Dial Peer Configuration on Voice Gateway Routers, Cisco IOS Release 15M&T
Dial Peer Overview
Dial Peer Overview
Last Updated: December 3, 2012
Configuring dial peers is the key to implementing dial plans and providing voice services over an IP packet network. Dial peers are used to identify call source and destination endpoints and to define the characteristics applied to each call leg in the call connection.
This chapter contains the following sections:
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.
A traditional voice call over the public switched telephone network (PSTN) uses a dedicated 64K circuit end to end. In contrast, a voice call over the packet network is made up of discrete segments or call legs. A call leg is a logical connection between two routers or between a router and a telephony device. A voice call comprises four call legs, two from the perspective of the originating router and two from the perspective of the terminating router, as shown in the figure below.
A dial peer is associated with each call leg. Attributes that are defined in a dial peer and applied to the call leg include the codec, quality of service (QoS), voice activity detection (VAD), and fax rate. To complete a voice call, you must configure a dial peer for each of the four call legs in the call connection.
Depending on the call leg, a call is routed using one of the two types of dial peers:
Both POTS and voice-network dial peers are needed to establish voice connections over a packet network.
When a voice call comes into the router, the router must match dial peers to route the call. For inbound calls from a POTS interface that are being sent over the packet network, the router matches a POTS dial peer for the inbound call leg and a voice-network dial peer for the outbound call leg. For calls coming into the router from the packet network, the router matches an outbound POTS dial peer to terminate the call and an inbound voice-network dial peer for features such as codec, VAD, and QoS.
The figure below shows the call legs and associated dial peers necessary to complete a voice call.
The following configurations show an example of a call being made from 4085554000 to 3105551000. The figure below shows the inbound POTS dial peer and the outbound voice over IP (VoIP) dial peer that are configured on the originating router. The POTS dial peer establishes the source of the call (via the calling number or voice port), and the voice-network dial peer establishes the destination by associating the dialed number with the network address of the remote router.
In this example, the dial string 14085554000 maps to telephone number 555-4000, with the digit 1 plus the area code 408 preceding the number. When you configure the destination pattern, set the string to match the local dialing conventions.
The figure below shows the inbound VoIP dial peer and outbound POTS dial peer that are configured on the terminating router to complete the call. Dial peers are of local significance only.
In the previous configuration examples, the last four digits in the VoIP dial peer's destination pattern were replaced with wildcards. Which means that from Router A, calling any telephone number that begins with the digits "1310555" will result in a connection to Router B. This behavior implies that Router B services all numbers beginning with those digits. From Router B, calling any telephone number that begins with the digits "1408555" will result in a connection to Router A. This behavior implies that Router A services all numbers beginning with those digits.
The only exception to the previous example occurs when both POTS dial peers share the same router, as shown in the figure below. In this circumstance, you do not need to configure a voice-network dial peer.
This type of configuration is similar to the configuration used for hairpinning, which occurs when a voice call destined for the packet network is instead routed back over the PSTN because the packet network is unavailable.
POTS Dial Peers
POTS dial peers retain the characteristics of a traditional telephony network connection. POTS dial peers map a dialed string to a specific voice port on the local router, normally the voice port connecting the router to the local PSTN, PBX, or telephone.
Voice-Network Dial Peers
Voice-network dial peers are components on an IP network to which a voice gateway router points via the component's IP address specified in the session-target command for a particular matching dial peer. The four types of voice-network dial peers (VoIP, voice over ATM (VoATM), voice over Frame Relay (VoFR), and multimedia mail over IP (MMoIP)) are determined according to the given packet network technology and are described as follows:
Data Dial Peers
Before Cisco IOS Release 12.2(11)T, a Cisco voice gateway would try to match a voice dial peer before matching and processing a modem call. If a voice dial peer was matched, the call was processed as voice. If there was no voice dial peer match, only then was a call considered to be a modem call. Voice calls always received preference over modem calls. Also, there was no way to assign a subset of addresses in the numbering plan for data calls.
In Cisco IOS Release 12.2(11)T, an interim solution in the form of application called "data_dialpeer" was introduced to enable gateways to identify dial peers. The application enabled the handling of certain matched calls as modem calls. Refer to the Fine-Grain Address Segmentation in Dial Peers feature documentation in Cisco IOS Release 12.2(11)T for more information.
In Cisco IOS Release 12.2(13)T, formal support for data dial peers was released in the form of the Dial-Peer Support for Data Calls feature, which enables the configuration and order assignment of dial peers so that the gateway can identify incoming calls as voice or data (modem). You can use the dial-peer data and dial-peer search commands to perform this configuration. Refer to the "Data Dial Peers" section on page 33 for configuration steps and examples.
Creating a Dial Peer Configuration Table
Before you can configure dial peers, you must obtain specific information about your network. One way to identify this information is to create a dial peer configuration table. This table should contain all the telephone numbers and access codes for each router that is carrying telephone traffic in the network. Because most installations require integrating equipment into an existing voice network, the telephone dial plans are usually preset.
The figure below shows an example of a network in which Router A, with an IP address of 10.1.1.1, connects a small sales branch office to the main office through Router B, with an IP address of 10.1.1.2.
Three telephone numbers in the sales branch office need dial peers configured for them. Router B is the primary gateway to the main office; as such, it needs to be connected to the company's PBX. Four devices need dial peers, all of which are connected to the PBX, configured for them in the main office.
The table below shows the peer configuration table for the example in the figure above.
The term codec stands for coder-decoder . A codec is a particular method of transforming analog voice into a digital bit stream (and vice versa) and also refers to the type of compression used. Several different codecs have been developed to perform these functions, and each one is known by the number of the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) standard in which it is defined. For example, two common codecs are the G.711 and the G.729 codecs.
Codecs use different algorithms to encode analog voice into digital bit streams and have different bit rates, frame sizes, and coding delays associated with them. Codecs also differ in the amount of perceived voice quality they achieve. Specialized hardware and software in the digital signal processors (DSPs) perform codec transformation and compression functions, and different DSPs may offer different selections of codecs.
Select the same type of codec at both ends of the call. For instance, if a call was coded with a G.729 codec, it must be decoded with a G.729 codec. Codec choice is configured on dial peers.
The table below lists the H.323, SIP, and MGCP codecs that are supported for voice.
1 Annex A is used in the Cisco platforms that are supported in this software release.
2 For dynamic payload types.
For more information, refer to the " Dial Planning " chapter in this document and see the Cisco IOS Voice Port Configuration Guide.
Clear Channel (G.Clear) Codec
G.Clear guarantees bit integrity when transferring a DS-0 through a gateway server, supports the transporting of nonvoice circuit data sessions through a Voice over IP (VoIP) network, and enables the VoIP networks to transport ISDN and switched 56 circuit-switched data calls. With the availability of G.Clear, ISDN data calls that do not require bonding can be supported.
In a transit application, because it is possible to have a mix of voice and data calls, not supporting G.Clear limits the solution to voice-only calls. The end-user application is in charge of handling packet loss and error recovery. This packet loss management precludes the use of clear channel with some applications unless the IP network is carefully engineered.
In an MGCP environment, the voice gateway backhauls the public switched telephony network (PSTN) signaling channel to the call agent. The call agent examines the bearer capability and determines when a G.Clear call should be established.
Adaptive Differential PCM Voice Codec G.726
Adaptive differential pulse code modulation (ADPCM) voice codec operates at bit rates of 16, 24, and 32 kbps. ADPCM provides the following functionality:
The internet Low Bitrate Codec (iLBC) Has the following benefits:
Platforms that iLBC Supports
iLBC is supported on Cisco AS5350XM and Cisco AS5400XM Universal Gateways with Voice Feature Cards (VFCs) and IP-to-IP gateways with no transcoding and conferencing.
Using iLBC with SIP
Using iLBC with H.323
See H245, version 12 document at http://www.packetizer.com/ipmc/h245/
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:
For more configuration guidance, see the Cisco IOS Unified Communications Toll Fraud Prevention paper.
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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.
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