Encryption

The Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature provides support for Cisco Secure Survivable Remote Site Telephony (SRST) and voice security features that include authentication, integrity, and encryption of voice media and related call control signaling.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for Media and Signaling Authentication and Encryption

Make sure that the following tasks have been completed before configuring the Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature:

- Cisco IOS Media Gateway Control Protocol (MGCP) is configured.

- Cisco Unified Communications Manager 4.1(2) or a later release is running.

- Cisco Secure SRST is configured on the router. For more information on configuring secure SRST on the router, refer to the document Setting up Secure SRST.

- Cisco IOS gateways have the prerequisite Cisco IOS images installed. Voice security features are delivered on Advanced IP Services or Advanced Enterprise Services images.

It is recommended that IP security (IPsec) be configured on the Cisco IOS gateway. Both software and hardware-based IPsec connections are supported.

For more information on configuring Cisco IOS-based (software) IPsec, refer to the following:

- Cisco IOS Security Configuration Guide, Release 12.3

- Cisco IOS Security Command Reference, Release 12.3

For more information on configuring hardware-based IPsec on the gateway, refer to the following:

- Cisco 2621 Modular Access Router with AIM-VPN/BP Security Policy

- Cisco 2651 Modular Access Router with AIM-VPN/BP Security Policy

- Cisco 3640 Modular Access Router with AIM-VPN/BP Security Policy

- Cisco 3660 Modular Access Router with AIM-VPN/BP Security Policy

It is recommended that IPsec be configured on the Cisco CallManager. For more information, refer to the Microsoft Knowledge Base article " Configuring IPsec Between a Microsoft Windows 2000 Server and a Cisco Device. "

If you want to interoperate with Cisco IP phones, make sure that the following tasks have been completed before configuring the Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature:

- Cisco CallManager is set up for secure mode operation, and a certificate trust list (CTL) client is installed. For more information on CTL client setup, refer to Cisco IP Phone Authentication and Encryption for Cisco CallManager 4.0(1), "Authentication, Integrity and Encryption" chapter.

- The phones are configured to support secure calls if the gateways will interoperate with Cisco IP phones. For more information on Cisco IP phone configuration, refer to the following:

- Cisco IP Phone Model 7960G and 7940G Administration Guide for Cisco CallManager Release 4.2, "Security Configuration Menu" section.

- Cisco IP Phone 7970 Administration Guide for Cisco CallManager Release 4.x and later, "Understanding Security Features for Cisco IP Phones" section.

Restrictions for Media and Signaling Authentication and Encryption

The Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature is supported on Cisco IOS MGCP 0.1 and Cisco IOS MGCP 1.0.

Cisco IOS MGCP SRTP support on the Cisco AS5400XM gateway is limited to the c5510 digital signal processors (DSP) series.

Cisco IOS MGCP gateways support voice security features on the following endpoints only: T1, E1, FXS, and FXO.

When a Cisco IOS MGCP voice gateway is used in conjunction with the Cisco CallManager, the automatic download feature that allows you to complete the gateway configuration on the Cisco CallManager server by downloading the configuration to that gateway through a TFTP server is not supported with voice security features.

Voice security during conferencing, transcoding, and music-on-hold is not supported.

Note

If one component in the voice gateway path is not secure, the entire call falls back to nonsecure mode.

The table below provides a list of supported IP phones, gateways and network modules for voice security features.

Table 5

Supported Products for Voice Security Features

Supported Cisco IP Phones

Supported Gateways

Supported Network Modules

- Cisco IP Phone 7940

- Cisco IP Phone 7960

- Cisco IP Phone 7970

- Cisco 2600XM

- Cisco 2691

- Cisco 2811

- Cisco 2821

- Cisco 2851

- Cisco 3640A

- Cisco 3660

- Cisco 3700

- Cisco 3825

- Cisco 3845

- Cisco VG224

- Cisco AS5400XM

- EVM-HD

- NM-HDV2

- NM-HDV2-1T1/E1

- NM-HDV2-2T1/E1

- NM-HD-1V

- NM-HD-2V

- NM-HD-2VE

- PVDM2

Voice security features impact quality of service (QoS) as follows:

- The Secure Real-Time Transport Protocol Control Protocol (SRTCP) packet size increases by an 80-bit authentication tag, a 31-bit index field, and a 1-bit encryption flag.

- The bandwidth of Real-Time Transport Protocol (RTP) streams increases slightly with the introduction of the 32-bit authentication tag on every SRTP packet sent. Additional bandwidth is required for supported SRTP codecs as shown in the table below.

Table 6

SRTP Codec Bandwith Requirements

Codec

Packetization Period (milliseconds)

RTP Bandwidth (kbps)

SRTP Bandwidth (kbps)

G.711 mu-law, G.711 A-law

10-20

96-80

99.2-81.6

G.729, G.729A

10-220

40-9.454

43.2-9.6

Only Clear Channel, G.711, and G.729 codecs support voice security features.

Voice security features support channel density on the TI-5510 DSP as shown in the table below.

Table 7

TI -5510 DSP Channel Density

Codec

Number of Nonsecure Calls

Number of Secure Calls

Clear Channel, G.711

16

10

G.729

6

6

G.729A

8

8

Use the codec complexity command in voice-card configuration mode to specify secure codec complexity and call density per DSP.

Information About Media and Signaling Authentication and Encryption

- Benefits of Media and Signaling Authentication and Encryption

- Feature Design

Benefits of Media and Signaling Authentication and Encryption

- Provides privacy and confidentiality for voice calls

- Protects against voice security violations

Feature Design

The Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature implements voice security features that include signaling authentication along with media and signaling encryption on MGCP gateways.

The feature provides secure VoIP calls by addressing security requirements for privacy, integrity, and confidentiality of voice conversations. The Cisco IP telephony network establishes and maintains authenticated communications using authentication and encryption technology. Signaling authentication validates that no tampering has occurred to signaling packets during transmission.

Encryption, the process of converting clear-text data into enciphered data, provides data integrity and authentication. IPsec, a standards-based set of security protocols and algorithms, ensures that signaling information (that is, Dual Tone Multi-Frequency (DTMF) digits, passwords, personal identification numbers (PINs), and encryption keys) that is sent between the gateway and Cisco CallManager is encrypted. Media encryption using standards-based SRTP ensures that media streams between supported devices are secure.

Voice security features support the following capabilities between gateways and from gateways to IP phones that support the encryption feature:

- Gateway to Cisco CallManager call control authentication and encryption using IPsec

- Media encryption and authentication of voice RTP streams using SRTP

- Exchange of RTP Control Protocol (RTCP) information using SRTCP

- SRTP to RTP fallback for calls between secure and nonsecure endpoints

- Secure to clear-text fallback for new calls during SRST operation

The figure below shows a typical topology where voice security features are deployed.

Figure 12

Voice Security Features in the Telephony Network

MGCP Gateway Behavior and Voice Security Features

To implement voice security features in Cisco CallManager networks, the MGCP gateway communicates with Cisco CallManager over a secure IPsec connection that provides encryption of IP packets. To ensure that your signaling information is secure, estacblish an IPsec connection between the CallManager and the gateways, as described in the Prerequisites for Media and Signaling Authentication and Encryption section. You can verify that the IPsec tunnel is secure using the commands listed in the Verifying Voice Security Features section.

Note

Although you may enable media authentication and encryption without signaling encryption, this practice is discouraged. If the gateway to Cisco CallManager connection is not secure, media keys will be sent in clear-text and your voice call will not be considered secure.

After the IPsec tunnel is established, all call control and signaling of MGCP packets between the gateway and Cisco CallManager go through the secured IPsec tunnel, with the Cisco CallManager directing the MGCP gateway to set up and tear down SRTP streams. SRTP media keys are distributed by Cisco CallManager through the secured IPsec tunnel.

Cisco implements voice security features on MGCP gateways by supporting the SRTP package and SRTP Session Description Protocol (SDP) extensions, as defined in the Internet Engineering Task Force (IETF) specifications draft-ietf-mmusic-sdescriptions-02.txt (Security Descriptions for Media Streams and RFC 4568, Session Description Protocol (SDP) Security Descriptions for Media Streams ).

SRTP package capability is disabled by default. Use the Cisco IOS command-line interface (CLI) to enable the feature. For more information, see theConfiguring Voice Security Features section.

Cisco uses the Internet Key Exchange (IKE) standard to implement IPsec. IKE provides authentication of the IPsec peers and negotiates IPsec keys and IPsec security associations (SAs). An IPsec SA describes how two or more entities will use security services to communicate securely. For example, an IPsec SA defines the encryption algorithm, the authentication algorithm, and the shared session key to be used during the IPsec connection. Both IPsec and IKE require and use SAs to identify the parameters of their connections. IKE can negotiate and establish its own SA. The IPsec SA is established either by IKE or by manual user configuration. IKE has two phases of key negotiation: phase 1 and phase 2. Phase 1 negotiates a security association (a key) between two IKE peers. The key negotiated in phase 1 enables IKE peers to communicate securely in phase 2. During phase 2 negotiation, IKE establishes keys (security associations) for other applications, such as IPsec.

The Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature also implements an extended codec selection algorithm that combines selection of a codec with selection of a cryptographic suite to be used to encrypt the RTP stream. Cisco IOS Release 12.3(11)T supports the AES_CM_128_HMAC_SHA1_32 cryptographic suite, which includes the AES-128-countermode encryption algorithm and the Hashed Message Authentication Codes (HMAC) Secure Hash Algorithm1 (SHA1) authentication algorithm.

Note

MGCP for SRTP on Cisco IOS gateways can be configured to use either MGCP 1.0 signaling support with the Cisco public switched telephone network (PSTN) Gateway (PGW) 2200 carrier-class call agent, or MGCP 0.1 signaling support with Cisco Unified Communications Manager.

Voice Security Features Interoperability with Endpoints

Cisco IOS MGCP gateways support voice security features on T1, E1, FXS, and FXO endpoints supported by network modules listed in Voice Security Features Interoperability with Endpoints, thereby enabling secure calls from analog phone to analog phone, or fax machine to fax machine. Similarly, secure calls are enabled from time- division multiplexing (TDM) endpoints or analog phones to Cisco IP phones. For a Cisco IP Phone to make and receive secure calls, all endpoints, that is, phones of all call participants, must support voice security features. If a call is nonsecure, no special icon displays on the phone. If a call is secure, the phone displays either the authenticated or encrypted call icons. For more information on secure call icons, refer to Cisco IP Phone 7970 Administration Guide for Cisco CallManager, Release 4.x or later, "Identifying Encrypted and Authenticated Phone Calls" section.

How to Configure Media and Signaling Authentication and Encryption Feature

- Installing Cisco CallManager

- Configuring IPsec on Cisco CallManager

- Configuring the Cisco PGW

- Configuring Voice Security Features

- Configuring Secure IP Telephony Calls

- Verifying Voice Security Features

Installing Cisco CallManager

This task installs Cisco CallManager and configures it to work with IPsec and the Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature.

SUMMARY STEPS

1.    Install Cisco CallManager on the server.

2.    Determine the Windows OS version by going to C:\utils and double-clicking MCSVer.exe program. If you have Windows 2000.2.6sr3, no additional Windows upgrade is required. Go to Installing Cisco CallManager. If you have Windows 2000.2.5 or a prior version, you must upgrade to Windows 2000.2.6. Go to Installing Cisco CallManager. If you have Windows 2000.2.6, you must upgrade to Windows 2000.2.6sr3. Go to Installing Cisco CallManager.

3.    Upgrade from Windows 2000.2.5 or a prior version.

4.    Upgrade from Windows 2000.2.6 to Windows 2000.2.6.sr3.

5.    Upgrade Cisco CallManager to version 4.1.

6.    Use the ping command on both the gateway and Cisco CallManager to test the connection between the gateway and Cisco CallManager. Go to the Configuring IPsec on Cisco CallManager.

DETAILED STEPS

Step 1 Install Cisco CallManager on the server.

- Insert Cisco CallManager HW Detection CD version 2000.2.6, Disk1.

- When prompted, insert Cisco CallManager Base OS CD , Disk3 or 4.

Step 2 Determine the Windows OS version by going to C:\utils and double-clicking MCSVer.exe program. If you have Windows 2000.2.6sr3, no additional Windows upgrade is required. Go to Installing Cisco CallManager. If you have Windows 2000.2.5 or a prior version, you must upgrade to Windows 2000.2.6. Go to Installing Cisco CallManager. If you have Windows 2000.2.6, you must upgrade to Windows 2000.2.6sr3. Go to Installing Cisco CallManager.

Step 3 Upgrade from Windows 2000.2.5 or a prior version.

- Go to http://www.cisco.com/cgi-bin/tablebuild.pl/cmva-3des , to download the following files:

- win-OS-Upgrade-K9.2000-2-6.exe.

- win-OS-Upgrade-K9.2000-2-6-Readme.htm

Follow the steps listed in the ReadMe file.

Step 4 Upgrade from Windows 2000.2.6 to Windows 2000.2.6.sr3.

- Go to http://www.cisco.com/cgi-bin/tablebuild.pl/cmva-3des , to download the following files:

- win-OS-Upgrade-K9.2000-2-6sr3.exe

- win-OS-Upgrade-K9.2000-2-6sr3-Readme.htm.

Follow the steps listed in the ReadMe file.

Step 5 Upgrade Cisco CallManager to version 4.1.

- Go to http://www.cisco.com/cgi-bin/tablebuild.pl/cmva-3des .

- Copy CiscoCallManagerUpgrade.exe to the local system.

- Run the upgrade.

Step 6 Use the ping command on both the gateway and Cisco CallManager to test the connection between the gateway and Cisco CallManager. Go to the Configuring IPsec on Cisco CallManager.

Configuring IPsec on Cisco CallManager

This task configures the IPsec connection between the MGCP gateway and the Cisco CallManager.

SUMMARY STEPS

1.    Create an IPsec policy on the Windows 2000 server.

2.    Build a filter from the Cisco CallManager to the gateway.

3.    Build a filter from the gateway to the Cisco CallManager.

4.    Configure a rule to negotiate tunnel security.

5.    Set key exchange security methods.

6.    Assign the new IPsec policy to the Windows 2000 gateway.

7.    Use the ping command on both the gateway and Cisco CallManager to test the connection between the gateway and Cisco CallManager.

8.    Run ipsecmon.exe on the Cisco CallManager to verify the configuration.

9.    Use the show crypto isakmp sa command on the gateway to verify the IPsec configuration.

DETAILED STEPS

Step 1 Create an IPsec policy on the Windows 2000 server.

- Use the Microsoft Management Console (MMC) to work on the IP Security Policy Management snap-in. Click Start, click Run, and then enter secpol.msc.

- Right-click IP Security Policies on Local Machine, and then click Create IP Security Policy.

- Click Next, and then type a name for your policy.

- Clear the Activate the default response rule check box, and then click Next.

- Click Finish, while keeping the Editcheck box chosen.

Step 2 Build a filter from the Cisco CallManager to the gateway.

- In the properties for the new policy created in Configuring IPsec on Cisco CallManager, clear the Use Add Wizard check box, and then click Add to create a new rule.

- On the IP Filter List tab, click Add.

- Enter an appropriate name for the filter list, clear the Use Add Wizard check box, and then click Add.

- In the Source address area, choose the option My IP Addressfrom the drop-down arrow. Enter the Cisco CallManager IP address.

- In the Destination address area, click A specific IP Subnetfrom the drop-down arrow. Enter the IP address of the router interface in the same subnet as the Cisco CallManager.

- Clear the Mirroredcheck box.

- On the Protocol tab, make sure the protocol type is set to Any. (IPsec tunnels do not support protocol-specific or port-specific filters.)

- (Optional) If you want to enter a description for your filter, click the Descriptiontab. It is recommended that you give the filter the same name you used for the filter list. The filter name is displayed in the IPsec monitor when the tunnel is active.

- Click OK, and then click Close.

Step 3 Build a filter from the gateway to the Cisco CallManager.

- On the IP Filter List tab, click Add.

- Type an appropriate name for the filter list, clear the Use Add Wizard check box, and then click Add.

- In the Source address area, click A specific IP Subnetfrom the dropdown arrow. Enter the IP address of the router interface in the same subnet as the Cisco CallManager.

- In the Destination address area, choose the option My IP Addressfrom the dropdown arrow.

- Clear the Mirroredcheck box.

- (Optional) If you want to enter a description for your filter, click the Description tab.

- Click OK, and then click Close.

Step 4 Configure a rule to negotiate tunnel security.

- On the IP Filter List tab, click the filter list you created in Configuring IPsec on Cisco CallManager.

- On the Tunnel Setting tab, choose the option Tunnel Setting - encryption peers. For Cisco-Microsoft and for Microsoft-Cisco, configure the setting according to: http://www.cisco.com/en/US/tech/tk583/tk372/technologies_configuration_example09186a00800b12b5.shtml

- On the Connection Type tab, click All network connections.

- On the Filter Action tab, clear the Use Add Wizard check box, and then click Add to create a new filter action.

Note

You must create a new filter; otherwise the default filter action allows incoming traffic in the clear.

- Keep the Negotiate security option enabled, and click the Accept unsecured communication, but clear the always respond using IPsec check box.

Note

You must perform this step to ensure secure operation.

- Choose the Custom option to add a security method. Click the Data integrity and encryption box for Encapsulating Security Payload (ESP). Click MD5 for the Integrity algorithm. Click DESfor the Encryption algorithm. Check the Generate a new Key every 3600 seconds box.

- Click OK. On the General tab, enter a name for the new filter action and then click OK.

- Choose the filter action you created in Configuring IPsec on Cisco CallManager.

- On the Authentication Methods tab, perform the steps to configure a preshared key.

Note

The preshared key must match the key configured on the router.

- Click Close.

Step 5 Set key exchange security methods.

- Right-click the IP Security Policy created in Configuring IPsec on Cisco CallManager and choose Properties.

- Click the General tab.

- Click the Advancedbutton.

- Click the Methods button.

- Ensure that the security Method with the following settings is at the top of the preference order: Type--IKE, Encryption--DES, Integrity--SHA1, Diffie-Hellman--Low(1)

- Save the configuration.

Step 6 Assign the new IPsec policy to the Windows 2000 gateway.

- In the IP Security Policies on Local Machine MMC snap-in, right-click the new policy, and then click Assign. A green arrow appears in the folder icon next to the new policy.

Step 7 Use the ping command on both the gateway and Cisco CallManager to test the connection between the gateway and Cisco CallManager.

Step 8 Run ipsecmon.exe on the Cisco CallManager to verify the configuration.

Step 9 Use the show crypto isakmp sa command on the gateway to verify the IPsec configuration.

Configuring the Cisco PGW

MGCP for SRTP on Cisco IOS gateways can be configured for use with the Cisco PSTN gateway (PGW) 2200 carrier-class call agent.

To configure this feature, you must first tell the Cisco PGW 2200 Softswitch that the media gateways support SRTP. Then you specify that SIP and TDM trunk groups support SRTP.

For a detailed description of the configuration tasks, see the "Secure Real-time Transport Protocol Support" feature guide at the following location:

http://www.cisco.com/en/US/docs/voice_ip_comm/pgw/9/feature/module/9.8_1_/SRTP.html

Configuring Voice Security Features

This task configures voice security features on the Cisco IOS MGCP gateway.

Before You Begin

We strongly recommend that you first establish an IPsec connection between the Cisco CallManager and the MGCP gateway before you use the MGCP SRTP package. Otherwise, media keys will be sent in clear text and your voice call will not be considered secure. For more information, see the "Installing Cisco CallManager" and "Configuring IPsec on Cisco CallManager" sections.

SUMMARY STEPS

1.   enable

2.   configure terminal

3.   mgcp package-capability srtp-package

4.   mgcp validate call-agent source-ipaddr

5.   mgcp crypto rfc-preferred

6.   voice-card slot

7.   codec complexity secure

8.   exit

DETAILED STEPS

Step 1 enable

Enables privileged EXEC mode.

- Enter your password if prompted.

Example:

Router> enable

Step 2 configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

Step 3 mgcp package-capability srtp-package

Enables the MGCP gateway capability to process SRTP packages.

Example:

Router(config)# mgcp package-capability srtp-package

Step 4 mgcp validate call-agent source-ipaddr

(Optional) Enables MGCP application validation that packets received are sent by a configured call agent.

Example:

Router(config)# mgcp validate call-agent source-ipaddr

Step 5 mgcp crypto rfc-preferred

(Optional) Enables support for the media-level SDP a=crypto attribute on the Cisco IOS MGCP gateway.

Example:

Router(config)# mgcp crypto rfc-preferred

Step 6 voice-card slot

Enters voice-card configuration mode and configures the voice card in the specified network module slot.

Example:

Router(config)# voice-card 1

Step 7 codec complexity secure

Restricts the number of channels per NM-HDV network module from four to two, enabling SRTP support on the TI-549 DSP.

Note

You need not specify secure codec complexity for TI-5510 DSPs, which support SRTP capability in all complexity modes.

Example:

Router(config-voice-card)# codec complexity secure

Step 8 exit

Exits the current configuration mode.

Example:

Router(config-voice-card)# exit

Configuring Secure IP Telephony Calls

This task enables secure IP telephony calls from gateway to IP phone.

Voice security features use digital certificates contained in eTokens for device authentication. This process validates the identity of a device and ensures that the entity is who it claims to be. Device authentication occurs between the Cisco CallManager server and supported IP phones when each entity accepts the certificate of the other entity. Cisco implements device authentication using the CTL feature on the Cisco CallManager. The CTL Client creates a certificate on each server in the cluster and generates a CTL file in the TFTP Path of the server for the phones to download. This file provides the IP phone with a list of certified hosts that it can trust. For more information, refer to Cisco IP Phone Authentication and Encryption for Cisco CallManager 4.0(1), "Signaling Authentication" chapter .

Before You Begin

- CTL Provider service must be running on the Cisco CallManager server.

- Smart Card service must be running on the Cisco CallManager server.

- Two USB eTokens are required.

SUMMARY STEPS

1.    Install CiscoCTLClient.exe from c:\CiscoPlugins\Client\.

2.    Launch Cisco CTL Client from the desktop shortcut.

3.    Enter the Cisco CallManager IP address and password, then click Next.

4.    Choose Set CallManager Cluster to Secure Mode, then click Next.

5.    Click Add for Security Token Information.

6.    Click Add Tokensfor CTL Entries.

7.    When prompted, insert the first USB eToken, then click OK.

8.    Repeat Configuring Secure IP Telephony Calls and Configuring Secure IP Telephony Calls for the second eToken.

9.    Click Finish for CTL Entries, then enter your eToken Password when prompted and click OK.

10.    Verify that voice security features are enabled.

DETAILED STEPS

Step 1 Install CiscoCTLClient.exe from c:\CiscoPlugins\Client\.

Step 2 Launch Cisco CTL Client from the desktop shortcut.

Step 3 Enter the Cisco CallManager IP address and password, then click Next.

Step 4 Choose Set CallManager Cluster to Secure Mode, then click Next.

Step 5 Click Add for Security Token Information.

Step 6 Click Add Tokensfor CTL Entries.

Step 7 When prompted, insert the first USB eToken, then click OK.

Step 8 Repeat Configuring Secure IP Telephony Calls and Configuring Secure IP Telephony Calls for the second eToken.

Step 9 Click Finish for CTL Entries, then enter your eToken Password when prompted and click OK.

Step 10 Verify that voice security features are enabled.

- Open Cisco CallManager Administration, choose Access System, then Enterprise Parameters. Scroll down to Security Parameters, and verify that Cluster Security is set to 1.

- Set the Cisco CallManager Enterprise Parameter to Encryptedto force all devices in the cluster to run encrypted mode. You can also set each IP phone individually to encrypted mode by choosing Device, then Phone, then Find, then Security Mode = Encrypted. Reboot the IP phones and verify that the Security Mode displays Encrypted under Security Settings.

Verifying Voice Security Features

This task verifies voice security feature configuration and MGCP gateway to Cisco CallManager IPsec connections.

SUMMARY STEPS

1.   show mgcp

2.   show mgcp connection

3.   show mgcp srtp {summary| detail [endpoint]}

4.   show mgcp statistics

5.   show call active voice

6.   show voice call port

7.   show voice call status

8.   show voice call status call-id

9.   show voice dsp

10.   show rtpspi call

11.   show rtpspi statistics

12.   show ccm-manager

13.   show crypto engine accelerator statistic

14.   show crypto ipsec sa

15.   show crypto isakmp sa

16.   show crypto session

17.   show crypto session detail

DETAILED STEPS

Step 1 show mgcp

Use this command to display the state of the mgcp package-capability srtp-package and mgcp validate call-agent source-ipaddr commands.

Example:

Router# show mgcp

MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE

MGCP call-agent: 10.7.0.200 Initial protocol service is MGCP 0.1

The following line shows that call-agent validation is enabled:

Example:

MGCP validate call-agent source-ipaddr ENABLED

MGCP block-newcalls DISABLED

MGCP send SGCP RSIP: forced/restart/graceful/disconnected DISABLED

MGCP quarantine mode discard/step

MGCP quarantine of persistent events is ENABLED

MGCP dtmf-relay for VoIP disabled for all codec types

MGCP dtmf-relay for VoAAL2 disabled for all codec types

MGCP voip modem passthrough disabled

MGCP voaal2 modem passthrough disabled

MGCP voip modem relay: Disabled.

MGCP TSE payload: 100

MGCP T.38 Named Signalling Event (NSE) response timer: 200

MGCP Network (IP/AAL2) Continuity Test timer: 200

MGCP 'RTP stream loss' timer disabled

MGCP request timeout 500

MGCP maximum exponential request timeout 4000

MGCP gateway port: 2427, MGCP maximum waiting delay 3000

MGCP restart delay 0, MGCP vad DISABLED

MGCP rtrcac DISABLED

MGCP system resource check DISABLED

MGCP xpc-codec: DISABLED, MGCP persistent hookflash: DISABLED

MGCP persistent offhook: ENABLED, MGCP persistent onhook: ENABLED

MGCP piggyback msg DISABLED, MGCP endpoint offset DISABLED

MGCP simple-sdp ENABLED

MGCP undotted-notation DISABLED

MGCP codec type g711ulaw, MGCP packetization period 20

MGCP JB threshold lwm 30, MGCP JB threshold hwm 150

MGCP LAT threshold lwm 150, MGCP LAT threshold hwm 300

MGCP PL threshold lwm 1000, MGCP PL threshold hwm 10000

MGCP CL threshold lwm 1000, MGCP CL threshold hwm 10000

MGCP playout mode is adaptive 60, 4, 200 in msec

MGCP Fax Playout Buffer is 300 in msec

MGCP media (RTP) dscp: ef, MGCP signaling dscp: af31

MGCP default package: line-package

The following lines show that the srtp-packagecommand is enabled:

Example:

MGCP supported packages: gm-package dtmf-package mf-package trunk-package

line-package ms-package dt-package mo-package mt-package

sst-package fxr-package srtp-package

MGCP Digit Map matching order: shortest match

SGCP Digit Map matching order: always left-to-right

MGCP VoAAL2 ignore-lco-codec DISABLED

MGCP T.38 Fax is ENABLED

MGCP T.38 Fax ECM is ENABLED

MGCP T.38 Fax NSF Override is DISABLED

MGCP T.38 Fax Low Speed Redundancy: 0MGCP T.38 Fax High Speed Redundancy: 0

MGCP control bound to interface FastEthernet0/0

MGCP media bind :DISABLED

MGCP Upspeed payload type for G711ulaw: 0,  G711alaw: 8

MGCP Dynamic payload type for G.726-16K codec

MGCP Dynamic payload type for G.726-24K codec

MGCP Dynamic payload type for G.Clear codec

Step 2 show mgcp connection

Use this command to display information on active connections, including the encryption suite.

Example:

Router# show mgcp connection

Endpoint        Call_ID(C) Conn_ID(I) (P)ort (M)ode (S)tate (CO)dec (E)vent[SIFL] (R)esult[EA] Encryption(K)

The following line shows that encryption status is enabled, K=1.

Example:

1. S1/DS1-0/1   C=2,1,2  I=0x2  P=18204,0  M=2  S=4,4 CO=1 E=0,0,0,0  R=0,0 K=1

Step 3 show mgcp srtp {summary| detail [endpoint]}

Use this command to display SRTP connections and validate master keys and salts for endpoints.

Example:

Router# show mgcp srtp summary

MGCP SRTP Connection Summary

Endpoint             Conn Id    Crypto Suite

aaln/S3/SU0/0        8          AES_CM_128_HMAC_SHA1_32

aaln/S3/SU0/1        9          AES_CM_128_HMAC_SHA1_32

S3/DS1-0/1           6          AES_CM_128_HMAC_SHA1_32

S3/DS1-0/2           7          AES_CM_128_HMAC_SHA1_32

4 SRTP connections active

Router# show mgcp srtp detail

MGCP SRTP Connection Detail for Endpoint *

Definitions: CS=Crypto Suite, KS=HASHED Master Key/Salt, SSRC=Syncronization Source, ROC=Rollover Counter, KDR=Key Derivation Rate, SEQ=Sequence Number, FEC=FEC Order, MLT=Master Key Lifetime, MKI=Master Key Index:MKI Size

Endpoint aaln/S3/SU0/0 Call ID 2 Conn ID 8

Tx:CS=AES_CM_128_HMAC_SHA1_32 KS=3NaOYXS9dLoYDaBHpzRejREfhf0= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Rx:CS=AES_CM_128_HMAC_SHA1_32 KS=llYCQoqxtxtdf7ECe+x+DK+G9v4= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Endpoint aaln/S3/SU0/1 Call ID 101 Conn ID 9

Tx:CS=AES_CM_128_HMAC_SHA1_32 KS=llYCQoqxtxtdf7ECe+x+DK+G9v4= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Rx:Not Configured

Endpoint S3/DS1-0/1 Call ID 1 Conn ID 6

Tx:CS=AES_CM_128_HMAC_SHA1_32 KS=3NaOYXS9dLoYDaBHpzRejREfhf0= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Rx:CS=AES_CM_128_HMAC_SHA1_32 KS=llYCQoqxtxtdf7ECe+x+DK+G9v4= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Endpoint S3/DS1-0/2 Call ID 100 Conn ID 7

Tx:CS=AES_CM_128_HMAC_SHA1_32 KS=llYCQoqxtxtdf7ECe+x+DK+G9v4= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Rx:Not Configured

4 SRTP connections displayed

Router# show mgcp srtp detail S3/DS1-0/*

MGCP SRTP Connection Detail for Endpoint S3/DS1-0/*

Definitions: CS=Crypto Suite, KS=HASHED Master Key/Salt, SSRC=Syncronization Source, ROC=Rollover Counter, KDR=Key Derivation Rate, SEQ=Sequence Number, FEC=FEC Order, MLT=Master Key Lifetime, MKI=Master Key Index:MKI Size

The following lines allow you to compare and validate a hashed version of the master key and salt, as indicated by the KS field, without the display revealing the actual master key and salt.

Example:

Endpoint S3/DS1-0/1 Call ID 1 Conn ID 6

Tx:CS=AES_CM_128_HMAC_SHA1_32 KS=3NaOYXS9dLoYDaBHpzRejREfhf0= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Rx:CS=AES_CM_128_HMAC_SHA1_32 KS=llYCQoqxtxtdf7ECe+x+DK+G9v4= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Endpoint S3/DS1-0/2 Call ID 100 Conn ID 7

Tx:CS=AES_CM_128_HMAC_SHA1_32 KS=llYCQoqxtxtdf7ECe+x+DK+G9v4= SSRC=Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=0x80000000 MKI=0:0

Rx:Not Configured

2 SRTP connections displayed

Step 4 show mgcp statistics

Use this command to display statistics, including dropped packets from unconfigured call agents.

Example:

Router# show mgcp statistics

UDP pkts rx 0, tx 0

Unrecognized rx pkts 0, MGCP message parsing errors 0

Duplicate MGCP ack tx 0, Invalid versions count 0

The following line shows the number of dropped packets from unconfigured call agents.

Example:

rx pkts from unknown Call Agent 0

CreateConn rx 0, successful 0, failed 0

DeleteConn rx 0, successful 0, failed 0

ModifyConn rx 0, successful 0, failed 0

DeleteConn tx 0, successful 0, failed 0

NotifyRequest rx 0, successful 0, failed 0

AuditConnection rx 0, successful 0, failed 0

AuditEndpoint rx 0, successful 0, failed 0

RestartInProgress tx 0, successful 0, failed 0

Notify tx 0, successful 0, failed 0

ACK tx 0, NACK tx 0

ACK rx 0, NACK rx 0

IP address based Call Agents statistics:

No Call Agent message.

System resource check is DISABLED. No available statistic

Step 5 show call active voice

Use this command to display encryption statistics.

Example:

Router# show call active voice

GENERIC: SetupTime=21072 Index=0 PeerAddress= PeerSubAddress= PeerId=0

PeerIfIndex=0 LogicalIfIndex=0 ConnectTime=0 CallState=3 CallSecurity = On CallOrigin=2 ChargedUnits=0

InfoType=0 TransmitPackets=375413 TransmitBytes=7508260 ReceivePackets=377734

ReceiveBytes=7554680

VOIP: ConnectionId[0x19BDF910 0xAF500007 0x0 0x58ED0] RemoteIPAddress=17635075

RemoteUDPPort=16394 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1

SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=600

GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=110

LoWaterPlayoutDelay=64 ReceiveDelay=94 VADEnable=0 CoderTypeRate=0

GENERIC: SetupTime=21072 Index=1 PeerAddress=+14085271001 PeerSubAddress=

PeerId=0 PeerIfIndex=0 LogicalIfIndex=5 ConnectTime=21115 CallState=4 CallOrigin=1

ChargedUnits=0 InfoType=1 TransmitPackets=377915 TransmitBytes=7558300

ReceivePackets=375594 ReceiveBytes=7511880 TotalPacketsEncrypted=375594

The following lines show statistics for encrypted and decrypted packets.

Example:

TotalPacketsDecrypted=375594 DecryptionFailurePacketCount=0 TotalPacketsAuthenticated=375594 AuthenticationFailurePacketCount=0 DuplicateReplayPacketCount=0 OutsideWindowReplayPacketCount=0

TELE: ConnectionId=[0x19BDF910 0xAF500007 0x0 0x58ED0] TxDuration=16640

VoiceTxDuration=16640 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=4

OutSignalLevel=-440 InSignalLevel=-440 InfoActivity=2 ERLLevel=227

SessionTarget=

Step 6 show voice call port

Use this command to display SRTP statistics.

Example:

Router# show voice call 1/0/0

1/0/0

vtsp level 0 state = S_CONNECTvpm level 1 state = FXSLS_CONNECT

vpm level 0 state = S_UP

calling number , calling name unavailable, calling time 01/08 03:44

c3745_13#       ***DSP VOICE TX STATISTICS***

Tx Vox/Fax Pkts: 108616, Tx Sig Pkts: 0, Tx Comfort Pkts: 0

Tx Dur(ms): 2172320, Tx Vox Dur(ms): 2172320, Tx Fax Dur(ms): 0

***DSP VOICE RX STATISTICS***

Rx Vox/Fax Pkts: 108602, Rx Signal Pkts: 0, Rx Comfort Pkts: 0

Rx Dur(ms): 2172320, Rx Vox Dur(ms): 2171990, Rx Fax Dur(ms): 0

Rx Non-seq Pkts: 3, Rx Bad Hdr Pkts: 0

Rx Early Pkts: 0, Rx Late Pkts: 0

***DSP VOICE VP_DELAY STATISTICS***

Clk Offset(ms): -2819596, Rx Delay Est(ms): 65

Rx Delay Lo Water Mark(ms): 65, Rx Delay Hi Water Mark(ms): 65

***DSP VOICE VP_ERROR STATISTICS***

Predict Conceal(ms): 250, Interpolate Conceal(ms): 0

Silence Conceal(ms): 0, Retroact Mem Update(ms): 0

Buf Overflow Discard(ms): 0, Talkspurt Endpoint Detect Err: 0

***DSP LEVELS***

TDM Bus Levels(dBm0): Rx -37.7 from PBX/Phone, Tx -35.5 to PBX/Phone

TDM ACOM Levels(dBm0): +5.0, TDM ERL Level(dBm0): +5.0

TDM Bgd Levels(dBm0): -35.9, with activity being silence

***DSP VOICE ERROR STATISTICS***

Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0

***DSP VOICE SRTP STATISTICS***

The following lines show voice SRTP statistics.

Example:

*Jan  8 2004 04:21:01.743 PAT: TotalPacketsEncrypted: 108616  TotalPacketsDecrypted: 108602

DecryptionFailurePacketCount: 0  TotalPacketsAuthenticated: 108602

AuthenticationFailurePacketCount: 0 DuplicateReplayPacketCount: 0

OutsideWindowReplayPacketCount: 0  packetsBadReceivedSSRC: 0

Note

When a T.38 fax call (nonsecure) is attempted and the fax call goes through, then switches back to secure voice (SRTP) mode, output for the show voice call port command displays an authentication failure packet count of 20. This is a normal occurrence and should not affect voice quality. The authentication failure packet count occurs because the gateways do not switch back to secure voice at the same time; that is, one side of the call is in SRTP voice mode for a short period of time while the other side is in T.38 fax mode.

Example:

Step 7 show voice call status

Use this command to display status of all voice ports.

Example:

Router# show voice call status

CallID     CID  ccVdb      Port      DSP/Ch  Called #   Codec    Dial-peers

0x5        11DE 0x660B24D0 1/0/0     1/1                g711ulaw 999100/0

0x7        11E1 0x665031A8 1/0:23.-1 1/2    *           g729ar8  0/999

0x11       11E4 0x6652B3B4 1/1:1.1   1/3     232222     g729ar8  999/0

3 active calls found

Step 8 show voice call status call-id

Use this command to display status of a specific call.

Example:

Router# show voice call status 5

Gathering information (10 seconds)...

CallID     Port      DSP/Ch  Codec    Rx/Tx     En/De     ERL/Reflctr Jitter

0x5        1/0/0     1/1     g711ulaw 500/500   500/500   5.0/3       65/0

Router# show voice call status 7

Gathering information (10 seconds)...

CallID     Port      DSP/Ch  Codec    Rx/Tx     En/De     ERL/Reflctr Jitter

0x7        1/0:23.-1 1/2     g729ar8  500/500   500/500   6.0/4       70/0

Router# show voice call status 11

Gathering information (10 seconds)...

CallID     Port      DSP/Ch  Codec    Rx/Tx     En/De     ERL/Reflctr Jitter

0x11       1/1:1.1   1/3     g729ar8  500/500   500/500   7.0/4       70/0

Step 9 show voice dsp

Use this command to display the status of DSP voice channels.

Example:

Router# show voice dsp

DSP   DSP        DSPWARE  CURR   BOOT                   PAK TX/RX

TYPE NUM CH CODEC    VERSION  STATE STATE   RST AI VOICEPORT TS ABORT PACK COUNT

==== === == ======== ======= ===== ======= === == ======== === ==== ===========

C549 1   01 {medium} 4.4.3   IDLE   idle       0   0   1/0:0       1       0     9357/9775

C549 1   02 {medium} 4.4.3   IDLE   idle             0   1/0:0        2     0     0/0

C549 2   01 {medium} 4.4.3   IDLE   idle       0   0   1/0:0       3       0     0/0

C549 2   02 {medium} 4.4.3   IDLE   idle             0   1/0:0      4   0     0/0

C549 3   01 {medium} 4.4.3   IDLE   idle       0   0   1/0:0       5       0         0/13

C549 3   02 {medium} 4.4.3   IDLE   idle             0   1/0:0       6       0     0/13

Step 10 show rtpspi call

Use this command to display active SRTP call details.

Example:

Router# show rtpspi call

RTP Service Provider info:

No. CallId dstCallId Mode      LocalRTP RmtRTP LocalIP    RemoteIP   SRTP

1   6      5         Snd-Rcv   18662    19392  0xA0A0A0D  0xA0A0A0B    1

2   8      7         Snd-Rcv   18940    16994  0xA0A0A0D  0xA0A0A0B    1

3   16     17        Snd-Rcv   19038    17198  0xA0A0A0D  0xA0A0A0B    1

Step 11 show rtpspi statistics

Use this command to display RTP statistics.

Example:

Router# show rtpspi statistics

RTP Statistics info:

No. CallId     Xmit-pkts Xmit-bytes Rcvd-pkts  Rcvd-bytes Lost pkts  Jitter Late

nc

1   6          0x842C    0x54AC30   0x842A     0x54AAE8   0x0        0x41     0x2

2   8          0x52B8    0x7C140    0x52B5     0x7C0F8    0x0        0x46     0x2

3   16         0x2EB0    0x46080    0x2EAF     0x46068    0x0        0x46     0x2

Step 12 show ccm-manager

Use this command to display the status and availability of Cisco CallManager.

Example:

Router# show ccm-manager

MGCP Domain Name: router

Priority        Status                   Host

============================================================

Primary         Registered               10.10.10.130

First Backup    Duplicate of Primary     10.10.10.130

Second Backup   None

Current active Call Manager:    10.10.10.130

Backhaul/Redundant link port:   2428

Failover Interval:              30 seconds

Keepalive Interval:             15 seconds

Last keepalive sent:            04:06:40 PAT Jan 8 2004 (elapsed time: 00:00:04)

Last MGCP traffic time:         04:06:40 PAT Jan 8 2004 (elapsed time: 00:00:04)

Last failover time:             None

Last switchback time:           None

Switchback mode:                Graceful

MGCP Fallback mode:             Enabled/OFF

Last MGCP Fallback start time:  03:42:25 PAT Jan 8 2004

Last MGCP Fallback end time:    03:42:44 PAT Jan 8 2004

MGCP Download Tones:            Disabled

Backhaul Link info:

Link Protocol:      TCP

Remote Port Number: 2428

Remote IP Address:  10.10.10.130

Current Link State: OPEN

Statistics:

Packets recvd:   7

Recv failures:   0

Packets xmitted: 13

Xmit failures:   0

PRI Ports being backhauled:

Slot 1, port 0

Configuration Error History:

FAX mode: cisco

Step 13 show crypto engine accelerator statistic

Use this command to display statistics and error counters for the onboard hardware accelerator of the router for IPsec encryption.

Example:

Router# show crypto engine accelerator statistic

Virtual Private Network (VPN) Module in slot : 0

Statistics for Hardware VPN Module since the last clear

of counters 1814 seconds ago

638 packets in                         638 packets out

88640 bytes in                         87601 bytes out

0 paks/sec in                          0 paks/sec out

0 Kbits/sec in                         0 Kbits/sec out

315 packets decrypted                  323 packets encrypted

37680 bytes before decrypt             49921 bytes encrypted

21104 bytes decrypted                  67536 bytes after encrypt

0 packets decompressed                 0 packets compressed

0 bytes before decomp                  0 bytes before comp

0 bytes after decomp                   0 bytes after comp

0 packets bypass decompr               0 packets bypass compres

0 bytes bypass decompres               0 bytes bypass compressi

0 packets not decompress               0 packets not compressed

0 bytes not decompressed               0 bytes not compressed

1.0:1 compression ratio                1.0:1 overall

33 commands out                        33 commands acknowledged

Last 5 minutes:

60 packets in                          60 packets out

0 paks/sec in                          0 paks/sec out

121 bits/sec in                        120 bits/sec out

1720 bytes decrypted                   1140 bytes encrypted

46 Kbits/sec decrypted                 30 Kbits/sec encrypted

1.0:1 compression ratio                1.0:1 overall

Errors:

ppq full errors         :        0   ppq rx errors           : 0

cmdq full errors        :        0   cmdq rx errors          : 0

no buffer               :        0   replay errors           : 0

dest overflow           :        0   authentication errors   : 0

Other error             :        0   RNG self test fail      : 0

DF Bit set              :        0   Hash Miscompare         : 0

Unwrappable object      :        0   Missing attribute       : 0

Invalid attrribute value:        0   Bad Attribute           : 0

Verification Fail       :        0   Decrypt Failure         : 0

Invalid Packet          :        0   Invalid Key             : 0

Input Overrun           :        0   Input Underrun          : 0

Output buffer overrun   :        0   Bad handle value        : 0

Invalid parameter       :        0   Bad function code       : 0

Out of handles          :        0   Access denied           : 0

Warnings:

sessions_expired        :        0   packets_fragmented      : 0

general:                :        0

HSP details:

hsp_operations          :        0   hsp_sessions            : 0

Step 14 show crypto ipsec sa

Use this command to display the settings used by current SAs.

Example:

Router# show crypto ipsec sa

interface: FastEthernet0/0

Crypto map tag: Gateway, local addr. 10.10.10.13

protected vrf:

local  ident (addr/mask/port/port): (10.10.10.13/255.255.255.255/0/0)

remote ident (addr/mask/port/port): (10.10.10.130/255.255.255.255/0/0)

current_peer: 10.10.10.130:500

PERMIT, flags={origin_is_acl,}

#pkts encaps: 324, #pkts encrypt: 324, #pkts digest: 324

#pkts decaps: 316, #pkts decrypt: 316, #pkts verify: 316

#pkts compressed: 0, #pkts decompressed: 0

#pkts not compressed: 0, #pkts compr. failed: 0

#pkts not decompressed: 0, #pkts decompress failed: 0

#send errors 71, #recv errors 0

local crypto endpt.: 10.10.10.13, remote crypto endpt.: 10.10.10.130

path mtu 1500, media mtu 1500

current outbound spi: 9073D35

inbound esp sas:

spi: 0x9FCB508(167556360)

transform: esp-3des esp-sha-hmac ,

in use settings ={Tunnel, }

slot: 0, conn id: 5121, flow_id: 1, crypto map: gateway

crypto engine type: Hardware, engine_id: 2

sa timing: remaining key lifetime (k/sec): (4446388/1913)

ike_cookies: 6A391EE1 E57F3670 D4D78758 2F5C8E7C

IV size: 8 bytes

replay detection support: Y

spi: 0xD132AE54(3509759572)

transform: esp-3des esp-sha-hmac ,

in use settings ={Tunnel, }

slot: 0, conn id: 5123, flow_id: 3, crypto map: gateway

crypto engine type: Hardware, engine_id: 2

sa timing: remaining key lifetime (k/sec): (4402107/1913)

ike_cookies: 6A391EE1 E57F3670 D4D78758 2F5C8E7C

IV size: 8 bytes

replay detection support: Y

inbound ah sas:

inbound pcp sas:

outbound esp sas:

spi: 0x7D078A45(2097646149)

transform: esp-3des esp-sha-hmac ,

in use settings ={Tunnel, }

slot: 0, conn id: 5122, flow_id: 2, crypto map: gateway

crypto engine type: Hardware, engine_id: 2

sa timing: remaining key lifetime (k/sec): (4446388/1911)

ike_cookies: 6A391EE1 E57F3670 D4D78758 2F5C8E7C

IV size: 8 bytes

replay detection support: Y

spi: 0x9073D35(151469365)

transform: esp-3des esp-sha-hmac ,

in use settings ={Tunnel, }

slot: 0, conn id: 5124, flow_id: 4, crypto map: gateway

crypto engine type: Hardware, engine_id: 2

sa timing: remaining key lifetime (k/sec): (4402077/1911)

ike_cookies: 6A391EE1 E57F3670 D4D78758 2F5C8E7C

IV size: 8 bytes

replay detection support: Y

outbound ah sas:

outbound pcp sas:

protected vrf:

local  ident (addr/mask/prot/port): (10.10.10.13/255.255.255.255/0/0)

remote ident (addr/mask/prot/port): (10.10.10.131/255.255.255.255/0/0)

current_peer: 10.10.10.131:500

PERMIT, flags={origin_is_acl,}

#pkts encaps: 0, #pkts encrypt: 0, #pkts digest: 0

#pkts decaps: 0, #pkts decrypt: 0, #pkts verify: 0

#pkts compressed: 0, #pkts decompressed: 0

#pkts not compressed: 0, #pkts compr. failed: 0

#pkts not decompressed: 0, #pkts decompress failed: 0

#send errors 0, #recv errors 0

local crypto endpt.: 10.10.10.13, remote crypto endpt.: 10.10.10.131

path mtu 1500, media mtu 1500

current outbound spi: 0

inbound esp sas:

inbound ah sas:

inbound pcp sas:

outbound esp sas:

outbound ah sas:

outbound pcp sas:

Step 15 show crypto isakmp sa

Use this command to display current IKE SAs at a peer.

Example:

Router# show crypto isakmp sa

dst             src             state          conn-id slot

10.10.10.130    10.10.10.13     QM_IDLE              1    0

Step 16 show crypto session

Use this command to display the status of the current crypto session.

Example:

Router# show crypto session

Crypto session current status

Interface: FastEthernet0/0

Session status: UP-ACTIVE

Peer: 10.10.10.130/500

IKE SA: local 10.10.10.13/500 remote 10.10.10.130/500 Active

IPSEC FLOW: permit ip host 10.10.10.13 host 10.10.10.130

Active SAs: 4, origin: crypto map

Step 17 show crypto session detail

Use this command to display IPsec details and statistics of the current crypto session.

Example:

Router# show crypto session detail

Crypto session current status

Code: C - IKE Configuration mode, D - Dead Peer Detection

K - Keepalives, N - NAT-traversal, X - IKE Extended Authentication

Interface: FastEthernet0/0

Session status: UP-ACTIVE

Peer: 10.10.10.130/500 fvrf: (none) ivrf: (none)

Phase1_id: 10.10.10.130

Desc: (none)

IKE SA: local 10.10.10.13/500 remote 10.10.10.130/500 Active

Capabilities:(none) connid:1 lifetime:07:30:00

IPSEC FLOW: permit ip host 10.10.10.13 host 10.10.10.130

Active SAs: 4, origin: crypto map

Inbound:  #pkts dec'ed 335 drop 0 life (KB/Sec) 4402106/1800

Outbound: #pkts enc'ed 327 drop 71 life (KB/Sec) 4402076/180

Configuration Examples for Media and Signaling Authentication and Encryption

- Voice Security Features Example

Voice Security Features Example

The following example shows voice security features enabled:

Router# show running-config

Building configuration...

Current configuration : 2304 bytes

!

version 12.3

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname router

!

boot-start-marker

boot-end-marker

!

voice-card 1

no dspfarm

!

voice-card 2

no dspfarm

!

The following lines show secure codec complexity enabled:

voice-card 4

codec complexity secure

dspfarm

!

!

no aaa new-model

ip subnet-zero

!

ip cef

no ip domain lookup

!

ip domain name cisco.com

The IP domain name should match the domain name configured on Cisco CallManager.

!

Cisco CallManager-manager mgcp

!

crypto isakmp policy 1

authentication pre-share

lifetime 28800

crypto isakmp key cisco123 address 10.1.1.12

The crypto key should match the key configured on Cisco CallManager. This method and encapsulation mode should also match the method and encapsulation mode configured on Cisco CallManager. Other methods of key exchange are also supported. For more information refer to Cisco IOS Security Configuration Guide, Release 12.3 .

!

crypto ipsec transform-set rtpset esp-des esp-md5-hmac

mode transport

The crypto IPsec configuration should match the Cisco CallManager configuration.

!

crypto map rtp 1 ipsec-isakmp

set peer 10.1.1.12

set transform-set rtpset

match address 115

!

interface FastEthernet0/1

ip address 10.1.1.212 255.255.255.0

load-interval 30

duplex auto

speed auto

crypto map rtp

!

The following line shows the IPsec access list.

access-list 115 permit ip host 10.1.1.212 host 10.1.1.12

!

voice-port 1/0/0

!

voice-port 2/0/0

!

mgcp

mgcp call-agent 10.1.1.12 service-type mgcp version 0.1

The mgcp package-capability command enables the MGCP application ability to manage SRTP calls and advertise SRTP capability in SDP sent to remote gateways.

mgcp package-capability srtp-package

!

mgcp profile default

!

dial-peer voice 100 pots

application mgcpapp

port 1/0/0

!

dial-peer voice 200 pots

application mgcpapp

port 2/0/0

!

dial-peer voice 201 pots

application mgcpapp

port 2/0/1

!

dial-peer voice 202 pots

application mgcpapp

port 2/0/2

!

dial-peer voice 203 pots

application mgcpapp

port 2/0/3

!

dial-peer voice 101 pots

application mgcpapp

port 1/0/1

!

dial-peer voice 110 pots

application mgcpapp

port 1/1/0

!

dial-peer voice 111 pots

application mgcpapp

port 1/1/1

!

!

alias exec k show mgcp conn | inc K=

alias exec sr sh call active voi | inc SRTP

alias exec rs sh rtpspi call | inc Snd-Rcv

alias exec vc sh voi call

alias exec m sh mgcp conn

alias exec cav sh call active voi

alias exec rsa sh rtpspi call

alias exec cc clear counters

alias exec sta sh int fa0/1 stat

alias exec cef sh ip cef

!

line con 0

exec-timeout 0 0

logging synchronous

line aux 0

line vty 0 4

login

!

!

end

Additional References

The following sections provide references related to the Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways feature.

Related Documents

Related Topic

Document Title

Cisco CallManager configuration

Cisco IP Phone Authentication and Encryption for Cisco CallManager 4.0(1)

Cisco CallManager and IPsec configuration

- " How to Configure IPsec Tunneling in Windows 2000 ," Microsoft Knowledge Base article

- " Step-by-Step Guide to Internet Protocol Security (IPsec) ," "Building A Custom IPsec Policy" section, Microsoft Knowledge Base article

Cisco IP Phone 7940 and 7960 administration

Cisco IP Phone Model 7960G and 7940G Administration Guide for Cisco CallManager

Cisco IP Phone 7970 administration

Cisco IP Phone 7970 Administration Guide for Cisco CallManager

Cisco 2621 configuration

Cisco 2621 Modular Access Router with AIM-VPN/BP Security Policy

Cisco 2651 configuration

Cisco 2651 Modular Access Router with AIM-VPN/BP Security Policy

Cisco 3640 configuration

Cisco 3640 Modular Access Router with AIM-VPN/BP Security Policy

Cisco 3660 configuration

Cisco 3660 Modular Access Router with AIM-VPN/BP Security Policy

Secure SRST router configuration

http://www.cisco.com/univercd/cc/td/doc/product/voice/srst/srst33/srst33ad/sr_scur1.htm Setting Up Secure SRST

Advanced Encryption Standard (AES) feature

Advanced Encryption Standard

IPsec configuration

Cisco IOS Security Configuration Guide, Release 12.3

IPsec commands

Cisco IOS Security Command Reference, Release 12.3

MGCP configuration

MGCP and Related Protocols Configuration Guide

Cisco IOS voice configuration

Cisco IOS Voice Configuration Library

Cisco IOS voice command reference

http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_r/index.htm Cisco IOS Voice, Video, and Fax Command Reference, Release 12.3T

Configuring IPsec Between a Microsoft Windows 2000 Server and a Cisco Device

Configuring IPsec Between a Microsoft Windows 2000 Server and a Cisco Device

Secure Real-time Transport Protocol Support

Secure Real-time Transport Protocol Support

Standards

Standards

Title

IETF draft draft-ietf-mmusic-sdescriptions-02.txt

Security Descriptions for Media Streams

MIBs

MIB

MIBs Link

CISCO-VOICE-DIAL-CONTROL-MIB

To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://www.cisco.com/go/mibs

RFCs

RFC

Title

RFC 3711

Secure Real-time Transport Protocol

RFC 4040

RTP Payload Format for a 64 kbit/s Transparent Call

RFC 4568

Session Description Protocol (SDP) Security Descriptions for Media Streams

Technical Assistance

Description

Link

The Cisco Support website provides extensive online resources, including documentation and tools for troubleshooting and resolving technical issues with Cisco products and technologies.

To receive security and technical information about your products, you can subscribe to various services, such as the Product Alert Tool (accessed from Field Notices), the Cisco Technical Services Newsletter, and Really Simple Syndication (RSS) Feeds.

Access to most tools on the Cisco Support website requires a Cisco.com user ID and password.

http://www.cisco.com/techsupport

Feature Information for Media and Signaling Authentication and Encryption

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Table 8

Feature Information for Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways

Feature Name

Releases

Feature Information

Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways

12.3(11)T2 12.3(14)T

In 12.3(11)T2, this feature was introduced.

In 12.3(14)T support was added for the Cisco Secure SRST feature and the NM-HDV network module.

Support for MGCP 1.0 Call Control for SRTP on Cisco IOS Gateways

15.0(1)XA

This feature provides support for MGCP 1.0 call control for SRTP on Cisco IOS gateways, and for fax pass-through and the Clear Channel codec at the media level under MGCP 1.0 and 0.1.

The following command was introduced: mgcp crypto rfc-preferred.

Glossary

CCM --Cisco Call Manager.

CLI --command-line interface.

CTL --Certificate Trust List.

DTMF --dual-tone multifrequency

HMAC --Hashed Message Authentication Codes.

IETF --Internet Engineering Task Force. Standards body for Internet standards.

IKE --Internet Key Exchange.

IPsec --IP security.

MGCP --Multimedia Gateway Control Protocol.

PIN --Personal identification number.

RTCP --Real-Time Transport Protocol Control Protocol.

RTP --Real-Time Transport Protocol

SDP --Session Description Protocol.

SHA1 --Secure Hash Algorithm1.

SRST --Survivable Remote Site Telephony.

SRTP --Secure RTP.

SRTCP --Secure RTCP.

VoIP --Voice over IP.

Configuring MGCP CAS PBX and AAL2 PVC

This section provides information on configuring the MGCP Channel-Associated Signaling (CAS) Private-Branch-Exchange (PBX) and ATM Adaptation Layer 2 (AAL2) Permanent Virtual Circuit (PVC) feature.

Feature benefits include the following:

- The merged Simple Gateway Control Protocol/Media Gateway Control Protocol (SGCP/MGCP) software for residential gateways (RGWs), business gateways (BGWs), and trunking gateways (TGWs) enables easier development and growth of Cisco and customer solutions.

- MGCP CAS PBX and AAL2 PVC software meets customer requirements for CAS connectivity to traditional PBXs and regulatory requirements for support of 911, Barge In, and Busy Line Verify features.

For more information about this and related Cisco IOS voice features, see the following:

- "Overview of MGCP and Related Protocols" on page 3

- Entire Cisco IOS Voice Configuration Library--including library preface and glossary, other feature documents, and troubleshooting documentation--at http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Feature History for MGCP CAS PBX and AAL2 PVC

Release

Modification

12.1(5)XM

This feature was introduced on the following platforms: Cisco 1750, Cisco 2600 series, Cisco 3600 series, Cisco AS5300, Cisco MC3810, and Cisco uBR924.

12.2(2)T

This feature was integrated into this release on all previously supported platforms except the Cisco AS5300. A new command was added (mgcp rtp unreachable timeout) and an existing command was modified (mgcp sdp).

12.2(11)T

This feature was implemented on the Cisco AS5300 and Cisco AS5850.

Note

AAL2 PVC is not supported on the Cisco AS5850.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for MGCP CAS PBX and AAL2 PVC

Prerequisites are described in the "Prerequisites for Configuring MGCP and Related Protocols" section on page 3 .

Restrictions for MGCP CAS PBX and AAL2 PVC

MGCP CAS PBX and AAL2 PVC Software Caveats

- Only the Cisco MC3810 series platform supports MGCP 0.1 control of AAL2 voice transport in this Cisco IOS release.

- For the Cisco MC3810 series platform, the AAL2 PVC functionality is supported on an high-performance compression module (HCM) version of an digital signal processor (DSP) card; it is not supported on an voice compression module (VCM) version.

To check the type of DSP card, enter a show version command at the EXEC prompt.

- If you have an HCM card, the following line appears as part of the show versioninformation:

1 6-DSP (slot 2) High Performance Compression Module(v01.A0)

- If you have an VCM card, the following line appears as part of the show versioninformation:

1 6-DSP (slot 2) Voice Compression Module(v255.V7)

If you have an HCM card, the MGCP Basic CLASS and Operator Services (BCOS) features will function correctly. If you have an VCM card, the AAL2 PVC feature is not supported.

- The Cisco AS5300 multiservice platform supports only the Feature Group-D Operator Services (FGD-OS) Barge-In/Busy Line Verify and 911 features of the MGCP CAS PBX and AAL2 PVC software.

Features Not Supported

- Basic CLASS and Operator features are covered in the MGCP Basic CLASS and Operator Services software. For more information on these capabilities, see Configuring MGCP Basic CLASS and Operator Services .

- The MGCP CAS PBX and AAL2 PVC software has not implemented DSP clock slotting changes, Comfort Noise Indication, ATM SVCs, TGCP, AAL1, FXO support in SGCP, ATM on the Cisco 3660 platform, and VoIP Call Admission Control (CAC). These capabilities are part of other Cisco development efforts.

Information About MGCP CAS PBX and AAL2 PVC

The MGCP CAS PBX and AAL2 PVC features extend the earlier Simple Gateway Control Protocol (SGCP) Channel Associated Signaling (CAS) and AAL2 support onto the merged SGCP/MGCP software base to enable various service provider solutions.

MGCP CAS PBX and AAL2 PVC Features

- CAS termination and translation to MGCP on Business Gateways (BGWs) and Trunking Gateways (TGWs).

Digital CAS (E&M) interfaces are supported in addition to the analog (FXO, FXS, and E&M) interfaces.

For this feature release, the BGWs are the Cisco 3810 series and Cisco 2600 series routers. The TGWs are the Cisco 3600 series multiservice platforms.

- Support for CAS PBX and Feature Group D CAS Functions.

MGCP 0.1 has been expanded to support CAS packages that handle CAS PBX and Feature Group D CAS functions, including Barge-In/Busy Line Verify, and 911 capabilities on the TGW.

- Expanding MGCP 0.1 to control AAL2 voice transport.

The earlier version of the merged SGCP/MGCP stack supported only Voice over IP. The merged stack will now support both VoIP and VoAAL2.

Only the Cisco MC3810 series platforms supports this feature in this release.

- Addition of SGCP CAS PBX support to the existing merged SGCP/MGCP software stack.

The CAS PBX gateway features include CAS PBX trunks, digit maps, CAS events, and quarantine buffer software. These features were available in the existing standalone SGCP software; now they are supported in the merged stack.

- Consolidation of various SGCP and MGCP feature sets onto one software image for Residential Gateways (RGWs), BGWs, and TGWs.

For this feature release, the RGWs are the Cisco uBR924 cable router and Cisco 1750 access router.

The essential difference for current SGCP users is that support for the SGCP application has been replaced with the MGCP application. The MGCP application supports both SGCP commands and MGCP commands, permitting access to a larger feature set than with the SGCP application alone. The MGCP CAS PBX and AAL2 PVC software assumes the MGCP mode as the default environment. This allows the gateway to recognize both MGCP and SGCP messages sent by the call agent. However, the user can specify SGCP mode for certain messages that will be sent by the gateway (by using the sgcp option as the service type in the mgcp call-agent command).

Examples of Service Provider Solutions

MGCP CAS PBX and AAL2 PVC features support several types of service provider solutions:

- Residential cable access

A CLEC can use residential cable access to provide residential customers with basic telephony and data services. The figure below illustrates a possible residential cable access solution:

Figure 13

Residential Cable Access Solution

Note that in the figure above, the trunking gateway (the Cisco 3660 platform) requires support of incoming and outgoing MF signaling for the Barge-In and Busy-Line Verify features. The residential gateway (the Cisco uBR924 cable access router) must support the CLASS features and 911 capability.

- IP Centrex and IP PBX

In these solutions, a call agent provides business voice services traditionally offered by a circuit-based PBX. The figure below illustrates an IP Centrex solution:

Figure 14

IP Centrex Solution

In the figure above, the BGW (the Cisco 2600 platform) requires PBX connectivity to interface with the legacy PBX.

- Integrated Access

A CLEC or IXC can provide small, medium, and large businesses with integrated voice and data access services. The integrated access device can be located at the central office or on the customer's premises. Access to the subscriber can be analog or digital T1 interfaces in addition to DSL. Transport of voice and data can be over IP, Frame Relay, or ATM. The figure below illustrates an integrated access solution:

Figure 15

Integrated Access Solution

In the figure above, MGCP control of calls over the AAL2 PVCs is required on the BGWs (the Cisco 2600 series, Cisco 3600, and Cisco 3810 series platforms) to connect into the ATM network for VToA.

- Telecommuter or Small Office-Home Office

The figure below illustrates a telecommuter/small office-home office solution:

Figure 16

Telecommuter or Small Office-Home Office Solution

In the figure above, MGCP must control the calls over AAL2 PVCs, and an analog FXS interface is required.

How to Configure MGCP CAS PBX and AAL2 PVC

Some tasks indicate one or more configuration examples affected by the command. See the specific configuration example listing for the parameter values.

Configuring the Gateway

SUMMARY STEPS

1.   mgcp

2.   mgcp call-agent {ipaddr | hostname} [port] [service-type type] version version-number

3.   mgcp dtmf-relay voip codec {all | low-bit-rate} mode {cisco | nse | out-of-band}

4.   mgcp package-capability {as-package | atm-package | dtmf-package | gm-package | hs-package | nas-package | rtp-package | script-package | trunk-package}

5.   mgcp sgcp restart notify

6.   mgcp modem passthrough [voip | voaal2] mode [cisco | nse]

7.   mgcp tse payload type

8.   mgcp rtp unreachable timeout timer-value

9.   no mgcp timer receive-rtcp

10.   mgcp timer net-cont-test timer

11.   controller T1 0

12.   mode atm

13.   no shutdown

14.   exit

15.   mgcp quarantine mode process

16.   controller T1 1

17.   mode cas

18.   ds0-group channel-number timeslots range type signaling-type tone type addr info service service-type

19.   exit

20.   interface atm0 [subinterface-number [multipoint | point-to-point]]

21.   pvc [name] vpi/vci

22.   encapsulation aal-encap

23.   vbr-rt peak-rate average-rate [burst]

24.   vcci pvc-identifier

25.   exit

26.   exit

27.   dial-peer voice number pots

28.   application MGCPAPP

29.   exit

DETAILED STEPS

Step 1 mgcp

Starts the MGCP daemon.

Example:

Router(config)# mgcp

Step 2 mgcp call-agent {ipaddr | hostname} [port] [service-type type] version version-number

Configures the MGCP call agent and service type. If you want SGCP mode, use sgcp as the service type.

Example:

Router(config)# mgcp call-agent {ipaddr | hostname } [port ] [service-type type ] version version-number

Step 3 mgcp dtmf-relay voip codec {all | low-bit-rate} mode {cisco | nse | out-of-band}

(Optional. See Configuration Example 2.) Specifies compressed codecs for digit forwarding.

Example:

Router(config)# mgcp dtmf-relay voip codec {all | low-bit-rate} mode {cisco | nse | out-of-band}

Step 4 mgcp package-capability {as-package | atm-package | dtmf-package | gm-package | hs-package | nas-package | rtp-package | script-package | trunk-package}

(Optional. See Configuration Example 2.) Assigns packages to the gateway. Also refer to the mgcp default-package command.

Example:

Router(config)# mgcp package-capability {as-package | atm-package | dtmf-package | gm-package | hs-package | nas-package | rtp-package | script-package | trunk-package}

Step 5 mgcp sgcp restart notify

(Required only for SGCP mode with a call agent supporting RSIP. See Configuration Examples 4 through 9.) Causes MGCP to send SGCP RSIP messages.

Example:

Router(config-if)# mgcp sgcp restart notify

Step 6 mgcp modem passthrough [voip | voaal2] mode [cisco | nse]

(Optional for nse mode) Enables the gateway to process fax or modem messages. VoAAL2 does not support cisco.

Example:

Router(config-if)# mgcp modem passthrough [voip | voaal2] mode [cisco | nse]

Step 7 mgcp tse payload type

(Required for nsemode. See Step 6.) Enables the TSE payload for fax and modem messages.

Example:

Router(config)# mgcp tse payload type

Step 8 mgcp rtp unreachable timeout timer-value

(Optional) Enables detection of unreachable remote VoIP endpoints.

Example:

Router(config)# mgcp rtp unreachable timeout timer-value

Step 9 no mgcp timer receive-rtcp

(Required for non-RGWs. See Configuration Examples 2 through 9.) Turns off the RTP RTCP receive timeout interval at the gateway.

Example:

Router(config)# no mgcp timer receive-rtcp

Step 10 mgcp timer net-cont-test timer

(Optional for non-RGWs. See Configuration Examples 2 through 9.) Turns on the continuity test timeout interval at the gateway.

Example:

Router(config)# mgcp timer net-cont-test timer

Step 11 controller T1 0

(Required for ATM mode. See Configuration Examples 2 through 9.) Select s the T1 controller 0.

Example:

Router(config)# controller T1 0

Step 12 mode atm

(Required for ATM mode. See Configuration Examples 2 through 9.) Specifies that the controller will support ATM encapsulation and create ATM interface 0.

When the controller is set to ATM mode, the following takes place:

- Controller framing is automatically set to Extended SuperFrame (ESF).

- The linecode is automatically set to B8ZS.

Example:

Router(config-controller)# mode atm

Step 13 no shutdown

(Optional for ATM mode. See Configuration Examples 2 through 9.) Ensures that the controller is activated.

Example:

Router(config-controller)# no shutdown

Step 14 exit

(Required for ATM mode. See Configuration Examples 2 through 9.) Exits the current mode.

Example:

Router(config-controller)# exit

Step 15 mgcp quarantine mode process

(Optional) Turns on processing for SGCP quarantine mode.

Example:

Router(config)# mgcp quarantine mode process

Step 16 controller T1 1

(Required for CAS PBX. See Configuration Examples 3, 4, and 5.) Select the T1 controller 1.

Example:

Router(config)# controller T1 1

Step 17 mode cas

(Required for CAS PBX. See Configuration Examples 3, 4, and 5.) Specify that the controller will support CAS.

Example:

Router(config-controller)# mode cas

Step 18 ds0-group channel-number timeslots range type signaling-type tone type addr info service service-type

(Required for CAS PBX. See Configuration Examples 3, 4, and 5.) Configure the T1 timeslots for CAS calls.

Example:

Router(config-controller)# ds0-group channel-number timeslots range type signaling-type tone type addr info service service-type

Step 19 exit

(Required for CAS PBX. See Configuration Examples 3, 4, and 5.) Exit controller configuration mode.

Example:

Router(config-controller)# exit

Step 20 interface atm0 [subinterface-number [multipoint | point-to-point]]

(Required for ATM mode. See Configuration Examples 2 through 9.) Enter interface configuration mode to configure ATM interface 0 or an ATM subinterface.

Default for subinterfaces is multipoint.

For all scenarios: Set up three subinterfaces for point-to-point.

Example:

Router(config)# interface atm0 [subinterface-number [multipoint | point-to-point]]

Step 21 pvc [name] vpi/vci

(Required for ATM mode. See Configuration Examples 2 through 9.) Create an ATM PVC for voice traffic and enter ATM virtual circuit configuration mode.

Note

The ilmi and qsaaloptions are not supported for AAL2.

Example:

Router(config-if)# pvc [name ] vpi/vci

Step 22 encapsulation aal-encap

(Required for ATM mode. See Configuration Examples 2 through 9.) Set the encapsulation of the PVC for voice traffic. aal2 automatically creates channel identifiers (CIDs) 1 through 255.

Some of the Scenarios use aal5snap for ATM0.1 and ATM0.3. Use aal2 for ATM0.2.

Example:

Router(config-if-atm-vc)# encapsulation aal-encap

Step 23 vbr-rt peak-rate average-rate [burst]

(Required for ATM mode. See Configuration Examples 2 through 9.) Configures the PVC for the variable-bit-rate real-time (voice) traffic.

Example:

Router(config-if-atm-vc)# vbr-rt peak-rate average-rate [burst ]

Step 24 vcci pvc-identifier

(Optional for ATM mode. See Configuration Examples 2 through 9.) Assigns a unique identifier to the PVC.

Example:

Router(config-if-atm-vc)# vcci pvc-identifier

Step 25 exit

(Required for ATM mode. See Configuration Examples 2 through 9.) Exits the current mode.

Example:

Router(config-if-atm-vc)# exit

Step 26 exit

(Required for ATM mode. See Configuration Examples 2 through 9.) Exits the current mode.

Example:

Router(config-if)# exit

Step 27 dial-peer voice number pots

Enter dial peer configuration mode for the POTS dial peer.

Example:

Router(config)# dial-peer voice number pots

Step 28 application MGCPAPP

Initiates the MGCP protocol for the voice ports.

Example:

Router(config-dial-peer)# application MGCPAPP

Step 29 exit

Exits the current mode.

Example:

Router(config-dial-peer)# exit

Configuring Subcell Multiplexing for AAL2 Voice

This section describes the configuration tasks necessary to enable AAL2 common part sublayer (CPS) subcell multiplexing when the Cisco MC3810 series platform interoperates with a voice interface service module (VISM) in an MGX switch.

SUMMARY STEPS

1.   voice service voatm

2.   session protocol aal2

3.   subcell-mux

4.   end

DETAILED STEPS

Step 1 voice service voatm

(Required) Enters voice-service configuration mode.

Example:

Router(config)# voice service voatm

Step 2 session protocol aal2

(Required) Enters voice-service-session configuration mode and specifies AAL2 trunking.

Example:

Router(config-voice-service)# session protocol aal2

Step 3 subcell-mux

(Required) Enables subcell multiplexing.

By default, subcell multiplexing is not enabled.

Example:

Router(config-voice-service-session)# subcell-mux

Step 4 end

(Required) Exits the current mode.

Example:

Router(config-voice-service-session)# end

Configuring the Cable Access Router for SGCP and MGCP

The Cisco uBR924 cable access router requires standard per-port provisioning to work with MGCP CAS PBX and AAL2 PVC:

To access SGCP functionality, use the command:

S|0|ca1@call-agent.abc.com:2427|S|1|ca2@call-agent.abc.com:2427

To access MGCP functionality, use the command:

M|0|ca1@call-agent.abc.com:2427|M|1|ca2@call-agent.abc.com:2427

For either functionality type, port 0 points to call agent 1 and port 1 points to call agent 2. If needed, both ports can point to the same call agent.

Verifying the MGCP CAS PBX and AAL2 PVC Configurations

To verify configuration, use the following commands.

SUMMARY STEPS

1.   show dial-peer voice sum

2.   show running-configuration

DETAILED STEPS

Step 1 show dial-peer voice sum

Displays the status of the dial peer. The dial peer should be active. If it is not, use the no shut command to make it so.

Example:

Router# show dial-peer voice sum

Step 2 show running-configuration

Displays the current configuration settings.

Example:

Router# show running-configuration

Configuration Examples for MGCP CAS PBX and AAL2 PVC

- Example 1 MGCP Residential Gateway

- Example 2 MGCP Gateway using Voice over ATM AAL2

- Example 3 MGCP and SGCP EM Wink-Start

- Example 4 SGCP 1.5 CAS PBX using Voice over ATM AAL2

- Example 5 SGCP 1.5 CAS PBX using Voice over IP over ATM AAL5

- Example 6 SGCP 1.5 Analog EM PBX using Voice over ATM AAL2

- Example 7 SGCP 1.5 Analog EM PBX using Voice over IP over ATM AAL5

- Example 8 SGCP 1.5 RGW using Voice over ATM AAL2

- Example 9 SGCP 1.5 RGW using Voice over IP over ATM AAL5

Example 1 MGCP Residential Gateway

The following example illustrates the configuration for a Cisco MC3810 series platform with CAS running the MGCP application:

version 12.2

no service pad

service timestamps debug datetime msec

service timestamps log uptime

!

hostname Router

!

logging buffered

!

ip subnet-zero

ip host first 192.168.254.254

!

mgcp

mgcp call-agent 172.16.90.1

!

voice-card 0

codec complexity high

!

controller T1 0

framing esf

linecode b8zs

!

interface Ethernet0

ip address 172.16.92.3 255.255.0.0

!

interface Serial0

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface FR-ATM20

no ip address

shutdown

!

ip default-gateway 172.16.0.1

ip route 198.168.254.0 255.255.255.0 172.16.0.1

!

voice-port 1/1

!

dial-peer voice 1 pots

application MGCPAPP

port 1/1

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Example 2 MGCP Gateway using Voice over ATM AAL2

The following configuration illustrates a Cisco MC3810 series platform running the MGCP application using ATM AAL2 to carry voice traffic:

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname main.office

!

network-clock base-rate 56k

ip subnet-zero

no ip domain-lookup

ip host second 192.168.254.254

ip host first 192.168.254.253

!

mgcp

mgcp call-agent 172.16.117.4 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode nse

mgcp dtmf-relay voaal2 codec all

mgcp package-capability rtp-package

mgcp tse payload 100

mgcp timer receive-rtcp 100

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 171.16.121.1 255.255.0.0

!

interface Serial0

no ip address

no ip mroute-cache

shutdown

no fair-queue

!

interface Serial1

no ip address

shutdown

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

interface ATM0.2 point-to-point

pvc 2/200

vbr-rt 760 760 100

encapsulation aal2

vcci 2

!

interface FR-ATM20

no ip address

shutdown

!

router group1 1

redistribute connected

network 172.0.0.0

!

ip default-gateway 172.16.0.1

no ip http server

ip classless

ip route 192.168.254.0 255.255.255.0 172.16.0.1

!

dialer-list 1 protocol ip permit

dialer-list 1 protocol ipx permit

voice-port 1/1

codec g711ulaw

!

voice-port 1/2

shutdown

!

voice-port 1/6

shutdown

!

dial-peer voice 1 pots

application MGCPAPP

destination-pattern 2220001

port 1/1

!

line con 0

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Example 3 MGCP and SGCP EM Wink-Start

The following example illustrates an E&M wink-start configuration on the Cisco MC3810 series platform that can be defined for either the SGCP or MGCP modes:

version 12.2

no service pad

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname sales

!

network-clock base-rate 56k

ip subnet-zero

!

mgcp

no mgcp timer receive-rtcp

call rsvp-sync

!

voice service voatm

!

session protocol aal2

subcell-mux

!

voice-card 0

!

controller T1 0

mode atm

framing esf

clock source internal

linecode b8zs

!

controller T1 1

mode cas

framing esf

linecode b8zs

ds0-group 1 timeslots 1-24 type e&m-wink-start

!

interface Ethernet0

ip address 172.29.248.199 255.255.255.0

no ip route-cache

no ip mroute-cache

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.2 point-to-point

pvc 2/200

vbr-rt 1536 1536 100

encapsulation aal2

vcci 10

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

ip route 0.0.0.0 0.0.0.0 172.29.248.1

no ip http server

!

voice-port 1:1

dial-type mf

!

dial-peer cor custom

!

dial-peer voice 1 pots

application mgcpapp

destination-pattern 1

port 1:1

!

gatekeeper

shutdown

!

line con 0

exec-timeout 0 0

transport input none

line aux 0

line 2 3

line vty 0 4

login

length 0

!

ntp clock-period 17248569

ntp server 172.29.1.129

end

Example 4 SGCP 1.5 CAS PBX using Voice over ATM AAL2

The following figure and configuration illustrate the network connections for a Cisco MC3810 series platform with CAS running the MGCP application in SGCP 1.5 mode. ATM AAL2 carries voice traffic.

- T1/0 is configured to run ATM with three permanent virtual circuits (PVCs):

- 1 PVC with encapsulation AAL5 carries SGCP messages (signaling VC)

- 1 PVC with encapsulation AAL5 carries data traffic (data VC)

- 1 PVC with encapsulation AAL2 carries voice traffic (bearer VC)

This bearer VC has a vcci of 2 assigned to it. The service manager uses this vcci value and a selected channel identifier (CID) value for a voice call on this router.

For AAL2, allocate 200 ATM cells/sec (84.8K bits/sec) for each G711u no vad call, 100 ATM cells/sec (42.4K bits/sec) for each G726-32 no vad or G729a no vad call.

- In this configuration, T1/1 is configured as three DS-0 groups:

- 1 FXS ground start group

- 1 E&M immediate start group

- 1 E&M wink start group

For these voice ports, the dial type is set to mfto support mf dialing.

- mgcp sdp is configured to enable SGCP RSIP messages notification.

- mgcp modem passthrough mode is configured to allow nse processing of fax or modem calls.

Figure 17

SGCP 1.5 CAS PBX using Voice over ATM AAL2 Configuration

Router A Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

clock source line

linecode b8zs

!

controller T1 1

mode cas

framing esf

clock source line

linecode b8zs

ds0-group 1 timeslots 1-8 type e&m-immediate-start

ds0-group 2 timeslots 9-16 type e&m-wink-start

ds0-group 3 timeslots 17-24 type fxs-ground-start

!

interface Ethernet0

ip address 172.16.24.103 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.2 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

pvc 2/200

vbr-rt 1536 1400 100

encapsulation aal2

vcci 2

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.5 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

no ip http server

!

voice-port 1:1

!

voice-port 1:2

dial-type mf

!

voice-port 1:3

!

dial-peer voice 1 pots

application MGCPAPP

port 1:1

!

dial-peer voice 2 pots

application MGCPAPP

port 1:2

!

dial-peer voice 3 pots

application MGCPAPP

port 1:3

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Router B Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname B

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

clock source line

linecode b8zs

!

controller T1 1

mode cas

framing esf

clock source line

linecode b8zs

ds0-group 1 timeslots 1-8 type e&m-immediate-start

ds0-group 2 timeslots 9-16 type e&m-wink-start

ds0-group3 timeslots 17-24 type fxs-ground-start

!

interface Ethernet0

ip address 172.17.24.103 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.3 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

pvc 2/200

vbr-rt 1536 1400 100

encapsulation aal2

vcci 2

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.8 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

no ip http server

!

voice-port 1:1

!

voice-port 1:2

dial-type mf

!

voice-port 1:3

!

dial-peer voice 1 pots

application MGCPAPP

port 1:1

!

dial-peer voice 2 pots

application MGCPAPP

port 1:2

!

dial-peer voice 3 pots

application MGCPAPP

port 1:3

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Example 5 SGCP 1.5 CAS PBX using Voice over IP over ATM AAL5

The following figure and configuration illustrate the network connections for a Cisco MC3810 series platform with CAS running the MGCP application in SGCP 1.5 mode. Voice over IP over ATM AAL5 carries voice traffic.

This configuration is very similar to the AAL2 example in the previous section except that an AAL5 PVC is the bearer PVC for voice traffic.

This configuration has a loopback interface with an IP address assigned to it. During voice calls, the gateway gives this IP address to the service manager as the address for the other gateway of the voice connection to use as the destination IP address.

In the example below, Router A's loopback address is 192.168.1.0 and Router B's address is 192.168.5.0. If Router A originated a call to Router B, A would give 192.168.1.0 to the Service Manager and B would give 192.168.5.0. The IP route configuration commands of both routers direct the IP traffic into the voice bearer PVC since the loopback addresses are on different IP subnets.

For Voice over IP, allocate 300 ATM cells/sec (127.2K bits/sec) for each G711u no vad call, and 200 ATM cells/sec (84.8K bits/sec) for each G726-32 no vad or G729a no vad call.

Note

For G711u no vad calls, a T1 running ATM does not have enough bandwidth to carry 24 voice calls.

Figure 18

SGCP 1.5 CAS PBX using Voice over IP over ATM AAL5 Configuration

Router A Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp modem passthrough nse

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

controller T1 1

mode cas

framing esf

clock source line

linecode b8zs

ds0-group 1 timeslots 1-8 type e&m-immediate-start

ds0-group 2 timeslots 9-16 type e&m-wink-start

ds0-group 3 timeslots 17-24 type fxs-ground-start

framing esf

linecode b8zs

!

interface Loopback0

ip address 192.168.1.0 255.255.255.0

!

interface Ethernet0

ip address 172.16.24.103 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.2 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

ip address 10.0.0.5 255.0.0.0

pvc 1/2

vbr-rt 1536 1400 100

encapsulation aal5mux ip

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.8 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

ip route 10.0.0.15 255.0.0.0 ATM0.2

no ip http server

!

!

voice-port 1:1

!

voice-port 1:2

dial-type mf

!

voice-port 1:3

!

dial-peer voice 1 pots

application MGCPAPP

port 1:1

!

dial-peer voice 2 pots

application MGCPAPP

port 1:2

!

dial-peer voice 3 pots

application MGCPAPP

port 1:3

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Router B Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname B

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp modem passthrough nse

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

controller T1 1

mode cas

ds0-group 1 timeslots 1-8 type e&m-immediate-start

ds0-group 2 timeslots 9-16 type e&m-wink-start

ds0-group3 timeslots 17-24 type fxs-ground-start

framing esf

linecode b8zs

!

interface Loopback 0

ip address 192.168.5.0 255.255.255.0

!

interface Ethernet0

ip address 172.17.24.103 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.3 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

ip address 10.0.0.6 255.0.0.0

pvc 1/2

vbr-rt 1536 1400 100

encapsulation aal5mux ip

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.9 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

ip route 10.0.0.16 255.0.0.0 ATM0.2

no ip http server

!

!

voice-port 1:1

!

voice-port 1:2

dial-type mf

!

voice-port 1:3

!

dial-peer voice 1 pots

application MGCPAPP

port 1:1

!

dial-peer voice 2 pots

application MGCPAPP

port 1:2

!

dial-peer voice 3 pots

application MGCPAPP

port 1:3

!

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Example 6 SGCP 1.5 Analog EM PBX using Voice over ATM AAL2

The following figure and configuration illustrate the network connections for a Cisco MC3810 series platform with Analog E&M running the MGCP application in SGCP 1.5 mode. ATM AAL2 carries voice traffic.

This configuration is similar to the SGCP 1.5 CAS PBX using Voice over ATM AAL2 configuration, with these exceptions:

- No DS-0 groups are configured for T1/1 because the slot is used by analog voice.

- The E&M port must be configured to match the type of analog PBX to which the port is connected.

- E&M protocol is set to either E&M immediate or wink start. For wink start, set the dial-type to mf.

- Operation must be set to 2-w (for 2-wire) or 4-w (for 4-wire).

- Type is set to I, II, IV, or V.

In this example, the bearer PVC has enough bandwidth for two G711u no vad calls because the router has only two voice ports.

Figure 19

SGCP 1.5 Analog EandM PBX using Voice over ATM AAL2 Configuration

Router A Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

mgcp sgcp restart notify

isdn voice-call-failure 0

!

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 172.16.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.2 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

pvc 1/2

vbr-rt 1536 170 8

encapsulation aal2

vcci 2

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.5 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

no ip http server

!

voice-port 1/3

operation 4-wire

type 2

signal immediate

!

voice-port 1/4

operation 4-wire

type 2

dial-type mf

!

!

dial-peer voice 3 pots

application MGCPAPP

port 1/3

!

dial-peer voice 4 pots

application MGCPAPP

port 1/4

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Router B Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname B

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

mgcp sgcp restart notify

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 172.17.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.3 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

pvc 1/2

vbr-rt 1536 170 8

encapsulation aal2

vcci 2

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.6 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

no ip http server

!

voice-port 1/3

operation 2-wire

type 1

signal immediate

!

voice-port 1/4

operation 4-wire

type 2

dial-type mf

!

dial-peer voice 3 pots

application MGCPAPP

port 1/3

!

dial-peer voice 4 pots

application MGCPAPP

port 1/4

!

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Example 7 SGCP 1.5 Analog EM PBX using Voice over IP over ATM AAL5

The following figure and configuration illustrate the network connections for a Cisco MC3810 series platform RGW with analog FXS loopstart ports running the MGCP application in SGCP 1.5 mode. Voice over IP over ATM AAL5 carries voice traffic.

This configuration is similar to the SGCP 1.5 CAS PBX using Voice over IP over ATM AAL5 configuration, with these exceptions:

- No DS-0 groups are configured for T1/1 because the slot is used by analog voice.

- The E&M port must be configured to match the type of analog PBX to which the port is connected.

- E&M protocol is set to either E&M immediate or wink start. For wink start, set the dial-type to mf.

- Operation must be set to 2-w (for 2-wire) or 4-w (for 4-wire).

- Type is set to I, II, IV, or V.

In this example, the bearer PVC has enough bandwidth for two G711u no vad calls because the router has only two voice ports.

Figure 20

SGCP 1.5 Analog EandM PBX using Voice over IP over ATM AAL5 Configuration

Router A Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

mgcp sgcp restart notify

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Loopback0

ip address 10.0.0.2 255.0.0.

!

interface Ethernet0

ip address 172.16.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.5 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

ip address 10.0.0.6 255.0.0.0

pvc 1/2

vbr-rt 1536 260 8

encapsulation aal5mux ip

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.8 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

ip route 10.0.0.0 255.0.0.0 ATM0.2

no ip http server

!

voice-port 1/3

operation 4-wire

type 2

signal immediate

!

voice-port 1/4

operation 4-wire

type 2

dial-type mf

!

dial-peer voice 3 pots

application MGCPAPP

port 1/3

!

dial-peer voice 4 pots

application MGCPAPP

port 1/4

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Router B Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname B

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

mgcp sgcp restart notify

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Loopback0

ip address 10.0.0.3 255.0.0.0

interface Ethernet0

ip address 172.17.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.7 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

ip address 10.0.0.9 255.0.0.0

pvc 1/2

vbr-rt 1536 170 8

encapsulation aal5mux ip

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.10 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

ip route 10.0.0.20 255.0.0.0 ATM0.2

no ip http server

!

voice-port 1/3

operation 4-wire

type 2

signal immediate

!

voice-port 1/4

operation 4-wire

type 2

dial-type mf

!

dial-peer voice 3 pots

application MGCPAPP

port 1/3

!

dial-peer voice 4 pots

application MGCPAPP

port 1/4

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Example 8 SGCP 1.5 RGW using Voice over ATM AAL2

The following figure and configuration illustrate the network connections for a Cisco MC3810 series platform RGW with analog FXS port running the MGCP application in SGCP 1.5 mode. ATM AAL2 carries voice traffic.

This configuration is similar to the SGCP 1.5 CAS PBX using Voice over ATM AAL2 configuration, with these exceptions:

- No DS-0 groups are configured for T1/1 because the slot is used by analog voice.

- For RGW, the FXS ports' signaling are set to loop start, which is the default.

In this example, the bearer PVC has enough bandwidth for two G711u no vad calls because the router has only two voice ports.

Figure 21

SGCP 1.5 RGW using Voice over ATM AAL2 Configuration

Router A Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 172.16.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.2 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

pvc 1/2

vbr-rt 1536 170 8

encapsulation aal2

vcci 2

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.5 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

no ip http server

!

!

voice-port 1/1

!

voice-port 1/2

!

dial-peer voice 1 pots

application MGCPAPP

port 1/1

!

dial-peer voice 2 pots

application MGCPAPP

port 1/2

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Router B Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname B

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 172.17.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

no cdp enable

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.3 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

pvc 1/2

vbr-rt 1536 170 8

encapsulation aal2

vcci 2

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.6 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

no ip http server

!

voice-port 1/1

!

voice-port 1/2

!

dial-peer voice 1 pots

application MGCPAPP

port 1/1

!

dial-peer voice 2 pots

application MGCPAPP

port 1/2

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Example 9 SGCP 1.5 RGW using Voice over IP over ATM AAL5

The following figure and configuration illustrate the network connections for a Cisco MC3810 series platform RGW with analog FXS port running the MGCP application in SGCP 1.5 mode. Voice over IP over ATM AAL5 carries voice traffic.

This configuration is similar to the SGCP 1.5 CAS PBX Voice Over ATM AAL5 configuration, with these exceptions:

- No DS-0 groups are configured for T1/1 because the slot is used by analog voice.

- For RGW, the FXS ports' signaling are set to loop start, which is the default.

In this example, the bearer PVC has enough bandwidth for two G711u no vad calls because the router has only two voice ports.

Figure 22

SGCP 1.5 RGW using Voice over IP over ATM AAL5 Configuration

Router A Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname A

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 172.16.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.2 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

ip address 10.0.0.5 255.0.0.0

pvc 1/2

vbr-rt 1536 260 8

encapsulation aal5mux ip

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.8 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

ip route 10.0.0.10 255.0.0.0 ATM0.2

no ip http server

!

voice-port 1/1

!

voice-port 1/2

!

dial-peer voice 1 pots

application MGCPAPP

port 1/1

!

dial-peer voice 2 pots

application MGCPAPP

port 1/2

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Router B Configuration

version 12.2

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname B

!

network-clock base-rate 56K

ip subnet-zero

!

mgcp

mgcp call-agent 10.0.0.1 service-type sgcp version 1.5

mgcp sgcp restart notify

mgcp tse payload 100

no mgcp timer receive-rtcp

mgcp timer net-cont-test 3000

isdn voice-call-failure 0

!

!

cns event-service server

voice-card 0

!

controller T1 0

mode atm

framing esf

linecode b8zs

!

interface Ethernet0

ip address 172.17.24.101 255.255.0.0

!

interface Serial0

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface Serial1

no ip address

no ip route-cache

no ip mroute-cache

shutdown

!

interface ATM0

no ip address

ip mroute-cache

no atm ilmi-keepalive

!

interface ATM0.1 point-to-point

description signaling vc

ip address 10.0.0.3 255.0.0.0

pvc 1/1

vbr-rt 1536 64

encapsulation aal5snap

!

interface ATM0.2 point-to-point

description bearer vc

ip address 10.0.0.6 255.0.0.0

pvc 1/2

vbr-rt 1536 260 8

encapsulation aal5mux ip

!

interface ATM0.3 point-to-point

description data vc

ip address 10.0.0.7 255.0.0.0

pvc 1/100

encapsulation aal5snap

!

interface FR-ATM20

no ip address

no ip route-cache

shutdown

!

ip classless

ip route 10.0.0.12 255.0.0.0 ATM0.2

no ip http server

!

voice-port 1/1

!

voice-port 1/2

!

dial-peer voice 1 pots

application MGCPAPP

port 1/1

!

dial-peer voice 2 pots

application MGCPAPP

port 1/2

!

line con 0

exec-timeout 0 0

privilege level 15

transport input none

line aux 0

line 2 3

line vty 0 4

login

!

end

Tip

See the "Additional References for MGCP and SGCP" section for related documents, standards, and MIBs, and the " Glossary " for definitions of terms in this guide.

Secure Tone on MGCP TDM Gateways

The Secure Tone feature provides support for Cisco Unified Communications Manager (UCM) to play a secure or non secure tone when the secure status of the call changes.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for Secure Tone on MGCP TDM Gateways

- Cisco Unified Communications Manager (UCM) 8.0(3) or a later release is running.

- Cisco IOS Media Gateway Control Protocol (MGCP) version 0.1 is configured.

Restrictions for Secure Tone on MGCP TDM Gateways

- Cisco UCM 8.0 will play the secure or non-secure indication tone to the protected device only when a two-way media connection is established.

- This feature is available only on Cisco IOS secure images.

Secure Tone on MGCP TDM Gateways

Cisco UCM not provides support for including supplementary service calls such as transfer and conferencing also changes the way in which secure and non secure indication tones are played. This new feature of playing the secure indication tone is based on the overall secure status unlike the earlier versions when it was based on end-to-end device's protected status. If the overall status is secure, the Secure Indication Tone (SIT) will be played to the protected phone. If the overall secure status is non secure, the Non-Secure Indication Tone (NSIT) will be played to the protected phone. SIT or NSIT will be played when a two-way audio media connection is first established on the protected phone.

Cisco UCM sends a Notification Request (RQNT) with the newly defined tone ID for secure or non secure status. Following is a sample message sent from Cisco UCM to the MGCP gateway to play the secure tone when the status of the call changes:

RQNT 199 S2/DS1-0/1@nw053b-3745.cisco.com MGCP 0.1

X: 1

R: D/[0-9ABCD*#]

S: X+TONE/st

Q: process,loop

How to Configure Secure Tone on MGCP TDM Gateways

- Configuring Secure Tone on MGCP TDM Gateways

- Verifying and Troubleshooting Secure Tone on MGCP TDM Gateways

Configuring Secure Tone on MGCP TDM Gateways

SUMMARY STEPS

1.   enable

2.   configure terminal

3.   mgcp package-capability tone-package

4.   exit

DETAILED STEPS

Step 1 enable

Enables privileged EXEC mode.

- Enter your password if prompted.

Example:

Router> enable

Step 2 configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

Step 3 mgcp package-capability tone-package

Configures the secure tone capability for MGCP packages.

Example:

Router(config)# mgcp package-capability tone-package

Step 4 exit

Exits the current mode.

Example:

Router# exit

Verifying and Troubleshooting Secure Tone on MGCP TDM Gateways

SUMMARY STEPS

1.   show mgcp

2.   debug mgcp all

3.   debug mgcp packets

DETAILED STEPS

Step 1 show mgcp

This command is used for displaying the values for MGCP parameters. The parameters are displayed only when the tone package has been enabled using the mgcp package-capability tone-packagecommand.

Step 2 debug mgcp all

This command is used to enable all debug traces for the MGCP gateway.

Step 3 debug mgcp packets

This command is used for enabling debug traces for the MGCP packets.

Configuration Examples for Secure Tone on MGCP TDM Gateways

- Example Configuring Secure Tone for MGCP TDM Gateways

- Example Verifying Secure Tone for MGCP TDM Gateways

Example Configuring Secure Tone for MGCP TDM Gateways

The following example shows how to enable the trunk package, DTMF package, script package, and tone package on the gateway, and then names the trunk package as the default package for the gateway:

Router(config)# mgcp package-capability trunk-package

Router(config)# mgcp package-capability dtmf-package

Router(config)# mgcp package-capability script-package

Router(config)# mgcp package-capability tone-package

Router(config)# mgcp default-package trunk-package

Example Verifying Secure Tone for MGCP TDM Gateways

The following is a partial sample output from the show mgcp command when the package capability has been enabled for tone package:

Router# show mgcp

MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE

MGCP call-agent: 10.7.0.200 3460 Initial protocol service is MGCP 0.1

MGCP validate call-agent source-ipaddr DISABLED

MGCP block-newcalls DISABLED

MGCP send SGCP RSIP: forced/restart/graceful/disconnected DISABLED

MGCP quarantine mode discard/step

GCP quarantine of persistent events is ENABLED

MGCP dtmf-relay for VoIP disabled for all codec types

MGCP dtmf-relay for VoAAL2 disabled for all codec types

MGCP voip modem passthrough mode: NSE, codec: g711ulaw, redundancy: DISABLED,

MGCP voaal2 modem passthrough disabled

MGCP voip nse modem relay: Disabled

MGCP voip mdste modem relay: Enabled

SPRT rx v14 hold time: 50 (ms), SPRT tx v14 hold count: 16,

SPRT tx v14 hold time: 20 (ms), SPRT Retries: 12

SSE redundancy interval: 20 (ms), SSE redundancy packet: 3,

SSE t1 timer: 1000 (ms), SSE retries: 3

The following lines show that the tone-package keyword is enabled:

MGCP supported packages: gm-package dtmf-package mf-package trunk-package

line-package hs-package rtp-package script-package ms-package

dt-package mo-package mt-package sst-package mdr-package

fxr-package pre-package mdste-package srtp-package tone-package

Additional References

Related Documents

Related Topic

Document Title

Cisco IOS commands

Cisco IOS Master Commands List, All Releases

MGCP configuration

MGCP and Related Protocols Configuration Guide

Cisco IOS voice configuration

Cisco IOS Voice Configuration Library

Cisco IOS voice commands

Cisco IOS Voice Command Reference

Cisco IOS debug commands

Cisco IOS Debug Command Reference

Standards

Standard

Title

None

--

MIBs

MIB

MIBs Link

None

To locate and download MIBs for selected platforms, Cisco software releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://www.cisco.com/go/mibs

RFCs

RFC

Title

None

--

Technical Assistance

Description

Link

The Cisco Support and Documentation website provides online resources to download documentation, software, and tools. Use these resources to install and configure the software and to troubleshoot and resolve technical issues with Cisco products and technologies. Access to most tools on the Cisco Support and Documentation website requires a Cisco.com user ID and password.

http://www.cisco.com/cisco/web/support/index.html

Feature Information for Secure Tone on MGCP TDM Gateways

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Table 9

Feature Information for Secure Tone Support on MGCP TDM Gateway

Feature Name

Releases

Feature Information

Secure Tone Support on MGCP TDM Gateway

15.1(4)M

This feature provides support for the Cisco UCM to play a secure or non secure tone when the secure status of the call changes or when the media are reconnected after the call is answered.

The following command was introduced or modified: mgcp package capability tone-package.

DSP Voice Quality Statistics in DLCX Messages

The DSP Voice Quality Statistics in DLCX Messages feature provides a way to trace a Media Gateway Control Protocol (MGCP) call between a Cisco PGW 2200 Softswitch and the Cisco IOS gateway by including the MGCP call ID and the DS0 and digital signal processor (DSP) channel ID in call-active and call-history records.

The voice quality statistics are sent as part of the MGCP Delete Connection (DLCX) message. By correlating an MGCP call on the Cisco PGW 2200 Softswitch with a call record on the gateway, you can understand and debug additional statistics from the DSP for problems related to voice quality.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the Feature Information Table at the end of this document. Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for DSP Voice Quality Statistics in DLCX Messages

You must be using Cisco PGW 2200 version 9.4.1 or a later version with a patch level higher than CSCOgs008/CSCOnn008.

Restrictions for DSP Voice Quality Statistics in DLCX Messages

When the Secure Real-Time Transfer Protocol (SRTP) is enabled, the DLCX message will not report voice quality statistics. The following lines will be omitted for SRTP calls:

- DSP/Endpoint Configuration (EC)

- DSP/MOS K-Factor Statistics (KF)

- DSP/Concealment Statistics (CS)

- DSP/R-Factor Statistics (RF)

- DSP/User Concealment (UC)

Information About DSP Voice Quality Statistics in DLCX Messages

- Cisco PGW 2200

- MGCP

- Voice Quality Statistics

- Quality of Service for Voice

Cisco PGW 2200

A call agent (or media gateway controller) and softswitch are industry standard terms used to describe the network element that provides call control functionality to telephony and packet networks. The Cisco PGW 2200 Softswitch functions as a call agent or softswitch in "call control mode."

Note

All voice quality parameters for Cisco IOS Release 12.4(4)T and later releases are supported only on the Cisco PGW 2200 call agent.

A public switched telephone network (PSTN) gateway provides an interface between traditional Signaling System No. 7 (SS7) or non-SS7 networks and networks based on MGCP, H.323, and Session Initiation Protocol (SIP), which include signaling, call control, and time-division multiplexing/IP (TDM/IP) gateway functions. The Cisco PGW 2200 Softswitch, coupled with Cisco media gateways, functions as a PSTN gateway.

Caution

There is a significant performance degradation on the Cisco PGW 2200 if all connected gateways have the DSP Voice Quality Statistics in DLCX Messages feature enabled. Enabling voice quality statistics on the gateway should only be performed by Cisco personnel.

The Cisco PGW 2200 Softswitch, in either signaling or call control mode, provides a robust, carrier-class interface between PSTN and IP-based networks. Interworking with Cisco media gateways, the Cisco PGW 2200 Softswitch supports a multitude of applications and networks, including the following:

- Application service provider (ASP) termination

- Centralized routing and billing for clearinghouse of IP-based networks

- Dial access

- International and national transit networks

- Managed business voice applications

- Managed voice VPNs

- Network clearinghouse applications

- PSTN access for hosted and managed IP telephony

- PSTN access for voice over broadband networks

- Residential voice applications

MGCP

MGCP defines the call control relationship between call agents (CAs) and VoIP gateways that translate audio signals to and from the packet network. CAs are responsible for processing the calls.

An MGCP gateway handles the translation between audio signals and the packet network. The gateways interact with a CA, also called a media gateway controller (MGC), which performs signal and call processing on gateway calls. MGCP uses endpoints and connections to construct a call.

Endpoints are sources of or destinations for data and can be physical or logical locations in a device. Connections can be either point-to-point or multipoint. The gateway can be a Cisco router, an access server, or a cable modem, and the CA is a server from a third-party vendor.

Voice Quality Statistics

The Cisco PGW 2200 Softswitch can capture voice quality statistics sent from MGCP-controlled media gateways and can propagate the statistics into call detail records (CDRs) at the end of each call. The Cisco AS5x00 media gateways send voice quality statistics to the Cisco PGW 2200 Softswitch.

Most voice quality statistics are available from the DSP and are controlled using RTP Control Protocol (RTCP) report interval statistics polling. The mean and maximum values are calculated by Cisco IOS software-based polling, which results in additional CPU load for each call. The additional CPU load can be controlled by configuring polling interval by using the ip rtcp report interval command.

The playout delay, playout error, and DSP receive and transmit statistics are automatically polled periodically. Polling for the voice quality statistics, level, and error parameters can be added. For logging the voice quality statistics using syslog, the existing VoIP gateway accounting has been extended. For more information about statistics polling, see the ip rtcp report interval command in the Cisco IOS Voice Command Reference.

Table 10

Voice Quality Statistics for Cisco IOS Release 12.4(4)T and Later Releases

DSP Technology

Platform

Voice Quality Statistics

MSA V6

Cisco AS5350, Cisco AS5350XM, Cisco AS5400, Cisco AS5400HPX, Cisco 5400XM, and Cisco AS5850 with an NPE60 or NPE108 universal port feature card.

DSP/TX

DSP/RX

DSP/PD

DSP/PE

DSP/LE

DSP/ER

DSP/IC

TIC5510

- Cisco 2800 series and Cisco 3800 series integrated services routers with PVDM2 modules.

- Cisco VG224 voice gateway

- Cisco IAD2430 series integrated access devices.

- Cisco 2600XM, Cisco 2691, Cisco 3700 series access routers, and Cisco 2811, Cisco 2821, Cisco 2851, Cisco 3800 series integrated services routers with the following network modules: NM-HDV2, NM-HDV2-1T1/E1, NM-HD-1V, NM-HD-2V, NM-HD-2VE.

- Cisco 2821, Cisco 2851, Cisco 3825, and Cisco 3845 with the EVM-HD-8FXS/DID module.

DSP/TX

DSP/RX

DSP/PD

DSP/PE

DSP/LE

DSP/ER

DSP/IC

DSP/EC

DSP/KF

DSP/CS

DSP/RF

DSP/UC

DSP/DL

Quality of Service for Voice

The DSP Voice Quality Statistics in DLCX Messages feature is part of the Cisco quality of service (QoS) technology. QoS is the ability of a network to provide better service to selected network traffic over various technologies, including ATM, Ethernet and 802.1 networks, Frame Relay, SONET, and IP-routed networks that may use any or all of these underlying technologies.

QoS provides the following benefits:

- Control over bandwidth, equipment, and wide-area facilities--As an example, you can limit the bandwidth consumed over a backbone link by FTP or limit the queueing of an important database access.

- More efficient use of network resources--Network analysis management and accounting tools enable you to know what your network is being used for and ensure that you are servicing the most important traffic to your business.

- Ability to customize services--QoS enables ISPs to offer carefully customized grades of service differentiation to their customers.

- Coexistence of mission-critical applications--Cisco QoS technologies ensure that bandwidth and minimum delays required by time-sensitive multimedia and voice applications are available and that other applications using the link get their fair service without interfering with mission-critical traffic.

- Foundation for a fully integrated network--Cisco QoS technologies fully integrate a multimedia network, for example, by implementing weighted fair queueing (WFQ) to increase service predictability and by implementing IP precedence signaling to differentiate traffic. In addition, the availability of Resource Reservation Protocol (RSVP) allows you to take advantage of dynamically signaled QoS.

To deliver QoS across a network that comprises heterogeneous technologies (for example, IP, ATM, LAN switches), basic QoS architecture consists of the following three components:

- QoS within a single network element (for example, queueing, scheduling, and traffic shaping tools).

- QoS signaling techniques for coordinating end-to-end QoS between network elements.

- QoS policy, management, and accounting functions to control and administer end-to-end traffic across a network.

Voice Quality Parameters for Cisco IOS Release 12.4(4)T and Later Releases

The following voice quality parameters were introduced in Cisco IOS Release 12.4(4)T:

DSP/EC

The following parameters describe the configuration of a VoIP endpoint. You can define these parameters and they are useful for debugging and logging purposes because they capture the state of the endpoint.

- CI--Codec ID. A string or a number that identifies the voice codec which is currently used in the call.

- FM--Frame size. Native frame size, in milliseconds (ms), of the selected codec. An example of a frame size and codec combination is G.729a/30ms. For the G.711 codec, the frame size is a value that you can define in the voice dial peer. For example, G.711 at 80 bytes gives 10 ms per frame. G.711 at 240 bytes gives 30 ms per frame.

- FP--Frames per packet. Number of codec speech frames encapsulated into a single Real-time Transport Protocol (RTP) packet. Typical values are 1, 2, and 3. Packing lower number of frames per packet results in lower efficiency of IP bandwidth usage. The tradeoff is lower delays and higher robustness of the network.

- VS--Voice Activity Detection (VAD)-enabled flag. VAD is enabled when VS has a value of one. It results in compression of silent periods leading to reduced or zero packets per second. VAD is disabled when VS has a value of zero. It results in the transmission of continuous packets per second irrespective of active or silent periods on the transmission path.

- GT--Transmission gain factor (linear). Digital gain multiplier applied to the transmission on the signal path from the PSTN toward the network. GT is applied at the echo canceller Sout port. A gain factor of less than one indicates a loss pad.

- GR--Reception gain factor (linear). Digital gain multiplier applied to reception on the signal path from the network toward the PSTN. GR is applied at the echo canceller Rin port. A gain factor of less than one indicates a loss pad.

- JD--Jitter buffer mode. It consists of the following modes:

- Adaptive mode = 1

- Fixed mode (no timestamps) = 2

- Fixed mode (with timestamps) = 3

- Fixed mode (with passthrough) = 4

- JN--Jitter buffer nominal playout delay. Size of the jitter buffer in milliseconds. An adaptive jitter buffer tries to make the playout delay equal to the nominal (desired) delay when the observed jitter is small enough to allow this adjustment. For a fixed-mode jitter buffer, the nominal setting is the constant playout delay itself.

- JM--Minimum playout delay. Minimum playout delay setting for an adaptive-mode jitter buffer. The playout delay never goes below the minimum playout setting even if the observed jitter is zero. This setting is not used for a fixed-mode jitter buffer because the playout delay is fixed and constant at the nominal setting.

- JX--Maximum playout delay. Sets the limit for increasing the playout delay of an adaptive-mode jitter buffer. An adaptive buffer increases when the jitter is higher than the instantaneous playout delay value.

DSP/KF

K-factor is an endpoint mean opinion score (MOS) estimation algorithm defined in the ITU standard P.VTQ. It is a general estimator of the mean value of a perceptual evaluation of speech quality (PESQ) population for a specific impairment pattern.

The ITU standard P.862 defines and describes the PESQ as an objective method for end-to-end speech quality assessment of narrow band telephone networks and speech codecs.

Mean opinion score (MOS) is associated with the output of a well-designed listening experiment. All MOS experiments use a five-point PESQ scale as defined in the ITU standard P.862.1. The MOS estimate is inversely proportional to frame loss density. Clarity decreases as more frames are lost or discarded at the receiving end.

K-factor represents a weighted estimate of average user annoyance due to distortions caused by effective packet loss such as dropouts and warbles. It does not estimate the impact of delay-related impairments such as echo. It is an estimate of listening quality (MOS-LQO) rather than conversational quality (MOS-CQO), and measurements of average user annoyance range from 1 (poor voice quality) to 5 (very good voice quality).

K-factor is trained or conditioned by speech samples from numerous speech databases, where each training sentence or network condition associated with a P.862.1 value has a duration of eight seconds. For more accurate scores, K-factor estimates are generated for every 8 seconds of active speech.

K-factor and other MOS estimators are considered to be secondary or derived statistics because they warn a network operator of frame loss only after the problem becomes significant. Packet counts, concealment ratios, and concealment second counters are primary statistics because they alert the network operator before network impairment has an audible impact or is visible through MOS.

- KF--K-factor MOS-LQO estimate (instantaneous). Estimate of the MOS score of the last 8 seconds of speech on the reception signal path. If VAD is active, the MOS calculation is suspended during periods of received silence to avoid inflation of MOS scores for calls with higher silence fractions.

- AV--Average K-factor score. Running average of scores observed since the beginning of a call.

- MI--Minimum K-factor score. Minimum score observed since the beginning of a call, and represents the worst sounding 8-second interval.

- BS--Baseline (maximum) K-factor score. K-factor score that can be obtained for the defined codec.

- NB--Number of bursts. Number of burst loss events after a call is started. A burst loss is a contiguous run of concealment events of length greater than one.

- FL--Average frame loss count. Total number of frame losses and concealment events observed after starting a call. The ratio of FL/NB provides the mean burst length in frames. The total concealment duration of the call is provided by the parameter DSP/CS: CT.

- NW--Number of windows. Total number of K-factor windows observed after starting a call. The number of windows is directly proportional to the duration of a call.

- VR--Version ID. Version number that identifies a specific K-factor MOS score.

DSP/CS

DSP/CS measures packet (frame) loss and its effect on voice quality in an impaired network. The parameters for concealment statistics are as follows:

- CR--Concealment ratio (instantaneous). An interval-based average concealment rate, and is the ratio of concealment time over speech time for the last 3 seconds of active speech. When VAD is enabled, calculation of the concealment ratio is suspended during periods of speech silence. During this suspension, it may take more than 3 seconds for a new value to be generated.

- AV--Average CR. Average of all CR reports after starting a call.

- MX--Maximum CR. The maximum concealment ratio observed after starting a call.

- CS--Concealed time. The duration of time in seconds during which some concealment is observed.

- CT--Total concealment time. The total duration of time in milliseconds during which concealment is observed after starting a call.

- TT--Total speech time. The duration of time in milliseconds during which active speech is observed after starting a call.

- OK--Ok time. The duration of time in seconds during which no concealment is observed.

- SC--Severely concealed time. The duration of time in seconds during which a significant amount of concealment is observed. If the concealment observed is usually greater than 50 milliseconds or approximately five percent, it is possible that the speech is not very audible.

- TS--Concealment threshold. The threshold in milliseconds used to determine a second as severely concealed. The threshold for concealed seconds is 0 milliseconds, and for severely concealed seconds is 50 milliseconds.

DSP/RF

The R-factor helps in planning voice transmission. In ITU standards G.107 and G.113, the R-factor is defined as follows:

R = Ro - Is - Id - Ie-eff + A

The parameters for the R-Factor are as follows:

- Ro is based on the signal-to-noise ratio.

- Is is the simultaneous impairment factor and includes the overall loudness rating.

- Id is the delay impairment factor and includes talker (Idte) and listener (Idle) echos, and delays (Idd).

- Ie-eff is the equipment impairment factor and includes packet losses and the types of codecs.

- A is the advantage factor.

- ML--R-factor MOS listening quality objective. It reflects only packet loss and codec-related impairments and does not include delay effects.

- MC--R-factor MOS-CQE.

- R1--R-factor LQ profile 1.

- R2--R-factor LQ profile 2.

- IF--Effective codec impairment (Ie_eff).

- ID--Idd.

- IE--Codec baseline score (Ie). The tabulated baseline codec impairment factor.

- BL--Codec baseline (Bpl). The packet loss robustness factor for the codec being used.

- R0--R0 (default). The nominal value at which the signal-to-noise ratio is considered nominal.

DSP/UC

The parameters for user concealment are as follows:

- U1--User concealment seconds 1 count (UCS1)

- U2--User concealment seconds 2 count (UCS2)

- T1--UCS1 threshold in milliseconds

- T2--UCS2 threshold in milliseconds

DSP/DL

The parameters for delay statistics are as follows:

- RT--Round trip delay

- ED--End system delay

How to Configure DSP Voice Quality Statistics in DLCX Messages

- Configuring DSP Voice Quality Statistics in DLCX Messages

- What to Do Next

Configuring DSP Voice Quality Statistics in DLCX Messages

To configure voice quality statistics reporting for MGCP, use the following commands beginning in user EXEC mode.

SUMMARY STEPS

1.   enable

2.   configure terminal

3.   mgcp voice-quality stats [priorityvalue] [all]

4.   end

DETAILED STEPS

Step 1 enable

Enables privileged EXEC mode.

- Enter your password if prompted.

Example:

Router> enable

Step 2 configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

Step 3 mgcp voice-quality stats [priorityvalue] [all]

Enables voice quality statistics reporting for MGCP.

The following parameters are sent by default if the priority or all keywords are not used: DSP/TX, DSP/RX, DSP/PD, DSP/PE, DSP/LE, DSP/ER, DSP/IC.

Priority 1 parameters are: DSP/TX, DSP/RX, DSP/PD, DSP/LE, DSP/EC, DSP/CS, DSP/DL.

Priority 2 parameters are: DSP/PE, DSP/ER, DSP/IC, DSP/KF, DSP/RF, DSP/UC.

Using Priority 2 is similar to using the all keyword when the output contains the following parameters: DSP/TX, DSP/RX, DSP/PD, DSP/PE, DSP/LE, DSP/ER, DSP/IC, DSP/EC, DSP/KF, DSP/CS, DSP/RF, DSP/UC, and DSP/DL.

Example:

Router(config)# mgcp voice-quality-stats priority 1

Step 4 end

Exits global configuration mode and enters privileged EXEC mode.

Example:

Router(config)# end

What to Do Next

Use the following Troubleshooting Tips if you did not get the expected results after configuring voice quality statistics reporting for MGCP. See the "Troubleshooting Tips" section for additional guidelines.

Troubleshooting Tips

Use the debug mgcp packets command to display statistics reported in the DLCX message generated at the end of a call. The following is sample debug output:

Router# debug mgcp packets

DLCX 311216 s6/ds1-4/1@as5400a MGCP 0.1

C: 48A4B

I: 2

R:

S:

X: 4BFAF

*May 5 10:20:51.643: send_mgcp_msg, MGCP Packet sent to 19.0.2.10:2427 --->

*May 5 10:20:51.643: 250 311216 OK

P: PS=1469, OS=28943, PR=1518, OR=29923, PL=0, JI=100, LA=0

DSP/TX: PK=1448, SG=0, NS=23, DU=206450, VO=39000

DSP/RX: PK=1449, SG=0, CF=23, RX=206450, VO=38000, BS=0, BP=0, LP=0

DSP/PD: CU=100, MI=90, MA=110, CO=69352809, IJ=0

DSP/PE: PC=0, IC=0, SC=0, RM=6, BO=0, EE=0

DSP/LE: TP=-24, TX=-440, RP=-87, RM=-870, BN=0, ER=50, AC=90, TA=-24, RA=-87

DSP/ER: RD=0, TD=0, RC=0, TC=0

DSP/IC: IC=0

Verifying DSP Voice Quality Statistics in DLCX Messages

Use the following show commands to check your configuration:

SUMMARY STEPS

1.    Obtain the call ID by using the show call active voice compact command in privileged EXEC mode.

2.    Check the status of active calls using the call ID obtained from the show call active voice brief command.

3.    Verify your configuration using the show call history voice brief command.

DETAILED STEPS

Step 1 Obtain the call ID by using the show call active voice compact command in privileged EXEC mode.

Example:

Router# show call active voice compact

G<id> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>

Total call-legs: 2

G11D6 ORG T187 g729r8 TELE P

G11D6 ORG T0 g729r8 VOIP P 192.0.2.1:19324

Step 2 Check the status of active calls using the call ID obtained from the show call active voice brief command.

Example:

Router# show call active voice brief id 11D6

<ID>: <CallID> <start>.<index> +<connect> pid:<peer_id> <dir> <addr> <state>

dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>

IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>

delay:<last>/<min>/<max>ms <codec>

media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>

long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>

MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>

last <buf event time>s dur:<Min>/<Max>s

FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm

MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops>

speeds(bps): local <rx>/<tx> remote <rx>/<tx>

Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>

bw: <req>/<act> codec: <audio>/<video>

tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 0

Step 3 Verify your configuration using the show call history voice brief command.

Example:

Router# show call history voice brief

<ID>: <CallID> <start>.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>

dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)

IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>

delay:<last>/<min>/<max>ms <codec>

media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>

long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>

MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>

last <buf event time>s dur:<Min>/<Max>s

FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

Telephony <int> (callID) [channel_id] tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm

MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> disc:<cause code>

speeds(bps): local <rx>/<tx> remote <rx>/<tx>

Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>

bw: <req>/<act> codec: <audio>/<video>

tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

Total call-legs: 0

Configuration Examples for DSP Voice Quality Statistics in DLCX Messages

- Example: Configuring DSP Voice Quality Statistics in DLCX Messages

Example: Configuring DSP Voice Quality Statistics in DLCX Messages

The following example shows how to enable voice quality statistics reporting for MGCP:

Router> enable

Router# configure terminal

Router(config)# mgcp voice-quality-stats

Router(config)# end

The following example shows the voice quality parameters selected for priority 1:

Router(config)# mgcp voice-quality-stats priority 1

16:38:20.461771 192.0.2.1:2427 192.0.2.4:2427 MGCP...... -> 250 1133 OK

P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=65, LA=0

DSP/TX: PK=118, SG=0, NS=1, DU=28860, VO=2350

DSP/RX: PK=0, SG=0, CF=0, RX=28860, VO=0, BS=0, LP=0, BP=0

DSP/PD: CU=65, MI=65, MA=65, CO=0, IJ=0

DSP/LE: TP=0, RP=0, TM=0, RM=0, BN=0, ER=0, AC=0

DSP/IN: CI=0, FM=0, FP =0, VS=0, GT=0, GR=0, JD=0, JN=0, JM=0,

DSP/CR: CR=0, MN=0, CT=0, TT=0,

DSP/DC: DC=0,

DSP/CS: CS=0, SC=0, TS=0,

DSP/UC: U1=0, U2=0, T1=0, T2=0

The following example shows the voice quality parameters selected for the all keyword:

Router(config)# mgcp voice-quality-stats all

16:38:20.461771 192.0.2.1:2427 192.0.2.4:2427 MGCP...... -> 250 1133 OK

P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=65, LA=0

DSP/TX: PK=118, SG=0, NS=1, DU=28860, VO=2350

DSP/RX: PK=0, SG=0, CF=0, RX=28860, VO=0, BS=0, LP=0, BP=0

DSP/PD: CU=65, MI=65, MA=65, CO=0, IJ=0

DSP/PE: PC=0, IC=0, SC=0, RM=0, BO=0, EE=0

DSP/LE: TP=0, RP=0, TM=0, RM=0, BN=0, ER=0, AC=0

DSP/ER: RD=0, TD=0, RC=0, TC=0

DSP/IC: IC=0

DSP/EC: CI=0, FM=0, FP =0, VS=0, GT=0, GR=0, JD=0, JN=0, JM=0, JX=0,

DSP/KF: KF=0, AV=0, MI=0, BS=0, NB=0, FL=0,

DSP/CS: CR=0, AV=0, MN=0, MX=0, CS=0, SC=0, TS=0, DC=0,

DSP/RF: ML=0, MC=0, R1=0, R2=0, IF=0, ID=0, IE=0, BL=0, R0=0,

DSP/UC: U1=0, U2=0, T1=0, T2=0,

DSP/DL: RT=0, ED=0

Additional References

Related Documents

Related Topic

Document Title

Cisco IOS commands

Cisco IOS Master Commands List, All Releases

Voice commands

- Cisco IOS Voice Command Reference-A through C

- Cisco IOS Voice Command Reference-D through I

- Cisco IOS Voice Command Reference-K through R

- Cisco IOS Voice Command Reference-S commands

- Cisco IOS Voice Command Reference-T through Z

How to configure QoS for Cisco features

Cisco IOS Quality of Service Configuration Guide

Cisco MGC documentation index

Cisco Media Gateway Controllers

How to configure MGCP

Configuring Media Gateway Protocol and Related Protocols

How to configure QoS for voice applications

Configuring Quality of Service for Voice

How to configure voice ports

Configuring Voice Ports

Enabling basic management protocols on Cisco access platforms

Enabling Management Protocols: NTP, SNMP, and Syslog

Release Notes, Cisco IOS Release 12.3

Release Notes Index, Cisco IOS Release 12.3

Standards and RFCs

Standard/RFC

Title

No new or modified standards or RFCs are supported by this feature, and support for existing standards or RFCs has not been modified by this feature.

--

MIBs

MIB

MIBs Link

No new or modified MIBs are supported by this feature, and support for existing MIBs has not been modified by this feature.

To locate and download MIBs for selected platforms, Cisco software releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://www.cisco.com/go/mibs

Technical Assistance

Description

Link

The Cisco Support and Documentation website provides online resources to download documentation, software, and tools. Use these resources to install and configure the software and to troubleshoot and resolve technical issues with Cisco products and technologies. Access to most tools on the Cisco Support and Documentation website requires a Cisco.com user ID and password.

http://www.cisco.com/cisco/web/support/index.html

Feature Information for DLCP Voice Quality Statistics in DLCX Messages

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Table 11

Feature Information for DLCP Voice Quality in DLCX Messages

Feature Name

Releases

Feature Information

DSP Voice Quality Statistics in DLCX Messages

12.3(3)

12.4(4)T

15.1(3)T

The DSP Voice Quality Statistics in DLCX Messages feature provides a way to trace a Media Gateway Control Protocol (MGCP) call between a Cisco PGW 2200 and the Cisco IOS gateway by including the MGCP call ID and the DS0 and digital signal processor (DSP) channel ID in call-active and call-history records.

In Cisco IOS Release 12.4(4)T, new voice quality parameters were introduced.
The following commands were introduced or modified: debug mgcp, mgcp voice-quality stats.

Glossary

AAA--Authentication, authorization, and accounting. Security services for packet networks.

AAL2--ATM Adaptation Layer 2. Supports traffic needing precise timing, such as compressed voice and video.

AAL5--ATM Adaptation Layer 5

AUEP--Audit Endpoint command. An MGCP or SGCP command that is used by the call agent to determine the status of a given endpoint on a gateway.

BGW--Business gateway.

bind--In MGCP, configuring the source address for signaling and media packets to the IP address of a specific interface.

business gateway--An xGCP media gateway which is a business customer premises equipment that has connection(s) to the VoIP network as well as connection(s) to the user's telephony equipment (typically a PBX, a corporate LAN or WAN). Such gateways are used to eliminate or reduce the need for individual medium (voice, data, and so forth) connectivity.

CA--Call agent.

CAC--Call admission control. Controls whether a call can be established, according to availability of local or network resources.

call agent--An intelligent entity in an IP telephony network which handles call control in an MGCP model Voice over IP network.

An intelligent entity in an IP telephony network that handles call control in an xGCP model Voice-over-IP network. A Call Agent is also known as a media gateway controller (MGC).

CAS--Channel Associated Signaling.A signaling technique that uses the same facility path for both voice and signaling traffic.

CCS--Common Channel Signaling

CID--AAL2 Channel Identifier

CLASS--Custom Local Area Subscriber Services, usually referred to as "Custom Calling" features

CLI--command-line interface.

CRCX--Create Connection request. Used in an MGCP call sequence by gateways to exchange SDP.

CSM--Call switching module. Card that contains digit collection logic to process incoming calls for automatic number information (ANI) and dialed number identification service (DNIS) digits.

DDR--Dial-on-demand routing.

disconnected procedure--When an endpoint attempts to communicate with its call agent and fails, the gateway may initiate the disconnected procedure, in which a timer (the disconnected timer) is started and RSIP messages are sent to the call agent at prescribed intervals until contact is established. This procedure ensures that an RSIP message is the first message to reach the call agent after communications are reestablished. The call agent may request an AUEP after the RSIP is received.

DNS--Domain Name System. Used to translate H.323 IDs, URLs, or e-mail IDs into IP addresses. DNS is also used to assist in locating remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.

DS-0--64-kbps channel in a T1/E1 line.

DTMF--Dual-tone multifrequency.Tones made by pushing buttons on a telephone. Tones used to send phone number digits to and from a switch. DTMF tones identify the numbers 0 through 9 and the * and # symbols.

E&M--Ear and Mouth analog signaling.

fax passthrough--The GW sends fax data encoded using a codec such as G.711 ulaw or G.711 alaw, resulting in a more reliable transmission. Fax passthrough provides repression of compression, echo cancellation, and other functions; issues redundant packets to ensure complete transmission; and provides a buffer to protect against clock skew.

FGD--Feature Group-D. Identifies a standardized service available to carriers delivered on a channelized T1 line.

FGD-OS--Feature Group-D Operator Services protocol. OS is a telephony signaling protocol for calls that originate from the Bell Operating Company (BOC) and are sent towards the carrier switch.

fx: extension--An extension of the local connection option used by the CA to instruct the GW to be in CA-controlled mode or GW-controlled mode.

FXO--Foreign Exchange Operator--An interface from a telephone to a PSTN central office or a station interface on a PBX.

FXS--Foreign Exchange Station--An interface that connects to a telephone, key set, or PBX to supply ring, voltage, and dial tone.

GW--Gateway.

IETF--Internet Engineering Task Force. Task force that consists of over 80 working groups responsible for developing Internet standards. The IETF operates under the auspices of the Internet Society.

IPDC--Internet Protocol Device Control. A device control specification.

ISP--Internet service provider.

MDCX--Modify connection request. Used in an MGCP call sequence by gateways to exchange SDP information.

media gateway--The emerging industry standard generic term for a gateway.Equipment that connects the PSTN or a PBX with the Voice-over-IP network. It is controlled by a call agent using MGCP.

MGC--Media gateway controller. The emerging industry standard generic term for the VSC.Another name for call agent.

MGCP--Media Gateway Control Protocol. A merging of the IPDC and SGCP protocols.

NAS--network access server. Communications processor that connects asynchronous devices to a LAN or WAN through network and terminal emulation software. Performs both synchronous and asynchronous routing of supported protocols.

NCS--Network-based Call Signaling. PacketCable protocol, profile of MGCP 1.0 for residential gateways.

NSE--Named Signaling Event. Format of RTP packets used for applications such as modem relay and fax relay. NSEs have different payload values than NTEs.

NTE--Named Telephony Event. Format of RTP packets used to transport DTMF digits as defined in RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals .

package--A set of signals and events that define a type of voice endpoint or connection. Examples include line-package, trunk-package, dtmf-package, and atm-package

payload type--Payload types are defined in RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals . A payload type is a number from 96 through 127 that identifies the type of payload carried in the packet (for example, a payload type of 121 denotes a Cisco DTMF relay payload; a payload type of 122 denotes a fax payload). The payload type should be identical on the GW and call agent.

PDU--protocol data units. Used by bridges to transfer connectivity information.

PRI--ISDN primary rate interface

PSTN--Public Switched Telephone Network

PVC--Permanent Virtual Circuit

residential gateway--An xGCP media gateway that is customer premises equipment and has one or more connections to the VoIP network as well as to user telephony equipment.

RGW--Residential gateway.

RSC--Router switch controller.

RSIP--Restart In Progress command. An MGCP or SGCP command that is used by the gateway to signal that an endpoint, or a group of endpoints, is being taken in or out of service.

RTCP--RTP Control Protocol. The protocol monitors an RTP connection and conveys information about the on-going session.

RTP--Real-Time Transport Protocol. The protocol provides end-to-end network transport functions for applications transmitting real-time data and services such as payload type identification, sequence numbering, timestamping, and delivery monitoring.

SDP--Session Description Protocol.Messages containing capabilities information that are exchanged between gateways.

SGCP--Simple Gateway Control Protocol. Controls Voice-over-IP gateways using an external call control element (called a call agent). SGCP is used to establish, maintain, and disconnect calls across an IP network.

SIP--Session Initiation Protocol.

SPVC--Soft Permanent Virtual Circuit

SVC--Switched Virtual Circuit

T1--24 64-Kbps time slots on a 1.544-Mbps serial interface.

TGW--see Trunking Gateway. Trunking gateway, also called the trunk side gateway. An xGCP media gateway that provides PSTN/IP gateway functionality.

trunking gateway--An xGCP media gateway that provides connectivity between the PSTN and VoIP networks. An external gateway control protocol (xGCP) media gateway that provides connectivity between the PSTN and VoIP networks.

TSE--Inband Telephony Signaling Events

VAD--Voice Activity Detection

VCC--Virtual Channel Connection (used where it may be a PVC, SPVC, or SVC)

VoIP--Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTS-like functionality, reliability, and voice quality. VoIP is a blanket term that generally refers to the Cisco standards-based approach (for example, H.323) to IP voice traffic.

VToA--Voice Trunking on ATM.

XCSP--External Call Service Provider. Subsystem that interoperates with external call protocols to provide services such as modem call setup and teardown.