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This chapter explains how to configure voice network modules with recEive and transMit (E&M), Foreign Exchange Office (FXO), and Foreign Exchange Station (FXS) interfaces for your router. Voice network modules convert telephone voice signals into a form that can be transmitted over an IP network. This chapter is divided into the following sections:
You need both a voice network module and a voice interface card for a voice connection. You can install one voice interface card in a 2-channel voice network module, and two voice interface cards in a 4-channel module. At least one other network module or WAN interface card must be installed in the router to provide the connection to the IP LAN or WAN.
Voice over IP (VoIP) enables your router to carry live voice traffic (for example, telephone calls and faxes) over an IP network. VoIP offers the following benefits:
The following is a list of terms common to voice technology:
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---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 VoIP, it specifies the voice coder rate of speech for a dial peer.
Dial peer---Addressable call endpoint. In VoIP, there are two kinds of dial peers: POTS and VoIP. POTS dial peers establish calls to and from a physical voice port on the router. VoIP dial peers establish calls to and from a voice port on another router, using VoIP.
DS0---64-kbps 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 valuable for Off-Premises eXtension (OPX) 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 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 VoIP, this is an IP network. VoIP peers point to specific VoIP devices.
Before you can configure your router to use VoIP, you must first do the following:
Whenever you install a new interface, or if you want to change the configuration of an existing interface, you must configure the interface. If you replace a module that was already configured, the router recognizes it and brings up the interface in the existing configuration.
Before you configure an interface, have the following information available:
 | TimeSaver Obtain this information from your system administrator or network plan before you begin router configuration. |
To configure a voice interface, you must use configuration mode (manual configuration). In this mode, you can enter Cisco IOS commands at the router prompt.
Before you begin, disconnect all WAN cables from the router to keep it from trying to run the AutoInstall process. The router tries to run AutoInstall whenever you power it ON if there is a WAN connection on both ends, and the router does not have a valid configuration file stored in NVRAM (for instance, when you add a new interface). It can take several minutes for the router to determine that AutoInstall is not connected to a remote Transmission Control Protocol/Internet Protocol (TCP/IP) host.
To enter configuration mode, follow this procedure:
Step 1 Connect a console to the router. If you need instructions for connecting a console, refer to the installation chapter of your router installation and configuration guide. Power ON the router.
Step 2 If the current configuration is no longer valid, after about one minute you see the following prompt:
Would you like to enter the initial dialog? [yes/no]:
Answer no. You now enter the normal operating mode of the router.
Step 3 After a few seconds, you see the user EXEC prompt (Router>). Type enable and the password to enter enable mode:
Router> enable Password: <password>
Configuration changes can be made only in enable mode. The prompt changes to the privileged EXEC (enable) prompt (Router#):
Router#
Step 4 Enter the configure terminal command to enter configuration mode:
Router# configure terminal Router(config)#
The router enters global configuration mode, indicated by the Router(config)# prompt.
Step 5 If you have not configured the router before, or want to change the configuration, configure global parameters, passwords, network management, and routing protocols. In this example, IP routing is enabled:
Router(config)# ip routing
For complete information about global configuration commands, refer to the Cisco IOS configuration guides and command references.
Step 6 If you have not already done so, configure the network module or WAN interface card that you plan to use for IP traffic. For instructions, see your router's installation and configuration guide or the configuration note for the network module or WAN interface card.
Step 7 To configure another interface, enter the exit command to return to the Router(config)# prompt.
Step 8 To configure the router for voice traffic, refer to the detailed instructions in the Voice over IP Configuration document.
Step 9 When you finish configuring interfaces, exit configuration mode and return to the enable prompt by pressing Ctrl-z. To see the current operating configuration, including any changes you just made, enter the show running-config command:
Router# show running-config
To see the configuration currently stored in NVRAM, enter the show startup-config command at the enable prompt:
Router# show startup-config
Step 10 The results of the show running-config and show startup-config commands differ from each other if you have made changes to the configuration, but have not yet written them to NVRAM. To write your changes to NVRAM, making them permanent, enter the copy running-config startup-config command at the enable prompt:
Router# copy running-config startup-config Building configuration. . . [OK] Router#
The router is now configured to boot in the new configuration.
The actual VoIP configuration procedure you complete depends on the 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 VoIP for simple FXS-to-FXS connections.
In this example, a very small company, consisting of two offices, has decided to integrate VoIP into its existing IP network. One basic telephony device is connected to Router RLB-1; therefore Router RLB-1 has been configured for one POTS peer and one VoIP peer. Router RLB-w and Router R12-e establish the WAN connection between the two offices. Because one POTS telephony device is connected to Router RLB-2, it has also been configured for only one POTS peer and one VoIP peer.
In this example, only the calling end (Router RLB-1) is requesting RSVP. Figure 4-1 illustrates the topology of this FXS-to-FXS connection example.
hostname rlb-1 ! Create voip dial-peer 10 dial-peer voice 10 voip ! Define its associated telephone number and IP address destination-pattern +4155264000 sess-target ipv4:40.0.0.1 ! Request RSVP req-qos guaranteedDelay ! Create pots dial-peer 1 dial-peer voice 1 pots ! Define its associated telephone number and voice port destination-pattern +4085264000 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 +4155264000 port 1/0/0 ! Create voip dial-peer 20 dial-peer voice 20 voip !Define its associated telephone number and IP address destination-pattern +4085264000 sess-target ipv4:10.0.0.1 ! Configure serial interface 0/0 interface Serial0/0 ip address 40.0.0.1 255.0.0.0 no ip mroute-cache ! Configure RTP header compression ip rtp header-compression ip rtp compression-connections 25 ! Enable RSVP on this interface ip rsvp bandwidth 96 96 fair-queue 64 256 3 clockrate 64000 ! Configure IGRP router igrp 888 network 10.0.0.0 network 20.0.0.0 network 40.0.0.0
The following example shows how to configure VoIP to link PBX users with E&M trunk lines.
In this example, a company wants to connect two offices: one in San Jose, California and the other in Salt Lake City, Utah. Each office has an internal telephone network using PBX, connected to the voice network by an E&M interface. Both the Salt Lake City and the San Jose offices are using E&M Port Type II, with four-wire operation and ImmediateStart signaling. Each E&M interface connects to the router using two voice interface connections. Users in San Jose dial "8-569" and then the extension number to reach a destination in Salt Lake City. Users in Salt Lake City dial "4-527" and then the extension number to reach a destination in San Jose.
Figure 4-2 illustrates the topology of this connection example.
hostname sanjose !Configure pots dial-peer 1 dial-peer voice 1 pots destination-pattern +527.... port 1/0/0 !Configure pots dial-peer 2 dial-peer voice 2 pots destination-pattern +527.... port 1/0/1 !Configure voip dial-peer 3 dial-peer voice 3 voip destination-pattern +569.... 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 +569.... port 1/0/0 !Configure pots dial-peer 2 dial-peer voice 2 pots destination-pattern +569.... port 1/0/1 !Configure voip dial-peer 3 dial-peer voice 3 voip destination-pattern +527.... 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 VoIP 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 4-3 illustrates the topology of this connection example.
! Configure pots dial-peer 1 dial-peer voice 1 pots destination-pattern +14085274000 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 +14085274000 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 VoIP to link users with the PSTN gateway using an FXO connection (PLAR mode).
In this example, PSTN users in Salt Lake City, Utah, can dial a local number and establish a private line connection in a remote location. As in the previous example, Router SLC in Salt Lake City is connected directly to the PSTN through an FXO interface.
Figure 4-4 illustrates the topology of this connection example.
! Configure pots dial-peer 1 dial-peer voice 1 pots destination-pattern +14085274000 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 +14085274000 session target ipv4:172.16.1.123 ! Configure the voice port voice port 1/0/0 connection plar 14085274000 ! Configure the serial interface interface serial 0/0 ip address 172.16.65.182 no shutdown
The following example shows how to configure a BRI port for direct-inward dialing (DID). This configuration allows the called number information from the ISDN Q.931 setup message to be used for routing on an ISDN line.
In this example, a call comes into router 1 on the BRI port. The DID information allows the router to route the call based on the called number. If the called number is 2xxx, the call is routed to router 2000, and if the called number is 3xxx, the call is routed to router 3000.
Figure 4-5 illustrates the topology of this connection example.
dial-peer voice 1 pots port 1/0/0 destination-pattern 1... direct-inward-dial dial-peer voice 2 voip session target ipv4:1.1.1.2 destination-pattern 2... dial-peer voice 3 voip session target ipv4:1.1.1.3 destination-pattern 3...
dial-peer voice 1 pots port 1/0/0 destination-pattern 2000
dial-peer voice 1 pots port 1/0/0 destination-pattern 3000
At this point you can proceed to the following:
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