Voice over IP for the Cisco 2600 Series

Voice over IP enables a Cisco 2600 series router to carry voice traffic (for example, telephone calls and faxes) over an IP network. In Voice over IP, the digital signal processor segments the voice signal into frames, which are then coupled in groups of two and stored in voice packets. These voice packets are transported using IP in compliance with ITU-T specification H.323. Because it is a delay-sensitive application, you need to have a well-engineered network end-to-end to successfully use Voice over IP. Fine-tuning your network to adequately support Voice over IP involves a series of protocols and features geared toward quality of service (QoS). Traffic shaping considerations must be taken into account to ensure the reliability of the voice connection.

Voice over IP is primarily a software feature; however, to use this feature on a Cisco 2600 series router, you must install a voice network module (VNM). The VNM can hold either two or four voice interface cards (VIC), each of which is specific to a particular signaling type associated with a voice port. For more information about the physical characteristics of the VNM, as well as installing or configuring a VNM in your Cisco 2600 series router, refer to the Voice Network Module and Voice Interface Card Configuration Note that came with your VNM.

Note Voice over IP for the Cisco 2600 series uses the same VNM, configuration procedure, and command syntax as Voice over IP for the Cisco 3600 series. Refer to Cisco IOS Release 11.3 (1)T Voice over IP for the Cisco 3600 Series feature module for complete configuration and command reference information.

How Voice over IP Handles a Typical Telephone Call

Before configuring Voice over IP on your Cisco 2600 series router, it helps to understand what happens at an application level when you place a call using Voice over IP. The general flow of a two-party voice call using Voice over IP is as follows:

1. The user picks up the handset; this signals an off-hook condition to the signaling application part of Voice over IP in the Cisco 2600 series router.

2. The session application part of Voice over IP issues a dial tone and waits for the user to dial a telephone number.

3. The user dials the telephone number; those numbers are accumulated and stored by the session application.

4. After enough digits are accumulated to match a configured destination pattern, the telephone number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to either the destination telephone number or a PBX that is responsible for completing the call to the configured destination pattern.

5. The session application then runs the H.323 session protocol to establish a transmission and a reception channel for each direction over the IP network. If the call is being handled by a PBX, the PBX forwards the call to the destination telephone. If RSVP has been configured, the RSVP reservations are put into effect to achieve the desired quality of service over the IP network.

6. The CODECs are enabled for both ends of the connection and the conversation proceeds using RTP/UDP/IP as the protocol stack.

7. Any call-progress indications (or other signals that can be carried in-band) are cut through the voice path as soon as end-to-end audio channel is established. Signaling that can be detected by the voice ports (for example, in-band DTMF digits after the call setup is complete) is also trapped by the session application at either end of the connection and carried over the IP network encapsulated in RTCP using the RTCP APP extension mechanism.

8. When either end of the call hangs up, the RSVP reservations are torn down (if RSVP is used) and the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger another call setup.

Benefits

Voice over IP offers the following benefits:

Toll bypass
Remote PBX presence over WANs
Unified voice/data trunking
POTS-Internet telephony gateways

List of Terms

ACOM---Term used in G.165, "General Characteristics of International Telephone Connections and International Telephone Circuits: Echo Cancellers." ACOM is the combined loss achieved by the echo canceller, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.

Call leg---A logical connection between the router and either a telephony endpoint over a bearer channel or another endpoint using a session protocol.

Channel Associated Signaling (CAS)---A form of signaling used on a T1 line. With CAS, a signaling element is dedicated to each channel in the T1 frame. This type of signaling is sometimes called Robbed Bit Signaling (RBS) because a bit is taken out (or robbed) from the user's data stream to provide signaling information to and from the switch.

CIR---Committed Information Rate. The average rate of information transfer a subscriber (for example, the network administrator) has stipulated for a Frame Relay PVC.

CODEC---Coder-decoder compression scheme or technique. In Voice over IP, it specifies the voice coder rate of speech for a dial peer.

Dial peer---An addressable call endpoint. In Voice over IP, there are two kinds of dial peers: POTS and VoIP.

DS0---A 64K channel on an E1 or T1 WAN interface.

DTMF---Dual tone multifrequency. Use of two simultaneous voice-band tones for dial (such as touch tone).

E&M---Stands for recEive and transMit (or Ear and Mouth). E&M is a trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco's E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines).

FIFO---First-in, first-out. In data communication, FIFO refers to a buffering scheme where the first byte of data entering the buffer is the first byte retrieved by the CPU. In telephony, FIFO refers to a queuing scheme where the first calls received are the first calls processed.

FXO---Foreign Exchange Office. An FXO interface connects to the PSTN's central office and is the interface offered on a standard telephone. Cisco's FXO interface is an RJ-11 connector that allows an analog connection to be directed at the PSTN's central office. This interface is of value for off-premise extension applications.

FXS---Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.

Multilink PPP---Multilink Point-to-Point Protocol. This protocol is a method of splitting, recombining, and sequencing datagrams across multiple logical data links.

PBX---Private Branch Exchange. Privately-owned central switching office.

PLAR---Private Line Auto Ringdown. This type of service results in a call attempt to some particular remote endpoint when the local extension is taken off-key.

POTS---Plain Old Telephone Service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the public switched telephone network.

POTS dial peer---Dial peer connected via a traditional telephony network. POTS peers point to a particular voice port on a voice network device.

PSTN---Public Switched Telephone Network. PSTN refers to the local telephone company.

PVC---Permanent Virtual Circuit.

QoS---Quality of Service, which refers to the measure of service quality provided to the user.

RSVP---Resource Reservation Protocol. This protocol supports the reservation of resources across an IP network.

Trunk---Service that allows quasi-transparent connections between two PBXes, a PBX and a local extension, or some other combination of telephony interfaces to be permanently conferenced together by the session application and signaling passed transparently through the IP network.

VoIP dial peer---Dial peer connected via a packet network; in the case of Voice over IP, this is an IP network. VoIP peers point to specific VoIP devices.

Platforms

This feature is supported on these platforms:

Cisco 3600 series routers
Cisco 2600 series routers

Prerequisites

Before you can configure your Cisco 2600 series router to use Voice over IP, you must first:

Establish a working IP network. For more information about configuring IP, refer to the "IP Overview," "Configuring IP Addressing," and "Configuring IP Services" chapters in the Network Protocols Configuration Guide, Part 1.
Install the one-slot or two-slot (NM-1V/NM-2V) voice network module into the appropriate bay of your Cisco router. For more information about the physical characteristics of the voice network module, or how to install it, refer to the installation documentation, Voice Network Module and Voice Interface Card Configuration Note, that came with your voice network module.
Complete your company's dial plan.
Establish a working telephony network based on your company's dial plan.
Integrate your dial plan and telephony network into your existing IP network topology. Merging your IP and telephony networks depends on your particular IP and telephony network topology. In general, we recommend the following suggestions:
- Use canonical numbers wherever possible. It is important to avoid situations where numbering systems are significantly different on different routers or access servers in your network.
- Make routing and/or dialing transparent to the user---for example, avoid secondary dial tones from secondary switches, where possible.
- Contact your PBX vendor for instructions about how to reconfigure the appropriate PBX interfaces.

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.

Supported MIBs and RFCs

This feature supports the following MIBs:

CISCO-ANALOG-VOICE-IF-MIB
CISCO-VOICE-DIAL-CONTROL-MIB
CISCO-VOICE-IF-MIB

For descriptions of supported MIBs and how to use MIBs, see Cisco's MIB website on CCO.

This feature supports the following RFCs:

RFC 1889---RTP: A Transport Protocol for Real-Time Applications, January 1996; H. Schulzrinne, GMD Fokus; S. Casner, Precept Software, Inc; R. Frederick, Xerox Palo Alto Research Centre; V. Jacobson, Lawrence Berkeley National Laboratory
RFC 1890---RTP Profile for Audio and Video Conferences with Minimal Control, January 1996; H. Schulzrinne, GMD Fokus
RFC 2127---ISDN Management Information Base using SMIv2, March 1997; G. Roeck, Editor; Cisco Systems
RFC 2128---Dial Control Management Information Base using SMIv2, March 1997; G. Roeck, Editor; Cisco Systems

Configuration Tasks

To configure Voice over IP on the Cisco 2600 series, you need to perform the following steps:

Step 1 Configure your IP network to support real-time voice traffic. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward Quality of Service (QoS). To configure your IP network for real-time voice traffic, you need to take into consideration the entire scope of your network, then select and configure the appropriate QoS tool or tools:

RSVP
Multilink PPP with Interleaving
RTP Header Compression
Custom Queuing
Weighted Fair Queuing

Refer to "Configure IP Networks for Real-Time Voice Traffic" section in the Cisco IOS Release 11.3(3)T Voice over IP for the Cisco 3600 Series feature module for information about how to select and configure the appropriate QoS tools to optimize voice traffic on your network.

Step 2 (Optional) If you plan to run Voice over IP over Frame Relay, you need to take certain factors into consideration when configuring Voice over IP for it to run smoothly over Frame Relay. For example, a public Frame Relay cloud provides no guarantees for QoS. Refer to the "Configure Frame Relay for Voice over IP" section in the Cisco IOS Release 11.3(3)T Voice over IP for the Cisco 3600 Series feature module for information about deploying Voice over IP over Frame Relay.

Step 3 Use the num-exp command to configure number expansion if your telephone network is configured so that you can reach a destination by dialing only a portion (an extension number) of the full E.164 telephone number. Refer to the "Configure Number Expansion" section in the Cisco IOS Release 11.3(3)T Voice over IP for the Cisco 3600 Series feature module for information about number expansion.

Step 4 Use the dial-peer voice command to define dial peers and switch to the dial-peer configuration mode. Each dial peer defines the characteristics associated with a call leg. A call leg is a discrete segment of a call connection that lies between two points in the connection. An end-to-end call is comprised of four call legs, two from the perspective of the source access server, and two from the perspective of the destination access server. Dial peers are used to apply attributes to call legs and to identify call origin and destination. There are two different kinds of dial peers:

POTS---Dial peer describing the characteristics of a traditional telephony network connection. POTS peers point to a particular voice port on a voice network device. To minimally configure a POTS dial peer, you need to configure the following two characteristics: associated telephone number and logical interface. Use the destination-pattern command to associate a telephone number with a POTS peer. Use the port command to associate a specific logical interface with a POTS peers. In addition, you can specify direct inward dialing for a POTS peer by using the direct-inward-dial command.
VoIP---Dial peer describing the characteristics of a packet network connection; in the case of Voice over IP, this is an IP network. VoIP peers point to specific VoIP devices. To minimally configure a VoIP peer, you need to configure the following two characteristics: associated destination telephone number and a destination IP address. Use the destination-pattern command to define the destination telephone number associated with a VoIP peer. Use the session-target command to specify a destination IP address for a VoIP peer.

: In addition, you can use VoIP peers to define characteristics such as IP precedence, additional QoS parameters (when RSVP is configured), CODEC, and VAD. Use the ip precedence command to define IP precedence. If you have configured RSVP, use either the req-qos or acc-qos command to configure QoS parameters. Use the codec command to configure specific voice coder rates. Use the vad command to disable voice activation detection and the transmission of silence packets.

Refer to the "Configure Dial Peers" section and the "Optimize Dial Peer and Network Interface Configurations" section in the Cisco IOS Release 11.3(3)T Voice over IP for the Cisco 3600 Series feature module for additional information about configuring dial peers and dial-peer characteristics.

Step 5 You need to configure your Cisco 2600 series router to support voice ports. In general, voice-port commands define the characteristics associated with a particular voice-port signaling type. Voice ports on the Cisco 2600 series support three basic voice signaling types:

FXO---Foreign Exchange Office interface
FXS---Foreign Exchange Station interface
E&M---"RecEive and TransMit" interface or the "Ear and Mouth" interface

Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, E&M ports might need specific voice-port values configured, depending on the specifications of the devices in your telephony network. For information about configuring voice ports, refer to the "Configuring Voice Ports" section in the Cisco IOS Release 11.3(3)T Voice over IP for the Cisco 3600 Series feature module.

Configuration Examples

The actual Voice over IP configuration procedure you complete depends on the actual topology of your voice network. The following configuration examples should give you a starting point. Of course, these configuration examples would need to be customized to reflect your network topology.

Configuration procedures are supplied for the following scenarios:

FXS-to-FXS Connection Using RSVP
Linking PBX Users with E&M Trunk Lines
PSTN Gateway Access Using FXO Connection
PSTN Gateway Access Using FXO Connection (PLAR Mode)

These examples are described in the following sections.

FXS-to-FXS Connection Using RSVP

The following example shows how to configure Voice over IP for simple FXS-to-FXS connections.

In this example, a very small company, consisting of two offices, has decided to integrate Voice over IP into its existing IP network. One basic telephony device is connected to Router RLB-1; therefore Router RLB-1 has been configured for one POTS peer and one VoIP peer. Router RLB-w and Router R12-e establish the WAN connection between the two offices. Because one POTS telephony device is connected to Router RLB-2, it has also been configured for only one POTS peer and one VoIP peer.

In this example, only the calling end (Router RLB-1) is request RSVP. Figure 1 illustrates the topology of this FXS-to-FXS connection example.

Figure 1: FXS-to-FXS Connection Example

Configuration for Router RLB-1

hostname rlb-1
! Create voip dial-peer 10
dial-peer voice 10 voip
! Define its associated telephone number and IP address
  destination-pattern +4155554000
  sess-target ipv4:40.0.0.1
! Request RSVP 
  req-qos guaranteedDelay
! Create pots dial-peer 1
dial-peer voice 1 pots
! Define its associated telephone number and voice port
  destination-pattern +4085554000
  port 1/0/0
! Configure serial interface 0/0
interface Serial0/0
  ip address 10.0.0.1 255.0.0.0
  no ip mroute-cache
! Configure RTP header compression
  ip rtp header-compression
  ip rtp compression-connections 25
! Enable RSVP on this interface
  ip rsvp bandwidth 48 48
  fair-queue 64 256 36
  clockrate 64000
router igrp 888
  network 10.0.0.0
  network 20.0.0.0
  network 40.0.0.0

Configuration for Router RLB-w

hostname rlb-w
! Configure serial interface 1/0
interface Serial1/0
  ip address 10.0.0.2 255.0.0.0
! Configure RTP header compression
  ip rtp header-compression
  ip rtp compression-connections 25
! Enable RSVP on this interface
  ip rsvp bandwidth 96 96
  fair-queue 64 256 3
! Configure serial interface 1/3
interface Serial1/3
  ip address 20.0.0.1 255.0.0.0
! Configure RTP header compression
  ip rtp header-compression
  ip rtp compression-connections 25
! Enable RSVP on this interface
  ip rsvp bandwidth 96 96
  fair-queue 64 256 3
! Configure IGRP
router igrp 888
  network 10.0.0.0
  network 20.0.0.0
  network 40.0.0.0

Configuration for Router R12-e

hostname r12-e
! Configure serial interface 1/0
interface Serial1/0
  ip address 40.0.0.2 25.0.0.0
! Configure RTP header compression
  ip rtp header-compression
  ip rtp compression-connections 25
! Enable RSVP on this interface
  ip rsvp bandwidth 96 96
  fair-queue 64 256 3
! Configure serial interface 1/3
interface Serial1/3
  ip address 20.0.0.2 255.0.0.0
! Configure RTP header compression
  ip rtp header-compression
  ip rtp compression-connections 25
! Enable RSVP on this interface
  ip rsvp bandwidth 96 96
  fair-queue 64 256 3
  clockrate 128000
! Configure IGRP
router igrp 888
  network 10.0.0.0
  network 20.0.0.0
  network 40.0.0.0

Configuration for Router RLB-2

hostname r1b-2
! Create pots dial-peer 2
dial-peer voice 2 pots
! Define its associated telephone number and voice-port
  destination-pattern +4155554000
  port 1/0/0
! Create voip dial-peer 20
dial-peer voice 20 voip
!Define its associated telephone number and IP address
  destination-pattern +4085554000
  sess-target ipv4:10.0.0.1
! Configure serial interface 0/0
interface Serial0/0
  ip address 40.0.0.1 255.0.0.0
  no ip mroute-cache
! Configure RTP header compression
  ip rtp header-compression
  ip rtp compression-connections 25
! Enable RSVP on this interface
  ip rsvp bandwidth 96 96
  fair-queue 64 256 3
  clockrate 64000
! Configure IGRP
router igrp 888
  network 10.0.0.0
  network 20.0.0.0
  network 40.0.0.0

Linking PBX Users with E&M Trunk Lines

The following example shows how to configure Voice over IP to link PBX users with E&M trunk lines.

In this example, a company wants to connect two offices: one in San Jose, California and the other in Salt Lake City, Utah. Each office has an internal telephone network using PBX, connected to the voice network by an E&M interface. Both the Salt Lake City and the San Jose offices are using E&M Port Type II, with four-wire operation and ImmediateStart signaling. Each E&M interface connects to the router using two voice interface connections. Users in San Jose dial "8-111" and then the extension number to reach a destination in Salt Lake City. Users in Salt Lake City dial "4-111" and then the extension number to reach a destination in San Jose.

Figure 2 illustrates the topology of this connection example.

Figure 2: Linking PBX Users with E&M Trunk Lines Example

Note This example assumes that the company already has established a working IP connection between its two remote offices.

Router SJ Configuration

hostname sanjose
!Configure pots dial-peer 1
dial-peer voice 1 pots
  destination-pattern +111....
  port 1/0/0
!Configure pots dial-peer 2
dial-peer voice 2 pots
  destination-pattern +111....
  port 1/0/1
!Configure voip dial-peer 3
dial-peer voice 3 voip
  destination-pattern +111....
  session target ipv4:172.16.65.182
!Configure the E&M interface
voice-port 1/0/0
  signal immediate
  operation 4-wire
  type 2
voice-port 1/0/1
  signal immediate
  operation 4-wire
  type 2
!Configure the serial interface
interface serial 0/0
  description serial interface type dce (provides clock)
  clock rate 2000000
  ip address 172.16.1.123
  no shutdown

Router SLC Configuration

hostname saltlake
!Configure pots dial-peer 1
dial-peer voice 1 pots
  destination-pattern +111....
  port 1/0/0
!Configure pots dial-peer 2
dial-peer voice 2 pots
  destination-pattern +111....
  port 1/0/1
!Configure voip dial-peer 3
dial-peer voice 3 voip
  destination-pattern +111....
  session target ipv4:172.16.1.123
!Configure the E&M interface
voice-port 1/0/0
  signal immediate
  operation 4-wire
  type 2
voice-port 1/0/0
  signal immediate
  operation 4-wire
  type 2
!Configure the serial interface
interface serial 0/0
  description serial interface type dte
  ip address 172.16.65.182
  no shutdown

Note PBXes should be configured to pass all DTMF signals to the router. We recommend that you do not configure store and forward tone.

Note If you change the gain or the telephony port, make sure that the telephony port still accepts DTMF signals.

PSTN Gateway Access Using FXO Connection

The following example shows how to configure Voice over IP to link users with the PSTN gateway using an FXO connection.

In this example, users connected to Router SJ in San Jose, California can reach PSTN users in Salt Lake City, Utah via Router SLC. Router SLC in Salt Lake City is connected directly to the PSTN through an FXO interface.

Figure 3 illustrates the topology of this connection example.

Figure 3: PSTN Gateway Access Using FXO Connection Example

Note This example assumes that the company already has established a working IP connection between its two remote offices.

Router SJ Configuration

! Configure pots dial-peer 1
dial-peer voice 1 pots
  destination-pattern +14085554000
  port 1/0/0
! Configure voip dial-peer 2
dial-peer voice 2 voip
  destination-pattern +9...........
  session target ipv4:172.16.65.182
! Configure the serial interface
interface serial 0/0
  clock rate 2000000
  ip address 172.16.1.123
  no shutdown

Router SLC Configuration

! Configure pots dial-peer 1
dial-peer voice 1 pots
  destination-pattern +9...........
  port 1/0/0
! Configure voip dial-peer 2
dial-peer voice 2 voip
  destination-pattern +14085554000
  session target ipv4:172.16.1.123
! Configure serial interface
interface serial 0/0
  ip address 172.16.65.182
  no shutdown

PSTN Gateway Access Using FXO Connection (PLAR Mode)

The following example shows how to configure Voice over IP to link users with the PSTN Gateway using an FXO connection (PLAR mode).

In this example, PSTN users in Salt Lake City, Utah, can dial a local number and establish a private line connection in a remote location. As in the previous example, Router SLC in Salt Lake City is connected directly to the PSTN through an FXO interface.

Figure 4 illustrates the topology of this connection example.

Figure 4: PSTN Gateway Access Using FXO Connection (PLAR Mode)

Note This example assumes that the company already has established a working IP connection between its two remote offices.

Router SJ Configuration

! Configure pots dial-peer 1
dial-peer voice 1 pots
  destination-pattern +14085554000
  port 1/0/0
! Configure voip dial-peer 2
dial-peer voice 2 voip
  destination-pattern +9...........
  session target ipv4:172.16.65.182
! Configure the serial interface
interface serial 0/0
  clock rate 2000000
  ip address 172.16.1.123
  no shutdown

Router SLC Configuration

! Configure pots dial-peer 1
dial-peer voice 1 pots
  destination-pattern +9...........
  port 1/0/0
! Configure voip dial-peer 2
dial-peer voice 2 voip
  destination-pattern +14085554000
  session target ipv4:172.16.1.123
! Configure the voice port
voice port 1/0/0
connection plar 14085554000
! Configure the serial interface
interface serial 0/0
  ip address 172.16.65.182
  no shutdown

Command Reference

All commands used with this feature are documented in the "Voice over IP for the Cisco 3600 Series Commands" section of the Cisco IOS Release 11.3(1)T Voice over IP for the Cisco 3600 Series feature module. Command implementation is identical to that used for the Cisco 3600 series.

Debug Commands

All debug commands used with this feature are documented in the "Voice over IP for the Cisco 3600 Series Debug Commands" section of the Cisco IOS Release 11.3(1)T feature Voice over IP for the Cisco 3600 Series feature module. Debug command implementation is identical to that used for the Cisco 3600 series.

Table of Contents

Voice over IP for the Cisco 2600 Series

Feature Summary

Platforms

Prerequisites

Supported MIBs and RFCs

Configuration Tasks

Configuration Examples

Command Reference

Debug Commands