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This chapter shows you how to configure Voice over IP (VoIP) on the Cisco 3600 series. For a description of the commands used to configure Voice over IP, refer to the "Voice-Related Commands" chapter in the Voice, Video, and Home Applications Command Reference.
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 two or four voice interface cards (VICs), each of which is specific to a particular signaling type associated with a voice port. For more information about the physical characteristics, 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:
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:
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 3600 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.
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.
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. Device that typically uses pulse code modulation to transform analog signals into a digital bit stream and digital signals back into analog signals. 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.
DTMF---Dual tone multifrequency. Use of two simultaneous voice-band tones for dial (such as touch tone).
E&M---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. QoS 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 PBXs, 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.
Before you can configure your Cisco 3600 series router to use Voice over IP, you must first:
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.
To configure Voice over IP on the Cisco 3600 series, you need to complete the following tasks:
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:
(a) Multilink PPP with Interleaving
(b) RTP Header Compression
(c) Custom Queuing
(d) 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.
(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.
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.
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:
(a) 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.
(b) 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.
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 "Optimize Dial Peer and Network Interface Configurations" section for additional information about optimizing dial-peer characteristics.
6. Configure Voice Ports
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:
(a) FXO---Foreign Exchange Office interface
(b) FXS---The Foreign Exchange Station interface
(c) E&M---The "Ear and Mouth" interface (or "RecEive and TransMit" 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" chapter.
(Optional) Voice over IP can be used with Microsoft NetMeeting (Version 2.x) when the Cisco 3600 series router is used as the voice gateway. Refer to the 'Configure Voice over IP for Microsoft NetMeeting" section for more information about configuring Voice over IP to support Microsoft NetMeeting.
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:
In general, backbone routers perform the following QoS functions:
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:
Each of these components is discussed in the following sections.
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:
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:
To configure Multilink PPP and interleaving on a configured and operational interface or virtual interface template, use the following commands in interface mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| ppp multilink | Enable Multilink PPP. | ||
| ppp multilink interleave | Enable real-time packet interleaving. | ||
| ppp multilink fragment-delay milliseconds | Optionally, configure a maximum fragment delay. | ||
| ip rtp reserve lowest-UDP-port range-of-ports [maximum-bandwidth] | Reserve a special queue for real-time packet flows to specified destination User Datagram Protocol (UDP) ports, allowing real-time traffic to have higher priority than other flows. This is only applicable if you have not configured RSVP. |
For more information about Multilink PPP, refer to the "Configuring Media-Independent PPP and Multilink PPP" chapter in the Dial Solutions Configuration Guide.
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
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 4.
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:
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:
| Command | Purpose |
|---|---|
ip rtp header-compression [passive] | Enable RTP header compression. |
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.
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:
| Command | Purpose |
|---|---|
Specify the total number of RTP header compression connections supported on an interface. |
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 Network Protocols Configuration Guide, Part 1.
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 "Performing Basic System Management" chapter in the Configuration Fundamentals Configuration Guide.
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 "Performing Basic System Management" chapter in the Configuration Fundamentals Configuration Guide.
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:
interface Serial0/0 ip 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 ip mtu 300 ip address 40.0.0.7 255.0.0.0 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:
The subinterface has been configured as follows:
For more information about Frame Relay, refer to the "Configuring Frame Relay" chapter in the Wide-Area Networking Configuration Guide.
In Figure 5, 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 Router 1 (located to the left of the IP cloud) are (408) 115-xxxx, (408) 116-xxxx, and (408) 117-xxxx, where xxxx identifies the individual dial peers by extension. The destination pattern (or expanded telephone number) associated with Router 2 (located to the right of the IP cloud) is (729) 555-xxxx.
Table 5 shows the number expansion table for this scenario.
| Extension | Destination Pattern | Num-Exp Command Entry |
|---|---|---|
5.... | 40811..... | num-exp 5.... 408115.... |
6.... | 40811..... | num-exp 6.... 408116.... |
7.... | 40811..... | num-exp 7.... 408117.... |
1... | 729555.... | num-exp 2.... 729555.... |
The information included in this example needs to be configured on both Router 1 and Router 2.
To define how to expand an extension number into a particular destination pattern, use the following command in global configuration mode:
| Command | Purpose |
|---|---|
num-exp extension-number extension-string | Configure number expansion. |
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.
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 6 and Figure 7. A call leg is a discrete segment of a call connection that lies between two points in the connection. All the call legs for a particular connection have the same connection ID.
There are two different kinds of dial peers:
Four call legs make comprise and end-to-end call---two from the perspective of the source router as shown in Figure 6, and two from the perspective of the destination router as shown in Figure 7. 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.
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 8, 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 phone number with a specific IP address.
To configure call connectivity between the source and destination as illustrated in Figure 8, enter the following commands on router 10.1.2.2:
dial-peer voice 1 pots destination-pattern 1408555.... port 1/0/0 dial-peer voice 2 voip destination-pattern 1310555.... 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 "1310555" 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 "1408555" 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 9 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 9, enter the following commands on router 10.1.1.2:
dial-peer voice 4 pots destination-pattern 1310555.... port 1/0/0 dial-peer voice 3 voip destination-pattern 1408555.... session target ipv4:10.1.2.2
Using the example in Figure 5, 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. Router 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 5 shows a diagram of this small voice network.
Table 6 shows the peer configuration table for the example illustrated in Figure 5.
| Commands | |||||||
|---|---|---|---|---|---|---|---|
| Dial Peer Tag | Ext | Dest-Pattern | Type | Voice Port | session target | CODEC | QoS |
| Router 1 |
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1 | 6.... | +1408116.... | POTS |
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10 |
| +1729555.... | VoIP |
| IPV4 10.1.1.2 | G.729 | Best Effort |
| Router 2 |
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11 |
| +1408116.... | VoIP |
| IPV4 10.1.1.1 | G.729 | Best Effort |
4 | 2.... | +1729555.... | POTS |
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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:
| Command | Purpose |
|---|---|
dial-peer voice number pots | Enter the dial-peer configuration mode to configure a POTS peer. |
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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| destination-pattern string | Define the telephone number associated with this POTS dial peer. | ||
| port slot-number/subunit-number/port | Associate this POTS dial peer with a specific voice port. |
For example, suppose there is a voice call whose E.164 called number is 1(310) 555-2222. If you configure a destination-pattern of "1310555" and a prefix of "9," the router will strip out "1310555" 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-Related Commands" section of the Voice, Video, and Home Applications Command Reference.
Direct inward dial (DID) is used to determine how the called number is treated for incoming POTS call legs. As shown in Figure 10, 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:
The four defined dial-peer elements are:
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 beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| dial-peer voice number pots | Enter the dial-peer configuration mode to configure a POTS peer. | ||
| direct-inward-dial | Specify direct inward dial for this POTS peer. |
For additional POTS dial-peer configuration options, refer to the "Voice-Related Commands" section of the Voice, Video, and Home Applications Command Reference.
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:
| Command | Purpose |
|---|---|
dial-peer voice number voip | Enter the dial-peer configuration mode to configure a VoIP peer. |
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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| destination-pattern string | Define the destination telephone number associated with this VoIP dial peer. | ||
| session target {ipv4:destination-address | dns:host-name} | Specify a destination IP address for this dial peer. |
For additional VoIP dial-peer configuration options, refer to the "Voice-Related Commands" section of the Voice, Video, and Home Applications Command Reference. For examples of how to configure dial peers, refer to the section, "Voice over IP Configuration Examples."
You can check the validity of your dial-peer configuration by performing the following tasks:
To give real-time voice traffic precedence over other IP network traffic, use the following commands, beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| dial-peer voice number voip | Enter the dial-peer configuration mode to configure a VoIP peer. | ||
| ip precedence number | Select a precedence level for the voice traffic associated with that dial peer. |
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.
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, starting in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| dial-peer voice number voip | Enter the dial-peer configuration mode to configure a VoIP peer. | ||
| req-qos [best-effort | controlled-load | guaranteed-delay] | Specify the desired quality of service to be used. |
For example, to specify guaranteed delay QoS for VoIP dial peer 108, enter the following:
dial-peer voice 108 voip destination-pattern +14085551234 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.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| dial-peer voice number voip | Enter the dial-peer configuration mode to configure a VoIP peer. | ||
| acc-qos [best-effort | controlled-load | guaranteed-delay] | Specify the QoS value below which an SNMP trap will be generated. |
| Step | Command | Purpose | ||
|---|---|---|---|---|
| dial-peer voice number voip | Enter the dial-peer configuration mode to configure a VoIP peer. | ||
| codec [g711alaw | g711ulaw | g729r8] | Specify the desired voice coder rate of speech. |
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 +14085551234 codec g711alaw session target ipv4:10.0.0.8
| Step | Command | Purpose | ||
|---|---|---|---|---|
| dial-peer voice number voip | Enter the dial-peer configuration mode to configure a VoIP peer. | ||
| vad | Disable the transmission of silence packets (enabling VAD). |
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 +14085551234 vad session target ipv4:10.0.0.8
Voice over IP simulates a trunk connection by creating virtual trunk tie lines between PBXs connected to Cisco 2600 and 3600 series routers on each side of a VoIP connection. (See Figure 11.) In this example, two PBXs are connected using a virtual trunk. PBX-A is connected to Router A via an E&M voice port; PBX-B is connected to Router B via an E&M voice port. The Cisco routers spoof the connected PBXs into believing that a permanent trunk tie line exists between them.
The routers on both sides of the Voice over IP connection must be configured for trunk connections. For the scenario described in Figure 11, configure Router A to support trunk connections as follows:
configure terminal voice-port 1/0/0 connection trunk +15105554000 dial-peer voice 10 pots destination-pattern +13085551000 port 1/0/0 dial-peer voice 100 voip session-target ipv4:172.20.10.10 destination-pattern +15105554000
For the scenario described in Figure 11, configure Router B to support trunk connections as follows:
configure terminal voice-port 1/0/0 connection trunk +13085551000 dial-peer voice 20 pots destination-pattern +15105554000 port 1/0/0 dial-peer voice 200 voip session-target ipv4:172.19.10.10 destination-pattern +13085551000
To configure virtual trunk connections in Voice over IP, use the connection trunk command. The following conditions must be met for Voice over IP to support virtual trunk connections:
VoIP establishes the trunk connection immediately after it is configured. Both ports on either end of the connection are dedicated until you disable trunking for that connection. If for some reason the link between the two switching systems goes down, the virtual trunk re-establishes itself after the link comes back up.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| dial-peer voice number pots | Enter dial-peer configuration mode and define a tag number for a POTS dial peer. | ||
| destination-pattern [+]string | Specify the telephone number associated with the POTS dial peer. | ||
| port slot-number/subunit-number/port | Associate the POTS dial peer with a specific voice port on the Cisco end router. | ||
| dial-peer voice number voip | Define a tag number for a VoIP dial peer. | ||
| session target ipv4:destination-address | Identify the IP address of the appropriate port on the destination end router. | ||
| destination-pattern [+]string | Identify the destination pattern (telephone number) of the VoIP dial peer call leg on the destination end router. | ||
| exit | Exit dial-peer configuration mode. | ||
| configure terminal | Enter global configuration mode. | ||
| voice-port slot-number/sub-unit-number/port | Enter voice-port configuration mode. | ||
| connection trunk string | Specify a straight tie-line connection (virtual trunk connection). The string argument refers to the destination pattern (telephone number) configured for the destination VoIP dial peer. The value you configure for the connection trunk command must exactly match the value configured for the VoIP dial peer. |
To configure Voice over IP to support NetMeeting, create a VoIP peer that contains the following information:
To configure NetMeeting to work with Voice over IP, complete the following steps:
Step 1 From the Tools menu in the NetMeeting application, select Options. NetMeeting will display the Options dialog box.
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 AS5300 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.
To initiate a call using Microsoft NetMeeting, perform the following steps:
Step 1 Click the Call icon from the NetMeeting application. Microsoft NetMeeting will open the call dialog box.
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.
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:
These examples are described in the following sections.
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.
hostname rlb-1 ! Create voip dial peer 10 dial-peer voice 10 voip ! Define its associated telephone number and IP address destination-pattern +4155554000 session target ipv4:40.0.0.1 ! Request RSVP req-qos guaranteed-delay ! 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
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
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
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 session 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
The following example shows how to configure Voice over IP to link PBX users with E&M trunk lines.
Figure 13 illustrates the topology of this connection example.
hostname sanjose !Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 555.... port 1/0/0 !Configure pots dial peer 2 dial-peer voice 2 pots destination-pattern 555.... port 1/0/1 !Configure voip dial peer 3 dial-peer voice 3 voip destination-pattern 119.... 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
hostname saltlake !Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 119.... port 1/0/0 !Configure pots dial peer 2 dial-peer voice 2 pots destination-pattern 119.... port 1/0/1 !Configure voip dial peer 3 dial-peer voice 3 voip destination-pattern 555.... 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
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 14 illustrates the topology of this connection example.
! 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
! 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
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 15 illustrates the topology of this connection example.
! 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
! 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
Posted: Thu Jul 29 18:17:36 PDT 1999
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