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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.

Cisco Unified Border Element Standards Compliance

This Cisco Unified Border Element is a special Cisco IOS software image that provides a network-to-network interface point for billing, security, call admission control, quality of service, and signaling interworking. This chapter describes basic gateway functionality, software images, topology, and summarizes supported features.

UnnamedNote

Cisco Product Authorization Key (PAK)--A Product Authorization Key (PAK) is required to configure some of the features described in this guide. Before you start the configuration process, please register your products and activate your PAK at the following URL http://www.cisco.com/go/license .

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.

Cisco Unified Border Element Cisco UBE Standards Compliance Features

This chapter contains the following configuration topics:

Cisco UBE Prerequisites and Restrictions

Prerequisites for Cisco Unified Border Element

Restrictions for Cisco Unified Border Element

Cisco UBE Standards Compliance

ENUM Support (RFC2916)

SIP--RFC 2782 Compliance with DNS SRV Queries

SIP - DNS SRV RFC2782 Compliance

SIP-to-SIP Extended Feature Functionality for Session Border Controllers

The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs). The SIP-to-SIP Extended Feature Functionality includes:

Call Admission Control (based on CPU, memory, and total calls)

Delayed Media Call

ENUM support

Configuring SIP Error Message Pass Through

Interoperability with Cisco Unified Communications Manager 5.0 and BroadSoft

Lawful Intercept

Media Inactivity

Modem Passthrough over VoIP

TCP and UDP interworking

Tcl scripts with SIP NOTIFY VoiceXML with SIP-to-SIP

Transport Layer Security (TLS)

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 SIP-to-SIP Extended Feature Functionality for Session Border Controllers

Cisco Unified Border Element

Cisco IOS Release 12.4(6)T or a later release must be installed and running on your Cisco Unified Border Element.

Cisco Unified Border Element (Enterprise)

Cisco IOS XE Release 3.1S or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Modem Passthrough over VoIP

The Modem Passthrough over VoIP feature provides the transport of modem signals through a packet network by using pulse code modulation (PCM) encoded packets.

Prerequisites for the Modem Passthrough over VoIP Feature

VoIP enabled network.

Cisco IOS Release 12.1(3)T must run on the gateways for the Modem Passthrough over VoIP feature to work.

Network suitability to pass modem traffic. The key attributes are packet loss, delay, and jitter. These characteristics of the network can be determined by using the Cisco IOS feature Service Assurance Agent.

Cisco Unified Border Element

Cisco IOS Release 12.4(6)T or a later release must be installed and running on your Cisco Unified Border Element.

Cisco Unified Border Element (Enterprise)

Cisco IOS XE Release 3.3S or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Restrictions for the Modem Passthrough over VoIP Feature

Cisco Unified Border Element (Enterprise)

If call started as g729, upon modem tone (2100Hz) detection both the outgoing gateway (OGW) and the trunking gateway (TGW) will genearate NSE packets towards peer side and up speed to g711 as Cisco UBE(Enterprise) passes these packets to the peer side.

UnnamedNote

That OGW and TGW display the new codec, but the Cisco UBE (Enterprise) continues to show the original codec g729 in the show commands.

Information about Configuring Modem Passthrough over VoIP

The Modem Passthrough over VoIP feature performs the following functions:

Represses processing functions like compression, echo cancellation, high-pass filter, and voice activity detection (VAD).

Issues redundant packets to protect against random packet drops.

Provides static jitter buffers of 200 milliseconds to protect against clock skew.

Discriminates modem signals from voice and fax signals, indicating the detection of the modem signal across the connection, and placing the connection in a state that transports the signal across the network with the least amount of distortion.

Reliably maintains a modem connection across the packet network for a long duration under normal network conditions.

For further details, the functions of the Modem Passthrough over VoIP feature are described in the following sections.

Modem Tone Detection

The gateway is able to detect modems at speeds up to V.90.

Passthrough Switchover

When the gateway detects a data modem, both the originating gateway and the terminating gateway roll over to G.711. The roll over to G.711 disables the high-pass filter, disables echo cancellation, and disables VAD. At the end of the modem call, the voice ports revert to the prior configuration and the digital signal processor (DSP) goes back to the state before switchover. You can configure the codec by selecting the g711alaw or g711ulaw option of the codec command.

See also the How to Configure Modem Passthrough over VoIP section in this document.

Controlled Redundancy

You can enable payload redundancy so that the Modem Passthrough over VoIP switchover causes the gateway to emit redundant packets.

Packet Size

When redundancy is enabled, 10-ms sample-sized packets are sent. When redundancy is disabled, 20-ms sample-sized packets are sent.

Clock Slip Buffer Management

When the gateway detects a data modem, both the originating gateway and the terminating gateway switch from dynamic jitter buffers to static jitter buffers of 200-ms depth. The switch from dynamic to static is to compensate for Public Switched Telephone Network (PSTN) clocking differences at the originating gateway and the terminating gateway. At the conclusion of the modem call, the voice ports revert to dynamic jitter buffers.

The figure below illustrates the connection from the client modem to a MICA technologies modem network access server (NAS).

Figure 1
Modem Passthrough Connection

How to Configure Modem Passthrough over VoIP

You can configure the Modem Passthrough over VoIP feature on a specific dial peer in two ways, as follows:

Globally in the voice-service configuration mode

Individually in the dial-peer configuration mode on a specific dial peer

By default, modem passthrough over VoIP capability and redundancy are disabled.

UnnamedTip

You need to configure modem passthrough in both the originating gateway and the terminating gateway for the Modem Passthrough over VoIP feature to operate. If you configure only one of the gateways in a pair, the modem call will not connect successfully.

Redundancy can be enabled in one or both of the gateways. When only a single gateway is configured for redundancy, the other gateway receives the packets correctly, but does not produce redundant packets.

See the following sections for the Modem Passthrough over VoIP feature. The two configuration tasks can configure separately or together. If both are configured, the dial-peer configuration takes precedence over the global configuration. Consequently, a call matching a particular dial-peer will first try to apply the modem passthrough configuration on the dial-peer. Then, if a specific dial-peer is not configured, the router will use the global configuration:

Configuring Modem Passthrough over VoIP Globally

For the Modem Passthrough over VoIP feature to operate, you need to configure modem passthrough in both the originating gateway and the terminating gateway so that the modem call matches a voip dial-peer on the gateway.

The default behavior for the voice-service configuration mode is no modem passthrough. This default behavior implies that modem passthrough is disabled for all dial peers on the gateway by default.

When using the voice service voip and modem passthrough nse commands on a terminating gateway to globally set up fax or modem passthrough with NSEs, you must also ensure that each incoming call will be associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate calls with dial peers by using the incoming called-number command to specify a sequence of digits that incoming calls can match.

To configure the Modem Passthrough over VoIP feature for all the connections of a gateway, use the following commands beginning in global configuration mode:

SUMMARY STEPS

1.    enable

2.    voice service voip

3.    modem passthrough nse [payload-type number] codec {g711ulaw | g711alaw} [redundancy] [maximum-sessions value]

4.    exit

5.    exit

DETAILED STEPS

Step 1 enable

Enables privileged EXEC mode.

Enter your password if prompted.

Example:

Device> enable

Step 2 voice service voip

Enters voice-service configuration mode.

Configures voice service for all the connections for the gateways.

Example:

Device(config)# voice service voip

Step 3 modem passthrough nse [payload-type number] codec {g711ulaw | g711alaw} [redundancy] [maximum-sessions value]

Configures the Modem Passthrough over VoIP feature The default behavior is no modem passthrough.

The payload type is an optional parameter for the nse keyword. Use the same payload-type number for both the originating gateway and the terminating gateway. The payload-type numbercan be set from 96 to 119. If you do not specify the payload-type number, the numberdefaults to 100. When the payload-type is 100, and you use the show running-configcommand, the payload-type parameter does not appear.

Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.

The redundancy keyword is an optional parameter for sending redundant packets for modem traffic.

The maximum-sessions keyword is an optional parameter for the redundancykeyword. This parameter determines the maximum simultaneous modem passthrough sessions with redundancy.

Example:

Device(config)# Router(conf-voi-serv)# modem passthrough nse payload-type 97 codec g711alaw redundancy maximum-sessions 3

Step 4 exit

Exits voice-service configuration mode.

Example:

Device(conf-voi-serv)# exit

Step 5 exit

Exits global configuration mode.

Example:

Device(config)# exit

Configuring Modem Passthrough over VoIP for a Specific Dial Peer

To enable Modem Passthrough on the VoIP dial peers on both the originating and terminating gateway, configure modem passthrough globally or explicitly on the dial peer.

For modem passthrough to operate, you must define VoIP dial peers on both gateways to match the call, for example, by using a destination pattern or an incoming called number. The modem passthrough parameters associated with those dial peers then will apply to the call.

UnnamedNote

When modem passthrough is configured individually for a specific dial peer, that configuration for the specific dial peer takes precedence over the global configuration.

To configure the Modem Passthrough over VoIP feature for a specific dial peer, use the following commands beginning in global configuration mode:

SUMMARY STEPS

1.    enable

2.    dial-peer voice number voip

3.    modem passthrough {system | nse [payload-type number] codec {g711ulaw | g711alaw}[redundancy]}

4.    exit

5.    exit

DETAILED STEPS

Step 1 enable

Enables privileged EXEC mode.

Enter your password if prompted.

Example:

Device> enable

Step 2 dial-peer voice number voip

Enters dial-peer configuration mode.

Configures a specific dial peer in dial-peer configuration mode.

Example:

Device(config)# dial-peer voice 5 voip

Step 3 modem passthrough {system | nse [payload-type number] codec {g711ulaw | g711alaw}[redundancy]}

Configures the Modem Passthrough over VoIP feature for a specific dial peer. The default behavior for the Modem Passthrough for VoIP feature in dial-peer configuration mode is modem passthrough system. As required, the gateway defaults to no modem passthrough.

When the system keyword is enabled, the following parameters are not available: nse, payload-type, codec, and redundancy. Instead the values from the global configuration are used.

The payload type is an optional parameter for the nse keyword. Use the same payload-type number for both the originating gateway and the terminating gateway. The payload-type numbercan be set from 96 to 119. If you do not specify the payload-type number, the numberdefaults to 100. When the payload-type is 100, and you use the show running-configcommand, the payload-type parameter does not appear.

Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.

The redundancy keyword is an optional parameter for sending redundant packets for modem traffic.

Example:

Device(config-dial-peer)# modem passthrough nse payload-type 97 codec g711alaw redundancy

Step 4 exit

Exits dial-peer configuration mode and returns to the global configuration mode.

Example:

Device(config-dial-peer)# exit

Step 5 exit

Exits global configuration mode.

Example:

Device(config)# exit

Troubleshooting Tips

To troubleshoot the Modem Passthrough over VoIP feature, perform the following steps:

Make sure that you can make a voice call.

Make sure that Modem Passthrough over VoIP is configured on both the originating gateway and the terminating gateway.

Make sure that both the originating gateway and the terminating gateway have the same named signaling event (NSE) payload-type number.

Make sure that both the originating gateway and the terminating gateway have the same maximum-sessions value when the two gateways are configured in the voice-service configuration mode.

Use the debug vtsp dsp and debug vtsp session commands to debug a problem.

Verifying Modem Passthrough over VoIP

To verify that the Modem Passthrough over VoIP feature is enabled, perform the following steps:

SUMMARY STEPS

1.    Enter the show run command to verify the configuration.

2.    Enter the show dial-peer voice command to verify that Modem Passthrough over VoIP is enabled.

DETAILED STEPS

Step 1 Enter the show run command to verify the configuration.

Step 2 Enter the show dial-peer voice command to verify that Modem Passthrough over VoIP is enabled.

Monitoring and Maintaining Modem Passthrough over VoIP

To monitor and maintain the Modem Passthrough over VoIP feature, use the following commands in privileged EXEC mode:

Command

Purpose

Device# show call active voice brief

Displays information for the active call table or displays the voice call history table. The brief option displays a truncated version of either option.

Device# show dial-peer voice 15 summary

Displays configuration information for dial peers. The number argument specifies a specific dial peer from 1 to 32767. The summary option displays a summary of all dial peers.

Configuration Examples

The following is sample configuration for the Modem Passthrough over VoIP feature:

version 12.1
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
voice service voip
	 modem passthrough nse codec g711ulaw redundancy maximum-session 5
!
!
resource-pool disable
!
!
!
!
!
ip subnet-zero
ip ftp source-interface Ethernet0
ip ftp username lab
ip ftp password lab
no ip domain-lookup
!
isdn switch-type primary-5ess
cns event-service server
!
!
!
!
!
mta receive maximum-recipients 0
!
!
controller T1 0
 framing esf
 clock source line primary
 linecode b8zs
 pri-group timeslots 1-24
!
controller T1 1
 shutdown
 clock source line secondary 1
!
controller T1 2
 shutdown
!
controller T1 3
 shutdown
!
!
!
interface Ethernet0
 ip address 1.1.2.2 255.0.0.0
 no ip route-cache
 no ip mroute-cache
!
interface Serial0:23
 no ip address
 encapsulation ppp
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-5ess
 isdn incoming-voice modem
 no peer default ip address
 no fair-queue
 no cdp enable
 no ppp lcp fast-start
!
interface FastEthernet0
 ip address 26.0.0.1 255.0.0.0
 no ip route-cache
 no ip mroute-cache
 load-interval 30
 duplex full
 speed auto
 no cdp enable
!
ip classless
ip route 17.18.0.0 255.255.0.0 1.1.1.1
no ip http server
!
!
!
!
voice-port 0:D
!
dial-peer voice 1 pots
 incoming called-number 55511..
 destination-pattern 020..
 direct-inward-dial
 port 0:D
 prefix 020
!
dial-peer voice 2 voip
 incoming called-number 020..
 destination-pattern 55511..
 modem passthrough nse codec g711ulaw redundancy
 session target ipv4:26.0.0.2
!
!
line con 0
 exec-timeout 0 0
 transport input none
line aux 0
line vty 0 4
 login
!
!
end

Feature Information for SIP-to-SIP Extended Feature Functionality for Session Border Controllers

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 1
Feature Information for Configuring SIP-to-SIP Extended Feature Functionality for Session Border Controllers

Feature Name
Releases
Feature Information

SIP-to-SIP Extended Feature Functionality for Session Border Controllers

12.4(6)T

The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs).

In Cisco IOS Release 12.4(6)S, this feature was implemented on the Cisco Unified Border Element

The following commands were introduced or modified: modem passthrough (dial-peer); modem passthrough (voice-service); show call active voice voice; show call history voice voice; show dial-peer voice; voice service.

SIP-to-SIP Extended Feature Functionality for Session Border Controllers

Cisco IOS XE Release 3.1S

Cisco IOS XE Release 3.3S

The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs).

In Cisco IOS Release 12.4(6)S, this feature was implemented on the Cisco Unified Border Element (Enterprise).

The following commands were introduced or modified: modem passthrough (dial-peer); modem passthrough (voice-service); show call active voice voice; show call history voice voice; show dial-peer voice; voice service.

SIP RFC 2782 Compliance with DNS SRV Queries

Effective with Cisco IOS XE Release 2.5, the Domain Name System Server (DNS SRV) query used to determine the IP address of the user endpoint is modified in compliance with RFC 2782 (which supersedes RFC 2052). The DNS SRV query prepends the protocol label with an underscore "_" character to reduce the risk of duplicate names being used for unrelated purposes. The form compliant with RFC 2782 is the default style.

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 SIP RFC 2782 Compliance with DNS SRV Queries

Cisco Unified Border Element

Cisco IOS Release 12.2(8)T or a later release must be installed and running on your Cisco Unified Border Element.

Cisco Unified Border Element (Enterprise)

Cisco IOS XE Release 2.5 or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Information SIP RFC 2782 Compliance with DNS SRV Queries

Session Initiation Protocol (SIP) on Cisco VoIP gateways uses the DNS SRV query to determine the IP address of the user endpoint. The query string has a prefix in the form of "protocol.transport." and is attached to the fully qualified domain name (FQDN) of the next hop SIP server. This prefix style originated in RFC 2052. Beginning with Cisco IOS XE Release 2.5, a second style, in compliance with RFC 2782, prepends the protocol label with an underscore "_"; for example, "_protocol._transport." The addition of the underscore reduces the risk of the same name being used for unrelated purposes. The form compliant with RFC 2782 is the default style.

How to Configure SIP-RFC 2782 Compliance with DNS SRV Queries

Configuring DNS Server Query Format RFC 2782 Compliance with DNS SRV Queries

Configuring DNS Server Query Format RFC 2782 Compliance with DNS SRV Queries

Compliance with RFC 2782 changes the DNS SVR protocol label style. RFC 2782 updates RFC 2052 by prepending the protocol label with an underscore character. The prefix format compliant with RFC 2782 is the default format. However, backward compatibility is available, allowing newer versions of Cisco IOS software to work with older networks that support only RFC 2052 DNS SVR prefix style.

To configure the format of DNS SRV queries to comply with RFC 2782, complete this task.

UnnamedNote

You do not have to perform this task if you want to use the default RFC 2782 format.

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    interface type number

4.    sip-ua

5.    srv version {1 | 2}

6.    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 interface type number

Configures an interface type and enters interface configuration mode

Example:

Router(config)# interface gigabitethernet 0/0/0

Step 4 sip-ua

Enters SIP UA configuration mode.

Example:

Router(config-if)# sip-ua

Step 5 srv version {1 | 2}

Generates DNS SRV queries in either RFC 2782 or RFC 2052 format.

1 --The query is set to the domain name prefix of protocol.transport. (RFC 2052 style).

2 --The query is set to the domain name prefix of _protocol._transport. (RFC 2782 style). This is the default.

Example:

Router(config-sip-ua)# srv version 2

Step 6 exit

Exits the current configuration mode.

Example:

Router(config-sip-ua)# exit

Verifying

The following example shows sample is output from the show sip-ua status command used to verify the style of DNS server queries:

Router# show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent bind status(signaling): DISABLED
SIP User Agent bind status(media): DISABLED
SIP max-forwards : 6
SIP DNS SRV version: 1 (rfc 2052)

Feature Information for SIP RFC 2782 Compliance with DNS SRV Queries

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.

ISR feature history table entry

Table 2
Feature Information for SIP: RFC 2782 Compliance with DNS SRV Queries

Feature Name

Releases

Feature Information

SIP: RFC 2782 Compliance of DNS SRV Queries

12.2(8)T, 12.2(11)T, 12.2(15)T

Effective with Cisco IOS XE Release 2.5, the DNS SRV query used to determine the IP address of the user endpoint is modified in compliance with RFC 2782 (which supersedes RFC 2052). The DNS SRV query prepends the protocol label with an underscore "_" character to reduce the risk of duplicate names being used for unrelated purposes. The form compliant with RFC 2782 is the default style.

The following command was introduced or modified: srv version.

ASR feature history table entry

Table 3
Feature Information for SIP: RFC 2782 Compliance with DNS SRV Queries

Feature Name

Releases

Feature Information

SIP: RFC 2782 Compliance of DNS SRV Queries

Cisco IOS XE Release 2.5

Effective with Cisco IOS XE Release 2.5, the DNS SRV query used to determine the IP address of the user endpoint is modified in compliance with RFC 2782 (which supersedes RFC 2052). The DNS SRV query prepends the protocol label with an underscore "_" character to reduce the risk of duplicate names being used for unrelated purposes. The form compliant with RFC 2782 is the default style.

The following command was introduced or modified: srv version.