Table of Contents

Configuring Voice over IP for the Cisco 3600 Series

Configuring Voice over IP for the Cisco 3600 Series

This chapter explains how to configure Voice over IP (VoIP) on Cisco 3600 series routers and contains the following sections:

• Introduction
  
  
• List of Terms
  
  
• Prerequisites Tasks
  
  
• How Voice over IP Handles a Typical Telephone Call
  
  
• Configuration Tasks
  
  
• Configure IP Networks for Real-Time Voice Traffic
  
  
  • Configure RSVP for Voice
    
    
  • Configure Multilink PPP with Interleaving
    
    
  • Configure RTP Header Compression
    
    
  • Configure Custom Queuing
    
    
  • Configure Weighted Fair Queuing
    
    
• Configure Frame Relay for Voice over IP
  
  
• Configure Number Expansion
  
  
  • Create a Number Expansion Table
    
    
  • Configure Number Expansion
    
    
• Configure Dial Peers
  
  
  • Create a Peer Configuration Table
    
    
  • Configure POTS Peers
    
    
  • Configure VoIP Peers
    
    
• Configure Voice Ports
  
  
  • Configuring FXO or FXS Voice Ports
    
    
  • Fine-Tune FXO and FXS Voice Ports
    
    
  • Configure E&M Voice Ports
    
    
  • Fine-Tune E&M Voice Ports
    
    
• Optimize Dial Peer and Network Interface Configurations
  
  
  • Configure IP Precedence for Dial Peers
    
    
  • Configure RSVP for Dial Peers
    
    
  • Configure CODEC and VAD for Dial Peers
    
    
• Configure Voice over IP for Microsoft NetMeeting
  

All of these tasks are described in the following sections.

Introduction

Voice over IP (VoIP) enables a Cisco 3600 series router to carry voice traffic (for example, telephone calls and faxes) over an IP network. Voice over IP is primarily a software feature; however, to use this feature on a Cisco 3600 series router, you must install a Voice Network Module (VNM). The VNM can hold either 2 or 4 Voice Interface Cards (VICs). Each VIC is specific to a particular signaling type associated with a voice port; therefore, VICs determine the type of signaling for the voice ports on that particular VNM. For more information about the physical characteristics of the VNM, as well as installing or configuring a VNM in your Cisco 3600 series router, refer to the Voice Network Module and Voice Interface Card Configuration Note that came with your VNM.

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.

Prerequisites Tasks

Before you can configure your Cisco 3600 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.

How Voice over IP Handles a Typical Telephone Call

Before configuring Voice over IP on your Cisco 3600 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:

Configuration Tasks

To configure Voice over IP on the Cisco 3600 series, you need to perform the following steps:

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

Refer to "Configure IP Networks for Real-Time Voice Traffic" section 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 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 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 for additional information about configuring dial peers and dial-peer characteristics.

Step 5   You need to configure your Cisco 3600 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 3600 series support three basic voice signaling types:

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.

Configure 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.

The important thing to remember is that QoS must be configured throughout your network—not just on the Cisco 3600 series 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 differ as well. To configure your IP network for real-time voice traffic, you need to take into consideration 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:

• Packet classification
  
  
• Admission control
  
  
• Bandwidth management
  
  
• Queuing
  
  

In general, backbone routers perform the following QoS functions:

• High-speed switching and transport
  
  
• Congestion management
  
  
• Queue management
  
  

Scalable QoS solutions require cooperative edge and backbone functions.

Although not mandatory, some QoS tools have been identified as being valuable in fine-tuning your network to support real-time voice traffic. To configure your IP network for QoS using these tools, perform one or more of the following tasks:

• Configure RSVP for Voice
  
  
• Configure Multilink PPP with Interleaving
  
  
• Configure RTP Header Compression
  
  
• Configure Custom Queuing
  
  
• Configure Weighted Fair Queuing
  
  

Each of these tasks is discussed in the following sections.

Configure RSVP for Voice

RSVP allows end systems to request a particular Quality of Service (QoS) from the network. Real-time voice traffic requires network consistency. Without consistent QoS, real-time traffic can experience jitter, insufficient bandwidth, delay variations, or information loss. RSVP works in conjunction with current queuing mechanisms. It is up to the interface queuing mechanism (such as weighted fair queuing or weighted random early detection) to implement the reservation.

RSVP can be equated to a dynamic access list for packet flows.

You should configure RSVP to ensure QoS if the following conditions exist in your network:

• Small scale voice network implementation
  
  
• Slow links
  
  
• Links with high utilization
  
  
• Links less than 2 Mbps
  
  
• Need for the best possible voice quality
  
  

Enable RSVP

To minimally configure RSVP for voice traffic, you must enable RSVP on each interface where priority needs to be set.

By default, RSVP is disabled so that it is backward compatible with systems that do not implement RSVP. To enable RSVP on an interface, use the following command in interface configuration mode:

This command starts RSVP and sets the bandwidth and single-flow limits. The default maximum bandwidth is up to 75 percent of the bandwidth available on the interface. By default, the amount reservable by a flow can be up to the entire reservable bandwidth.

On subinterfaces, this applies the more restrictive of the available bandwidths of the physical interface and the subinterface.

Reservations on individual circuits that do not exceed the single flow limit normally succeed. If, however, reservations have been made on other circuits adding up to the line speed, and a reservation is made on a subinterface which itself has enough remaining bandwidth, it will still be refused because the physical interface lacks supporting bandwidth.

Cisco AS5300s running VoIP and configured for RSVP request allocations per the following formula:

bps=packet_size+ip/udp/rtp header size * 50 per second
 

For G.729, the allocation works out to be 24,000 bps. For G.711, the allocation is 80,000 bps.

For more information about configuring RSVP, refer to the "Configuring RSVP" chapter of the Cisco IOS Release 11.3 Network Protocols Configuration Guide, Part 1.

RSVP Configuration Example

The following example enables RSVP and sets the maximum bandwidth to 100 kbps and the maximum bandwidth per single request to 32 kbps (the example presumes that both VoIP dial peers have been configured):

interface serial 1/0/0
 ip rsvp bandwidth 100 32
 fair-queue
 end
!
dial-peer voice 1211 voip
 req-qos controlled-load
!
dial-peer voice 1212 voip
 req-qos controlled-load

Configure Multilink PPP with Interleaving

Multi-class Multilink PPP Interleaving allows large packets to be multilink-encapsulated and fragmented into smaller packets to satisfy the delay requirements of real-time voice traffic; small real-time packets, which are not multilink-encapsulated, are transmitted between fragments of the large packets. The interleaving feature also provides a special transmit queue for the smaller, delay-sensitive packets, enabling them to be transmitted earlier than other flows. Interleaving provides the delay bounds for delay-sensitive voice packets on a slow link that is used for other best-effort traffic.

In general, Multilink PPP with interleaving is used in conjunction with weighted fair queuing and RSVP or IP Precedence to ensure voice packet delivery. Use Multilink PPP with interleaving and weighted fair queuing to define how data will be managed; use RSVP or IP Precedence to give priority to voice packets.

You should configure Multilink PPP if the following conditions exist in your network:

• Point-to-point connection using PPP Encapsulation
  
  
• Slow links
  
  

Multilink PPP support for interleaving can be configured on virtual templates, dialer interfaces, and ISDN BRI or PRI interfaces. To configure interleaving, you need to complete the following tasks:

• Configure the dialer interface or virtual template, as defined in the relevant chapters of the Cisco IOS Release 11.3 Dial Solutions Configuration Guide.
  
  
• Configure Multilink PPP and interleaving on the interface or template.
  
  

To configure Multilink PPP and interleaving on a configured and operational interface or virtual interface template, use the following commands in interface mode:

For more information about Multilink PPP, refer to the "Configuring Media-Independent PPP and Multilink PPP" chapter in the Cisco IOS Release 11.3 Dial Solutions Configuration Guide.

Multilink PPP Configuration Example

The following example defines a virtual interface template that enables Multilink PPP with interleaving and a maximum real-time traffic delay of 20 milliseconds, and then applies that virtual template to the Multilink PPP bundle:

interface virtual-template 1
 ppp multilink
 encapsulated ppp
 ppp multilink interleave
 ppp multilink fragment-delay 20
 ip rtp reserve 16384 100 64
 
multilink virtual-template 1

Configure RTP Header Compression

Real-Time Transport Protocol (RTP) is used for carrying packetized audio traffic over an IP network. RTP header compression compresses the IP/UDP/RTP header in an RTP data packet from 40 bytes to approximately 2 to 4 bytes (most of the time), as shown in Figure 2-1.

This compression feature is beneficial if you are running Voice over IP over slow links. Enabling compression on both ends of a low-bandwidth serial link can greatly reduce the network overhead if there is a lot of RTP traffic on that slow link.

Typically, an RTP packet has a payload of approximately 20 to 160 bytes for audio applications that use compressed payloads. RTP header compression is especially beneficial when the RTP payload size is small (for example, compressed audio payloads between 20 and 50 bytes).

You should configure RTP header compression if the following conditions exist in your network:

• Slow links
  
  
• Need to save bandwidth
  
  

Perform the following tasks to configure RTP header compression for Voice over IP. The first task is required; the second task is optional.

• Enable RTP Header Compression on a Serial Interface
  
  
• Change the Number of Header Compression Connections
  
  

Enable RTP Header Compression on a Serial Interface

To use RTP header compression, you need to enable compression on both ends of a serial connection. To enable RTP header compression, use the following command in interface configuration mode:

If you include the passive keyword, the software compresses outgoing RTP packets only if incoming RTP packets on the same interface are compressed. If you use the command without the passive keyword, the software compresses all RTP traffic.

Change the Number of Header Compression Connections

By default, the software supports a total of 16 RTP header compression connections on an interface. To specify a different number of RTP header compression connections, use the following command in interface configuration mode:

RTP Header Compression Configuration Example

The following example enables RTP header compression for a serial interface:

interface 0
 ip rtp header-compression
 encapsulation ppp
 ip rtp compression-connections 25
 

For more information about RTP header compression, see the "Configuring IP Multicast Routing" chapter of the Cisco IOS Release 11.3 Network Protocols Configuration Guide, Part 1.

Configure Custom Queuing

Some QoS features, such as IP RTP reserve and custom queuing, are based on the transport protocol and the associated port number. Real-time voice traffic is carried on UDP ports ranging from 16384 to 16624. This number is derived from the following formula:

16384 = 4(number of voice ports in the Cisco 3600 series router)
 

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 "Managing System Performance" chapter in the Cisco IOS Release 11.3 Configuration Fundamentals Configuration Guide.

Configure Weighted Fair Queuing

Weighted fair queuing ensures that queues do not starve for bandwidth and that traffic gets predictable service. Low-volume traffic streams receive preferential service; high-volume traffic streams share the remaining capacity, obtaining equal or proportional bandwidth.

In general, weighted fair queuing is used in conjunction with Multilink PPP with interleaving and RSVP or IP Precedence to ensure that voice packet delivery. Use weighted fair queuing with Multilink PPP to define how data will be managed; use RSVP or IP Precedence to give priority to voice packets. For more information about weighted fair queuing, refer to the "Managing System Performance" chapter in the Cisco IOS Release 11.3 Configuration Fundamentals Configuration Guide.

Configure Frame Relay for Voice over IP

You need to take certain factors into consideration when configuring Voice over IP for it to run smoothly over Frame Relay. A public Frame Relay cloud provides no guarantees for QoS. For real-time traffic to be transmitted in a timely manner, the data rate must not exceed the committed information rate (CIR) or there is the possibility that packets will be dropped. In addition, Frame Relay traffic shaping and RSVP are mutually exclusive. This is particularly important to remember if multiple DLCIs are carried on a single interface.

For Frame Relay links with slow output rates (less than or equal to 64 kbps), where data and voice are being transmitted over the same PVC, we recommend the following solutions:

• Separate DLCIs for voice and data—By providing a separate subinterface for voice and data, you can use the appropriate QoS tool per line. For example, each DLCI would use 32 kbps of a 64 kbps line.
  
  
  • Apply adaptive traffic shaping to both DLCIs.
    
    
  • Use RSVP or IP Precedence to prioritize voice traffic.
    
    
  • Use compressed RTP to minimize voice packet size.
    
    
  • Use weighted fair queuing to manage voice traffic.
    
    
• Lower MTU size—Voice packets are generally small. By lowering the MTU size (for example, to 300 bytes), large data packets can be broken up into smaller data packets that can more easily be interwoven with voice packets.
  
  

• CIR equal to line rate—Make sure that the data rate does not exceed the CIR. This is accomplished through generic traffic shaping.
  
  
  • Use RSVP or IP Precedence to prioritize voice traffic.
    
    
  • Use compressed RTP to minimize voice packet header size.
    
    
• Traffic shaping—Use adaptive traffic shaping to throttle back the output rate based on the BECN. If the feedback from the switch is ignored, packets (both data and voice) might be discarded. Because the Frame Relay switch does not distinguish between voice and data packets, voice packets could be discarded, which would result in a deterioration of voice quality.
  
  
  • Use RSVP, compressed RTP, reduced MTU size, and adaptive traffic shaping based on BECN to hold data rate to CIR.
    
    
  • Use generic traffic shaping to obtain a low interpacket wait time. For example, set Bc to 4000 to obtain an interpacket wait of 125 ms.
    
    

In Cisco IOS Release 11.3, Frame Relay Traffic Shaping is not compatible with RSVP. We suggest one of the following workarounds:

• Provision the Frame Relay PVC to have the CIR equal to the port speed.
  
  
• Use Generic Traffic Shaping with RSVP.
  
  

Frame Relay for Voice over IP Configuration Example

For Frame Relay, it is customary to configure a main interface and several subinterfaces, one subinterface per PVC. The following example configures a Frame Relay main interface and a subinterface so that voice and data traffic can be successfully transported:

interface Serial0/0
 mtu 300
 no ip address
 encapsulation frame-relay
 no ip route-cache
 no ip mroute-cache
 fair-queue 64 256 1000
 frame-relay ip rtp header-compression
 
interface Serial0/0.1 point-to-point
 mtu 300
 ip address 40.0.0.7 255.0.0.0
 ip rsvp bandwidth 48 48
 no ip route-cache
 no ip mroute-cache
 bandwidth 64
 traffic-shape rate 32000 4000 4000
 frame-relay interface-dlci 16
 frame-relay ip rtp header-compression
 

In this configuration example, the main interface has been configured as follows:

• MTU size is 300 bytes.
  
  
• No IP address is associated with this serial interface. The IP address must be assigned for the subinterface.
  
  
• Encapsulation method is Frame Relay.
  
  
• Fair-queuing is enabled.
  
  
• IP RTP header compression is enabled.
  
  

The subinterface has been configured as follows:

• MTU size is inherited from the main interface.
  
  
• IP address for the subinterface is specified.
  
  
• Bandwidth is set to 64 kbps.
  
  
• RSVP is enabled to use the default value, which is 75 percent of the configured bandwidth.
  
  
• Generic traffic shaping is enabled with 32 kbps CIR where Bc=4000 bits and Be=4000 bits.
  
  
• Frame Relay DLCI number is specified.
  
  
• IP RTP header compression is enabled.
  
  

For more information about Frame Relay, refer to the "Configuring Frame Relay" chapter in the Cisco IOS Release 11.3 Wide-Area Networking Configuration Guide.

Configure Number Expansion

In most corporate environments, the 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. Voice over IP can be configured to recognize extension numbers and expand them into their full E.164 dialed number by using two commands in tandem: destination-pattern and num-exp. Before you configure these two commands, it is helpful to map individual telephone extensions with their full E.164 dialed numbers. This can be done easily by creating a number expansion table.

Create a Number Expansion Table

In Figure 2-2, a small company wants to use Voice over IP to integrate its telephony network with its existing IP network. The destination pattern (or expanded telephone number) associated with Access Server 1 (located to the left of the IP cloud) is (408) 526-xxxx, where xxxx identifies the individual dial peers by extension. The destination pattern (or expanded telephone number) associated with Access Server 2 (located to the right of the IP cloud) is (729) 422-xxxx.

Table 2-1 shows the number expansion table for this scenario.

The information included in this example needs to be configured on both Access Server 1 and Access Server 2.

Configure Number Expansion

To define how to expand an extension number into a particular destination pattern, use the following command in global configuration mode:

You can verify the number expansion information by using the show num-exp command to verify that you have mapped the telephone numbers correctly.

After you have configured dial peers and assigned destination patterns to them, you can verify number expansion information by using the show dialplan number command to see how a telephone number maps to a dial peer.

Configure Dial Peers

The key point to understanding how Voice over IP functions is to understand dial peers. Each dial peer defines the characteristics associated with a call leg, as shown in Figure 2-3 and Figure 2-4. 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:

• 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.
  
• 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.
  

An end-to-end call is comprised of four call legs, two from the perspective of the source access server as shown in Figure 2-3, and two from the perspective of the destination access server as shown in Figure 2-4. A dial peer is associated with each one of these call legs. 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.

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 router's perspective. An inbound call leg originates outside the router. An outbound call leg originates from the router.

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 Voice over IP connections.

Establishing communication using Voice over IP is similar to configuring an IP static route: you are establishing a specific voice connection between two defined endpoints. As shown in Figure 2-5, for outgoing calls (from the perspective of the POTS dial peer 1), the POTS dial peer establishes the source (via the originating telephone number or voice port) of the call. The VoIP dial peer establishes the destination by associating the destination telephone number with a specific IP address.

To configure call connectivity between the source and destination as illustrated in Figure 2-5, enter the following commands on router 10.1.2.2:

dial-peer voice 1 pots
 destination-pattern 1408526....
 port 1/0/0
 
dial-peer voice 2 voip
 destination-pattern 1310520....
 session target ipv4:10.1.1.2
 

In the previous configuration example, the last four digits in the VoIP dial peer's destination pattern were replaced with wildcards. This means that from access server 10.1.2.2, calling any number string that begins with the digits "1310520" will result in a connection to access server 10.1.1.2. This implies that access server 10.1.1.2 services all numbers beginning with those digits. From access server 10.1.1.2, calling any number string that begins with the digits "1408526" will result in a connection to access server 10.1.2.2. This implies that access server 10.1.2.2 services all numbers beginning with those digits. For more information about stripping and adding digits, see the "Outbound Dialing on POTS Peers" section.

Figure 2-6 shows how to complete the end-to-end call between dial peer 1 and dial peer 4.

To complete the end-to-end call between dial peer 1 and dial peer 4 as illustrated in Figure 2-6, enter the following commands on router 10.1.1.2:

dial-peer voice 4 pots
 destination-pattern 1310520....
 port 1/0/0
 
dial-peer voice 3 voip
 destination-pattern 1408526....
 session target ipv4:10.1.2.2
 

Create a Peer Configuration Table

There is specific data relative to each dial peer that needs to be identified before you can configure dial peers in Voice over IP. One way to do this is to create a peer configuration table.

Using the example in Figure 2-2, Router 1, with an IP address of 10.1.1.1, connects a small sales branch office to the main office through Router 2. There are three telephones in the sales branch office that need to be established as dial peers. Access Server 2, with an IP address of 10.1.1.2, is the primary gateway to the main office; as such, it needs to be connected to the company's PBX. There are four devices that need to be established as dial peers in the main office, all of which are basic telephones connected to the PBX. Figure 2-2 shows a diagram of this small voice network.

Table 2-2 shows the peer configuration table for the example illustrated in Figure 2-2.

Configure POTS Peers

Once again, 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 number(s), 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 enter the dial-peer configuration mode (and select POTS as the method of voice-related encapsulation), use the following command in global configuration mode:

The number value of the dial-peer voice pots command is a tag that uniquely identifies the dial peer. (This number has local significance only.)

To configure the identified POTS peer, use the following commands in dial-peer configuration mode:

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.

For additional POTS dial-peer configuration options, refer to the "Voice over IP Commands for the Cisco 3600 Series" section.

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 Figure 2-7, 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.

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 has been 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:

• Called-number (DNIS)—Set of numbers representing the destination, which is derived from the ISDN setup message or CAS DNIS.
  
  
• Calling-number (ANI)—Set of numbers representing the origin, which is derived from the ISDN setup message or CAS DNIS.
  
  
• Voice port—The voice port carrying the call.
  
  

The four defined dial peer elements are:

• Destination pattern—A pattern representing the phone numbers to which the peer can connect.
  
  
• Answer address—A pattern representing the phone numbers from which the peer can connect.
  
  
• Incoming called-number—A pattern representing the phone numbers that associate an incoming call leg to a peer based on the called-number or DNIS.
  
  
• Port—The port through which calls to this peer are placed.
  
  

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-number
 else if the type is matched, associate calling-number with answer-address
 else if the type is matched, associate calling-number with destination-pattern
 else if the type is matched, associate voice port to port
 

This 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 DID for a particular POTS dial peer, use the following commands, initially in global configuration mode:

For additional POTS dial-peer configuration options, refer to the "Voice over IP Commands for the Cisco 3600 Series" chapter.

Configure VoIP Peers

Once again, 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 enter the dial-peer configuration mode (and select VoIP as the method of voice-related encapsulation), use the following command in global configuration mode:

The number value of the dial-peer voice voip command is a tag that uniquely identifies the dial peer.

To configure the identified VoIP peer, use the following commands in dial-peer configuration mode:

For additional VoIP dial-peer configuration options, refer to the "Voice over IP Commands for the Cisco 3600 Series" chapter. For examples of how to configure dial peers, refer to the chapter, "Voice over IP Configuration Examples for the Cisco 3600 Series."

Verify

You can check the validity of your dial-peer configuration by performing the following tasks:

• If you have relatively few dial peers configured, you can use the show dial-peer voice command to verify that the data configured is correct. Use this command to display a specific dial peer or to display all configured dial peers.
  
  
• Use the show dialplan number command to show the dial peer to which a particular number (destination pattern) resolves.
  
  

Tips

If you are having trouble connecting a call and you suspect the problem is associated with dial-peer configuration, you can try to resolve the problem by performing the following tasks:

• Ping the associated IP address to confirm connectivity. If you cannot successfully ping your destination, refer to the chapter "Configuring IP" in the Cisco IOS 11.3 Network Protocols Configuration Guide, Part 1.
  
  
• Use the show dial-peer voice command to verify that the operational status of the dial peer is up.
  
  
• Use the show dialplan number command on the local and remote routers to verify that the data is configured correctly on both.
  
  
• If you have configured number expansion, use the show num-exp command to check that the partial number on the local router maps to the correct full E.164 telephone number on the remote router.
  
  
• If you have configured a CODEC value, there can be a problem if both VoIP dial peers on either side of the connection have incompatible CODEC values. Make sure that both VoIP peers have been configured with the same CODEC value.
  
  
• Use the debug vpm spi command to verify the output string the router dials is correct.
  
  
• Use the debug cch323 rtp command to check RTP packet transport.
  
  
• Use the debug cch323 h225 command to check the call setup.
  
  

Configure Voice Ports

The Cisco 3600 currently provides only analog voice ports for its implementation of Voice over IP. The type of signaling associated with these analog voice ports depends on the interface module installed into the device. The Cisco 3600 series router supports either a two-port or four-port voice network module (VNM); VNMs can hold either 2 or 4 voice interface cards (VICs).

Each VIC is specific to a particular signaling type; therefore, VICs determine the type of signaling for the voice ports on that particular VNM. This means that even though VNMs can hold multiple VICs, each VIC on a VNM must conform to the same signaling type. For more information about the physical characteristics of VNMs and VICs or how to install them, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your VNM.

Voice ports on the Cisco 3600 series support three basic voice signaling types:

• FXO—Foreign Exchange Office interface. The FXO interface is an RJ-11 connector that allows a connection to be directed at the PSTN's central office (or to a standard PBX interface, if the local telecommunications authority permits). This interface is of value for off-premise extension applications.
  
  
• FXS—The Foreign Exchange Station interface is an RJ-11 connector that allows connection for basic telephone equipment, keysets, PBXs, and supplies ring, voltage, and dial tone.
  
  
• E&M—The "Ear and Mouth" interface (or "RecEive and TransMit" interface) is an RJ-48 connector that allows connection for PBX trunk lines (tie lines). It is a signaling technique for 2-wire and 4-wire telephone and trunk interfaces.
  
  

In general, voice port commands define the characteristics associated with a particular voice port signaling type. 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.

Configuring FXO or FXS Voice Ports

Under most circumstances the default voice port values are adequate for both FXO and FXS voice ports. If you need to change the default configuration for these voice ports, perform the following tasks. Items included in Step 1 and Step 2 are required; items included in Step 3 are optional.

Step 2   Configure the following mandatory voice-port parameters by using the indicated commands:

• Dial type (FXO only) using the dial-type command
  
  
• Signal type using the signal command
  
  
• Call progress tone using the cptone command
  
  
• Ring frequency (FXS only) using the ring frequency command
  
  
• Ring number (FXO only) using the ring number command
  
  

Step 3   Configure one or more of the following optional voice-port parameters by using the indicated commands:

• PLAR connection mode using the connection plar command
  
  
• Music-threshold using the music-threshold command
  
  
• Description using the description command
  
  
• Comfort noise (if VAD is activated—VAD is a dial-peer command) using the comfort-noise command
  
  

To configure FXO and FXS voice ports, use the following commands beginning in privileged EXEC mode:

Validation Tips

You can check the validity of your voice-port configuration by performing the following tasks:

• Pick up the handset of an attached telephony device and check for a dial tone.
  
  
• If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, then the voice port is most likely configured properly.
  
  
• Use the show voice-port command to verify that the data configured is correct.
  
  

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:

• Ping the associated IP address to confirm connectivity. If you cannot successfully ping your destination, refer to the Network Protocols Configuration Guide, Part 1.
  
  
• Use the show voice port command to make sure that the port is enabled. If the port is offline, use the no shutdown command.
  
  
• If you have configured E&M interfaces, make sure that the values pertaining to your specific PBX setup, such as timing and/or type, are correct.
  
  
• Check to see if the voice network module has been correctly installed. For more information, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your voice network module.
  
  

Fine-Tune FXO and FXS Voice Ports

Depending on the specifics of your particular network, you might need to adjust voice parameters involving timing, input gain, and output attenuation for FXO or FXS voice ports. Collectively, these commands are referred to as voice-port tuning commands.

To configure voice-port tuning for FXO and FXS voice ports, perform the following steps:

Step 2   For each of the following parameters, select the appropriate value using the commands indicated:

• Input gain using the input gain command
  
  
• Output attenuation using the output attenuation command
  
  
• Echo cancel coverage using the echo-cancel enable and echo-cancel coverage commands
  
  
• Non-linear processing using the non-linear command
  
  
• Initial digit timeouts using the timeouts initial command
  
  
• Interdigit timeouts using the timeouts interdigits command
  
  
• Timing other than timeouts, using the timing digit, timing inter-digit, timing pulse-digit and timing pulse-inter-digit commands.
  
  

To fine-tune FXO or FXS voice ports, use the following commands beginning in privileged EXEC mode:

Configure E&M Voice Ports

Unlike FXO and FXS voice ports, the default E&M voice-port parameters most likely will not be sufficient to enable voice data transmission over your IP network. E&M voice-port values must match those specified by the particular PBX device to which it is connected.

To configure an E&M voice port, perform the following steps. Items included in Step 1 and Step 2 are required; items included in Step 3 are optional.

Step 2   For each of the following required parameters, select the appropriate parameter value using the commands indicated:

• Dial type using the dial-type command
  
  
• Signal type using the signal command
  
  
• Call progress tone using the cptone command
  
  
• Operation using the operation command
  
  
• Type using the type command
  
  
• Impedance using the impedance command
  
  

Step 3   Select one or more of the following optional parameters, using the indicated commands:

• Connection mode using the connection plar command
  
  
• Music-threshold using the music-threshold command
  
  
• Description using the description command
  
  
• Comfort tone (if VAD is activated) using the comfort-noise command
  
  

To configure E&M voice ports, use the following commands beginning in privileged EXEC mode:

Validation Tips

You can check the validity of your voice-port configuration by performing the following tasks:

• Pick up the handset of an attached telephony device and check for a dial tone.
  
  
• If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, then the voice port is most likely configured properly.
  
  
• Use the show voice port command to verify that the data configured is correct.
  
  

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:

• Ping the associated IP address to confirm connectivity. If you cannot successfully ping your destination, refer to the Network Protocols Configuration Guide, Part 1.
  
  
• Use the show voice-port command to make sure that the port is enabled. If the port is offline, use the no shutdown command.
  
  
• If you have configured E&M interfaces, make sure that the values pertaining to your specific PBX setup, such as timing and/or type, are correct.
  
  
• Check to see if the voice network module has been correctly installed. For more information, refer to the installation document that came with your voice network module.
  
  

Fine-Tune E&M 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 E&M voice ports. Collectively, these commands are referred to as voice-port tuning commands.

To configure voice-port tuning for E&M voice ports, perform the following steps:

Step 2   For each of the following parameters, select the appropriate value, using the commands indicated:

• Input gain using the input gain command
  
  
• Output attenuation using the output attenuation command
  
  
• Echo cancel coverage using echo-cancel enable and echo-cancel coverage commands
  
  
• Non-linear processing using the non-linear command
  
  
• Initial digit timeouts using the timeouts initial command
  
  
• Interdigit timeouts using the timeouts interdigit command
  
  
• Timing other than timeouts using the timing clear-wait, timing delay-duration, timing delay-start, timing dial-pulse min-delay, timing digit, timing pulse, timing pulse-inter-digit, timing wink-duration, and timing wink-wait commands.
  
  

To fine-tune E&M voice ports, use the following commands beginning in privileged EXEC mode:

Optimize Dial Peer and Network Interface Configurations

Depending on how you have configured your network interfaces, you might need to configure additional VoIP dial-peer parameters. This section describes the following topics:

• Configure IP Precedence for Dial Peers
  
  
• Configure RSVP for Dial Peers
  
  
• Configure CODEC and VAD for Dial Peers
  
  

Configure IP Precedence for Dial Peers

If you want to give real-time voice traffic a higher priority than other network traffic, you can weight the voice data traffic associated with a particular VoIP dial peer by using IP Precedence. IP Precedence scales better than RSVP but provides no admission control.

To give real-time voice traffic precedence over other IP network traffic, use the following commands, beginning in global configuration mode:

In IP Precedence, the numbers 1 through 5 identify classes for IP flows; the numbers 6 through 7 are used for network and backbone routing and updates.

For example, to ensure that voice traffic associated with VoIP dial peer 103 is given a higher priority than other IP network traffic, enter the following:

dial-peer voice 103 voip
 ip precedence 5
 

In this example, when an IP call leg is associated with VoIP dial peer 103, all packets transmitted to the IP network via this dial peer will have their precedence bits set to 5. If the networks receiving these packets have been configured to recognize precedence bits, the packets will be given priority over packets with a lower configured precedence value.

Configure RSVP for Dial Peers

If you have configured your WAN or LAN interfaces for RSVP, you must configure the QoS for any associated VoIP peers. To configure quality of service for a selected VoIP peer, use the following commands, beginning in global configuration mode:

For example, to specify guaranteed delay QoS for VoIP dial peer 108, enter the following:

Dial-peer voice 108 voip
 destination-pattern +1408528
 req-qos controlled-load
 session target ipv4:10.0.0.8
 

In this example, every time a connection is made through VoIP dial peer 108, an RSVP reservation request is made between the local router, all intermediate routers in the path, and the final destination router.

To generate an SNMP trap message if the reserved QoS is less than the configured value for a selected VoIP peer, use the following commands, beginning from the global configuration mode:

Configure CODEC and VAD for Dial Peers

Coder-decoder (CODEC) and voice activity detection (VAD) for a dial peer determine how much bandwidth the voice session uses. CODEC typically is used to transform analog signals into a digital bit stream and digital signals back into analog signals—in this case, it specifies the voice coder rate of speech for a dial peer. VAD is used to disable the transmission of silence packets.

Configure CODEC for a VoIP Dial Peer

To specify a voice coder rate for a selected VoIP peer, use the following commands, initially beginning in global configuration mode:

The default for the codec command is g729r8; normally the default configuration for this command is the most desirable. If, however, you are operating on a high bandwidth network and voice quality is of the highest importance, you should configure the codec command for g711alaw or ulaw. Using this value will result in better voice quality, but it will also require higher bandwidth requirements for voice.

For example, to specify a CODEC rate of G.711a-law for VoIP dial peer 108, enter the following:

Dial-peer voice 108 voip
 destination-pattern +1408528
 codec g711alaw
 session target ipv4:10.0.0.8

Configure VAD for a VoIP Dial Peer

To disable the transmission of silence packets for a selected VoIP peer, use the following commands, beginning in global configuration mode:

The default for the vad command is enabled; normally the default configuration for this command is the most desirable. If you are operating on a high bandwidth network and voice quality is of the highest importance, you should disable vad. Using this value will result in better voice quality, but it will also require higher bandwidth requirements for voice.

For example, to enable VAD for VoIP dial peer 108, enter the following:

Dial-peer voice 108 voip
 destination-pattern +1408528
 vad
 session target ipv4:10.0.0.8

Configure Voice over IP for Microsoft NetMeeting

Voice over IP can be used with Microsoft NetMeeting (Version 2.x) when the Cisco 3600 series router is used as the voice gateway. Use the latest version of DirectX drivers from Microsoft on your PC to improve the voice quality of NetMeeting.

Configure Voice over IP to Support Microsoft NetMeeting

To configure Voice over IP to support NetMeeting, create a VoIP peer that contains the following information:

• Session Target—IP address or DNS name of the PC running NetMeeting
  
  
• CODEC—g711ulaw or g711alaw
  
  

Configure Microsoft NetMeeting for Voice over IP

To configure NetMeeting to work with Voice over IP, complete the following steps:

Step 2   Click the Audio tab.

Step 3   Click the "Calling a telephone using NetMeeting" check box.

Step 4   Enter the IP address of the Cisco 3600 series router in the IP address field.

Step 5   Under General, click Advanced.

Step 6   Click the "Manually configured compression settings" check box.

Step 7   Select the CODEC value CCITT ulaw 8000Hz.

Step 8   Click the Up button until this CODEC value is at the top of the list.

Step 9   Click OK to exit.

Initiate a Call Using Microsoft NetMeeting

To initiate a call using Microsoft NetMeeting, perform the following steps:

Step 2   From the Call dialog box, select call using H.323 gateway.

Step 3   Enter the telephone number in the Address field.

Step 4   Click Call to initiate a call to the Cisco 3600 series router from Microsoft NetMeeting.