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Table Of Contents
Voice over IP for the Cisco AS5800
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring IP Networks for Real-Time Voice Traffic
Configuring Custom Queuing and IP RTP Reserve
Verifying Voice Port Configuration
Inbound versus Outbound Dial Peers
Outbound Dialing on POTS Peers
Direct Inward Dial for POTS Peers
Distinguishing Voice and Modem Calls on the Cisco AS5800
Verifying Dial Peer Configuration
Configuring the Cisco AS5800 as an H.323 Gateway
Verifying Gateway Interface Configuration
Configuring the Cisco AS5800 for Interactive Voice Response
Configuring the Cisco 3640 as a Gatekeeper
Configuring the Cisco 2600 as a Gateway
Configuring the Cisco AS5800 as a Gateway
Voice over IP for the Cisco AS5800
The Voice over IP for the Cisco AS5800 feature adds Voice over IP carrier-class gateway functionality to the Cisco AS5800 platform. This document contains the following sections:
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Supported Standards, MIBs, and RFCs
Feature Overview
Voice over IP (VoIP) enables a Cisco AS5800 universal access server to provide voice and fax traffic, such as telephone calls and faxes, over an IP network. There are basically two different environments in which VoIP can be deployed: enterprise and service provider. Different strategies have been developed for deploying VoIP in both of these environments. The Cisco AS5800 universal access server can be configured for deployment in either an enterprise or a service provider environment but, because of the extensive capabilities of the Cisco AS5800 universal access server, it is more likely that it will function as a carrier class gateway in a service provider environment. This document, then, describes how to configure the Cisco AS5800 universal access server to act as a carrier class gateway in your VoIP network. To configure the Cisco AS5800 universal access server to perform in an enterprise environment, refer to the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module. The configuration steps for both the Cisco AS5300 access server and the Cisco AS5800 universal access server for an enterprise environment are identical.
Voice over IP in either the service provider or enterprise environment is primarily a software feature; however, to use this feature on the Cisco AS5800, you must install a VoIP feature card (VFC). The VFC uses the Cisco AS5800's T1/E1 and T3 Public Switched Telephone Network (PSTN) interfaces and local-area network (LAN) or wide-area network (WAN) routing capabilities to provide up to a 192 ports or channels (per VFC card) for VoIP packetized voice traffic.
Benefits
Two-Stage-Dial Toll Bypass
With Voice over IP on the Cisco AS5800, you can leverage your network's WAN infrastructure to offer long distance toll bypass services. Toll bypass occurs in two stages. For example, customers can be assigned an account number and a Personal Identification Number (PIN). When a user dials a local number or a 1-800-Internet Telephone Service Provider (ITSP) number, she connects to the local VoIP point of presence. She is then prompted by the Interactive Voice Response (IVR) to input her account and PIN numbers. Following authentication, a second dial tone allows her to enter an E.164 destination telephone number.
The local gatekeeper maps the E.164 destination telephone number to an IP address of a remote-zone gatekeeper, which then selects a gateway to terminate the call. The gateway encodes the call, encapsulates it in Real Time Protocol (RTP) packets and routes it over the WAN to the remote gateway. The remote gateway decodes the call and delivers it to the receiver.
For information about configuring IVR, refer to the Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms feature module.
Figure 1 Two-Stage Dial Toll Bypass
PSTN Voice-Traffic and Fax-Traffic Off load
Carriers can leverage their WAN infrastructure to off load voice and fax traffic from their congested PSTN networks by using the Cisco AS5800 as a carrier class voice gateway. In this application, PSTN traffic designated to be off-loaded is forwarded to a tandem switch connected to the Cisco AS5800 gateway. The AS5800 gateway then encapsulates the off-loaded PSTN traffic into RTP streams and routes it across the WAN.
The signaling interface between the PSTN and the Cisco AS5800 can be either Common Channel Signaling (CCS), with SS7 terminated by the VCO-4K service point or Channel Associated Signaling (CAS), configured in Direct Inward Dial (DID) mode. illustrates this application.
Figure 2 VoIP Used as a PSTN Gateway to Off load Voice Traffic and Fax Traffic
Universally Accessible Voice-Mail and Fax-Mail Services
VoIP on the Cisco AS5800 can be used to leverage the technology prefixes feature. Gateways (with voice/fax feature cards) that are connected to the voice-mail and fax-mail servers can be identified by gatekeepers based on a prefix prepended to an E.164 telephone number.
Additional Benefits
VoIP on the Cisco AS5800 can be used to provide the following additional benefits:
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Restrictions
To run Voice over IP on the Cisco AS5800, the AS5800 must have a version of the Cisco IOS software installed that supports DSDWare 3.1.7 (for example, Cisco IOS Release 12.0(4)XL or Cisco IOS Release 12.0(7)T).
Related Features and Technologies
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Related Documents
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Supported Platforms
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Supported Standards, MIBs, and RFCs
Standards
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MIBs
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For descriptions of supported MIBs and how to use MIBs, see the Cisco MIB web site on CCO at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
RFCs
None
Prerequisites
Before you can configure your Cisco AS5800 to use Voice over IP, you must first:
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For more information about any of the these configuration tasks, refer to the Cisco AS5800 Universal Access Server Software Installation and Configuration Guide, which shipped with your Cisco AS5800 and is available on the document CD-ROM.
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Configuration Tasks
After you have analyzed your dial plan and decided how to integrate it into your existing IP network, you are ready to configure your network devices to support Voice over IP. The actual configuration procedure depends entirely on the topology of your voice network, but, in general, you need to complete the following tasks:
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Configuring IP Networks for Real-Time Voice Traffic
You need to have a well-engineered network end-to-end when running delay-sensitive applications such as VoIP. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward Quality of Service (QoS). It is beyond the scope of this document to explain the specific details relating to wide-scale QoS deployment. Cisco IOS software provides many tools for enabling QoS on your backbone, such as Random Early Detection (RED), Weighted Random Early Detection (WRED), Fancy Queuing (meaning custom, priority, or weighted fair queuing), and IP Precedence. To configure your IP network for real-time voice traffic, you need to take into consideration the entire scope of your network, then select the appropriate QoS tool or tools. In addition, you must use the Cisco IOS ip cef command to ensure that Cisco Express Forwarding (CEF) is enabled.
QoS must be configured throughout your network—not just on the Cisco AS5800 devices running VoIP—to improve voice network performance. Not all QoS techniques are appropriate for all network routers. Edge routers and backbone routers in your network do not necessarily perform the same operations; the QoS tasks they perform might also differ. To configure your IP network for real-time voice traffic, you need to consider the functions of both edge and backbone routers in your network, then select the appropriate QoS tool or tools.
In general, edge routers perform the following QoS functions:
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In general, backbone routers perform the following QoS functions:
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Scalable QoS solutions require cooperative edge and backbone functions.
Configuring Custom Queuing and IP RTP Reserve
Although not required, you can use the custom queuing QoS tool to fine-tune your network for real-time voice traffic. Real-time voice traffic is carried on UDP ports ranging from 16384 to 32767. Custom Queuing and other methods for identifying high priority streams should be configured for these port ranges. For more information about custom queuing, refer to the "Congestion Management" chapter in the Cisco IOS Release 12.0 Quality of Service Configuration Guide. For more information about configuring IP RTP Priority, refer to the Cisco IOS Release 12.0(5)T IP RTP Priority feature module.
Configuring Voice Ports
When an ISDN interface on the Cisco AS5800 is carrying voice data, it is referred to as a voice port.
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Signaling in Voice over IP for the AS5800 is handled by ISDN PRI group configuration. After ISDN PRI is configured for both B and D channels for both ISDN PRI lines, you need to issue the isdn incoming-voice command on the serial interface (acting as the D channel) to ensure a dial tone.
Under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, you might need specific voice-port values configured, depending on the specifications of the devices in your telephony network. For more information on specific voice-port configuration commands, refer to either the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module or the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference.
To configure basic ISDN parameters for Voice over IP on the Cisco AS5800, perform the following steps:
As mentioned, under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. If you need to configure specific voice port parameters, perform the following steps beginning in privileged EXEC mode:
Fine-Tuning ISDN Voice Ports
Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation for voice ports. Collectively, these commands are referred to as voice-port tuning commands.
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To fine-tune ISDN voice ports, use the following commands beginning in privileged EXEC mode:
For more information on specific voice-port configuration commands or additional voice-port commands, refer to either the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module or the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference..
Verifying Voice Port Configuration
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Troubleshooting Tips
If you are having trouble connecting a call, and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:
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Configuring Dial Peers
The key point to understanding how VoIP functions is to understand dial peers. Each dial peer defines the characteristics associated with a call leg, as shown in and . A call leg is a discrete segment of a call connection that lies between two points in the connection. All of the call legs for a particular connection have the same connection ID.
There are two different kinds of dial peers:
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An end-to-end call comprises four call legs, two from the perspective of the source access server as shown in , and two from the perspective of the destination access server as shown in . A dial peer is associated with each call leg. Dial peers are used to apply attributes to call legs and to identify call origin and destination. Attributes applied to a call leg include QoS, codec, VAD, and fax rate.
Figure 3 Dial Peer Call Legs from the Perspective of the Source Router
Figure 4 Dial Peer Call Legs from the Perspective of the Destination Router
Inbound versus Outbound Dial Peers
Dial peers are used for both inbound and outbound call legs. It is important to remember that these terms are defined from the access server's perspective. An inbound call leg originates outside the access server. An outbound call leg originates from the access server.
For inbound call legs, a dial peer might be associated to the calling number or the port designation. Outbound call legs always have a dial peer associated with them. The destination pattern is used to identify the outbound dial peer. The call is associated with the outbound dial peer at setup time.
POTS peers associate a telephone number with a particular voice port so that incoming calls for that telephone number can be received and outgoing calls can be placed. VoIP peers point to specific devices (by associating destination telephone numbers with a specific IP address) so that incoming calls can be received and outgoing calls can be placed. Both POTS and VoIP peers are needed to establish VoIP connections.
Configuring POTS Peers
POTS peers enable incoming calls to be received by a particular telephony device. To configure a POTS peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its telephone numbers, and associate it with a voice port through which calls will be established. Under most circumstances, the default values for the remaining dial peer configuration commands will be sufficient to establish connections.
To configure a POTS dial peer, use the following commands beginning in global configuration mode:
For additional POTS dial-peer configuration commands, refer to the "Voice-Related Commands" section of the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference, the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module, and the Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module.
Outbound Dialing on POTS Peers
When a router receives a voice call, it selects an outbound dial peer by comparing the called number (the full E.164 telephone number) in the call information with the number configured as the destination pattern for the POTS peer. The router then strips out the left-justified numbers corresponding to the destination pattern matching the called number. If you have configured a prefix, the prefix will be put in front of the remaining numbers, creating a dial string, which the router will then dial. If all numbers in the destination pattern are stripped-out, the user will receive (depending on the attached equipment) a dial tone.
For example, suppose there is a voice call whose E.164 called number is 1 310 767-2222. If you configure a destination-pattern of "1310767" and a prefix of "9," the router will strip out "1310767" from the E.164 telephone number, leaving the extension number of "2222." It will then append the prefix, "9," to the front of the remaining numbers, so that the actual numbers dialed is "9, 2222." The comma in this example means that the router will pause for one second between dialing the "9" and the "2" to allow for a secondary dial tone.
Direct Inward Dial for POTS Peers
Direct inward dial (DID) is used to determine how the called number is treated for incoming POTS call legs. As shown in , incoming means from the perspective of the router. In this case, it is the call leg coming into the access server to be forwarded through to the appropriate destination pattern.
Figure 5 Incoming and Outgoing POTS Call Legs
Unless otherwise configured, when a call arrives on the access server, the server presents a dial tone to the caller and collects digits until it can identify the destination dial peer. After the dial peer is identified, the call is forwarded through the next call leg to the destination.
There are cases where it might be necessary for the server to use the called-number (DNIS) to find a dial peer for the outgoing call leg—for example, if the switch connecting the call to the server has already collected the digits. DID enables the server to match the called-number with a dial peer and then directly place the outbound call. With DID, the server does not present a dial tone to the caller and does not collect digits; it forwards the call directly to the configured destination.
To use DID and incoming called-number, a dial peer must be associated with the incoming call leg. Before doing this, it helps if you understand the logic behind the algorithm used to associate the incoming call leg with the dial peer.
The algorithm used to associate incoming call legs with dial peers uses three inputs (which are derived from signaling and interface information associated with the call) and four defined dial peer elements. The three signaling inputs are:
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The four defined dial peer elements are:
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Using the elements, the algorithm is as follows:
For all peers where call type (VoIP versus POTS) match dial peer type:if the type is matched, associate the called number with the incoming called-numberelse if the type is matched, associate calling-number with answer-addresselse if the type is matched, associate calling-number with destination-patternelse if the type is matched, associate voice port to portThis algorithm shows that if a value is not configured for answer-address, the origin address is used because, in most cases, the origin address and answer-address are the same.
To configure a POTS dial peer for direct inward dial, use the following commands beginning in global configuration mode:
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Distinguishing Voice and Modem Calls on the Cisco AS5800
When the Cisco AS5800 is handling both modem and voice calls, it needs to be able to identify the service type of the call—that is, whether or not the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the called-number (DNIS). In a mixed environment, where the server receives both modem and voice calls, you need to identify the service type of a call by using the incoming called-number command.
Without this, the server attempts to resolve whether an incoming call is a modem or voice call based on the interface over which the call comes. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.
It helps to understand the logic behind the algorithm the system uses to distinguish voice and modem calls. The algorithm is as follows:
If the called-number matches a number from the modem pool, handle the call as a modem callIf the called-number matches a configured dial peer incoming called number, handle the call as a voice callElse handle the call as a modem call by default modem poolIf there is no called-number information configured, call classification is handled as follows:
If the interface matches the interface configured for the modem pool, handle the call as a modem call.If the voice port matches the one configured as the dial peer port, handle the call as a voice callElse handle the call as a modem call by default modem poolTo identify the service type of a call to be voice, use the following commands beginning in global configuration mode:
Configuring VoIP Peers
VoIP peers enable outgoing calls to be made from a particular telephony device. To configure a VoIP peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its destination telephone number and destination IP address. As with POTS peers, under most circumstances, the default values for the remaining dial peer configuration commands will be adequate to establish connections.
To configure a VoIP peer, use the following commands beginning in global configuration mode:
For additional VoIP dial peer configuration options, refer to the "Voice-Related Commands" section of the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference, the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module, and the Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module.
Verifying Dial Peer Configuration
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Troubleshooting Tips
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Configuring the Cisco AS5800 as an H.323 Gateway
The Service Provider implementation of Voice over IP uses both gatekeepers and gateways. Because of the extensive capabilities of the Cisco AS5800 universal access server, it is likely that it will function as a carrier class gateway in a Service Provider environment. The final step in configuring the Cisco AS5800 for Voice over IP functionality is to configure one of its interfaces as a gateway interface. You can use either an interface that is connected to the gatekeeper or a loopback interface for the gateway interface. The interface that is connected to the gatekeeper is usually a LAN interface—Fast Ethernet, Ethernet, FDDI, or Token Ring.
To configure a gateway interface, perform the following steps beginning in the global configuration mode:
For more information about configuring gateways and gatekeepers, refer to the Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module.
Verifying Gateway Interface Configuration
Use the show gateway command to find the current registration information and status of the gateway.
Configuring the Cisco AS5800 for Interactive Voice Response
The Interactive Voice Response (IVR) Service Provider application provides IVR capabilities using Tool Command Language (TCL) scripts. For example, an IVR script is played when a caller receives a voice-prompt instruction to enter a specific type of information, such as a PIN. After playing the voice prompt, the IVR application collects the predetermined number of touch tones (digit collection) and forwards the collected digits to a server for storage and retrieval. Call records can be kept, and a variety of accounting functions performed.
Available IVR Scripts
The following is a description of the available IVR scripts:
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The caller can interrupt the message by entering digits for the account number which will trigger the prompt to enter the PIN number. If authentication fails the third time, the script plays the audio file auth_fail_final.au, then hangs up.
Configuring IVR
To use IVR with scripts, you need to configure the inbound POTS dial peer to support IVR, as well as enable IVR functionality by using the call application global configuration. To configure IVR, use the following commands beginning in the global configuration mode:
For more information about configuring IVR, refer to the Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms feature module.
Verifying IVR Configuration
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Configuration Example
The following configuration example shows an abbreviated configuration using a Cisco 2600 router and a CiscoAS5800 universal access server as gateways and a Cisco 3600 router as a gatekeeper. shows the network diagram for this particular scenario.
Figure 6 AS5800 Universal Access Server Acting as a Gateway
Configuring the Cisco 3640 as a Gatekeeper
! Configure the Ethernet interface to be used at the gatekeeper interface.interface Ethernet0/1ip address 172.30.00.00 255.255.255.0no ip directed-broadcastno logging event link-statusno keepalive!! Configure the gatekeeper interface and enable the interface.gatekeeperzone local gk3.gg-dn1 gg-dn1 173.50.00.00zone prefix gk3.gg-dn1 21*gw-type-prefix 9#* gw ipaddr 173.60.0.0 1720gw-type-prefix 6#* gw ipaddr 173.60.0.199 1720no use-proxy gk3.gg-dn1 default inbound-to terminalno shutdown!Configuring the Cisco 2600 as a Gateway
! Configure POTS and VoIP dial peers.dial-peer voice 88 voipdestination-pattern 11111tech-prefix 9#session ras!dial-peer voice 11 potsincoming called-number 11111destination-pattern 6#12345port 1/1/1prefix 12345!! Configure the gateway interface.interface Ethernet0/0ip address 173.60.0.199 255.255.255.0no ip directed-broadcastno ip mroute-cacheno logging event link-statusno keepaliveno cdp enabledh323-gateway voip interfaceh323-gateway voip id gk3.gg-dn1 ipaddr 173.30.0.0 1719h323-gateway voip h323-id gw6@gg-dn1h323-gateway voip tech-prefix 6#!Configuring the Cisco AS5800 as a Gateway
! Configure the T1 controller. (This configuration is for a T3 card.)controller T1 1/0/0:1framing esflinecode b8zspri-group timeslots 1-24!! Configure POTS and VoIP dial peers.dial-peer voice 11111 potsincoming called-number 12345destination-pattern 9#11111direct-inward-dialport 1/0/0:1:Dprefix 11111!dial-peer voice 12345 voipdestination-pattern 12345tech-prefix 6#session target ras!! Enable gateway functionality.gateway!! Enable Cisco Express Forwarding.ip cef!! Configure and enable the gateway interface.interface FastEthernet0/3/0ip address 173.60.0.0.255.255.255.0no ip directed-broadcast<no keepalive
