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Configuring Voice over IP on Cisco devices is simple because the same IOS that supports data networking tasks is used to configure voice. This section will introduce some of the more common Voice over IP configuration tasks. This section is intended not only to introduce real-world tasks that may be encountered, but also to provide a solid foundation of configuration information that will be necessary for the voice components of the Cisco CCIE Lab Exam.
Configuring FXS and FXO ports is trivial, as the default settings are generally suitable for most implementations.
Some of the more commonly configured settings are:
Port description
Signal type
Call progress tone
Voice activity detection
Comfort noise
The following code reveals how to configure these options on FXS/FXO ports:
router(config)#voice-port 1/0/0 router(config-voiceport)#description Test Voice Port router(config-voiceport)#cptone AU router(config-voiceport)#signal [loop-start | ground-start]
Generally, only one person speaks at a time during a telephone conversation (unless you are speaking with my mother-in-law!), and thus, in traditional voice networks, more than 50% of bandwidth is wasted in any voice call.
Enabling VAD allows for this otherwise wasted bandwidth to be used for other purposes. VAD detects when someone is speaking and cuts off voice traffic when there is silence. This allows the VoIP implementation to not transmit silence packets. VAD does have some limitations, however. The most prominent is clipped speech, which is due to VAD sometimes kicking in just after a person starts talking. This clipping is generally not noticeable, and the bandwidth savings generally outweighs any drawbacks. Another somewhat disconcerting effect of enabling VAD is that when there is no user speaking, it sounds as though the call has been disconnected. This can be overcome by enabling comfort noise, which generates some virtual background noise, allowing the user to recognize that the call is still connected. When VAD is enabled, comfort noise is also enabled by default.
VAD is configured on dial peers as follows:
router(config-voiceport)#vad router(config-voiceport)#comfort-noise
When configuring E&M ports, it is critical to remember that the default settings are usually not sufficient to enable voice transmission. This is because E&M ports are designed to connect directly to a PBX and must match the specifications of that equipment. The following settings must be configured:
Signal type
Call progress tone
Operation
Type
Impedance
router(config)#voice-port 1/0/0 router(config-voiceport)#signal [wink-start | immediate | delay-dial] router(config-voiceport)#cptone CountryCode router(config-voiceport)#operation [2-wire | 4-wire] router(config-voiceport)#type [1 | 2 | 3 | 5] router(config-voiceport)#impedance [600c | 600r | 900c | complex1 | complex2]
As with FXS/FXO ports, the description, VAD, and comfort noise can also be configured.
The connection command can be used to set special modes for voice ports. If the connection command is not implemented, the voice port will provide dial tone and collect digits from the handset or end device as per usual. The following list highlights the connection options and their use:
Private line automatic ringdown (PLAR) -- Used to automatically dial a destination pattern as soon as the receiver on the phone is lifted and the port is activated. No digits need to be entered to establish the connection.
Tie line -- Specifies that the particular port is a tie line connection to a PABX.
Trunk -- The "connection trunk" mode can be used for E&M-to-E&M trunks, FXO-FXS trunks, and FXS-FXS trunks.
PLAR-OPX -- Specifies a PLAR connection to an off-premises extension. Using this option, the local voice port provides a local response before the remote voice port receives an answer. This ensures that the call is answered before the call flow is completed.
One of the most commonly used connection commands is the PLAR function. This allows a user to lift the handset and be automatically connected to a remote number. Configuring PLAR is simply a matter of adding the connection command under the voice port and specifying the destination number to be called:
router(config)#voice-port 1/0/0 router(config-voiceport)#connection plar 5512341234
Correct PLAR configuration can be verified by issuing the show voice port command:
router#show voice port 1/0/0
Foreign Exchange Office
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is plar
Connection Number is 5512341234
As discussed previously, the dial plan is the heart of any VoIP implementation. Dial peers are the configuration element that implements the dial plan. Dial plan design and strategies will be discussed further in a forthcoming Study Guide. The following section will detail how to configure basic dial peers.
There are two forms of dial peers: POTS and voice network (VoIP, VoFR, and VoATM). The following diagram highlights the different call legs:
Figure 6. Call Legs
The POTS dial peer only requires the following for a basic setup:
Dial Peer TagsI strongly advocate the practice of using a dial peer tag that matches the destination pattern. This reduces troubleshooting time and makes configurations much easier to read. |
Tag number -- An arbitrary number used for identification; however, I recommend using the destination number as the tag. This makes for simplified troubleshooting and configuration.
Destination pattern -- The telephone number of the end device connected to this port.
Voice port -- The physical port on the router that the call will be passed to.
Very few commands are required to create a basic POTS dial peer. The following configures the router to send all calls destined for 1234 to the telephone on voice port 1/0/0:
router#configure terminal router(config)#dial-peer voice 1234 pots router(config-dial-peer)#destination-pattern 1234
As for the POTS dial peer, the voice network dial peer is also simple to configure. VoFR and VoATM configurations are discussed in later sections. The following parameters are all are required to be defined for a VoIP dial peer:
Tag number
IP address for session target
Destination telephone number pattern
Codecs Must Match!Note that it is critical that the codecs on both voice peers match; otherwise, the call will fail. |
The following example configures the router for a VoIP dial peer at 192.168.1.1 for a destination pattern (telephone number) of 1234 using codec G711.alaw:
router(config)#dial-peer voice tag-number voip router(config-dial-peer)#session target ipv4:192.168.1.1 router(config-dial-peer)#codec g711alaw router(config-dial-peer)#destination-pattern 1234
Another aspect of voice configuration is digit manipulation. Number expansion is a technique in which digits can be prepended to the number the user dials. This is commonly used by PABX maintainers for extension-level dialing. For example, a subscriber's desk number may be 1-234-5678; however, for ease of use, the network may be configured so that the user only needs to dial 5678 to reach that phone. By configuring number expansion, we can instruct the router to prepend the digits 1234 to the 5678 to complete the number.
router(config)num-exp 5678 12345678
Wildcards can be used with number expansion. For example, if you wished to configure all four-digit extensions that start with the digit 5 to be expanded to add the prefix 999, you could configure
router(config)num-exp 5... 9995...
Following is an example of a partial configuration of a corporate headquarters router. The configuration includes only the explicit portion needed to configure the Voice over Frame Relay network.
Figure 8. Frame Relay Network Topology
Configure the serial interface and subinterfaces:
Frame Relay must be the encapsulation of the serial interface. Frame Relay traffic shaping must also be defined here.
On the subinterfaces, define the segmentation for the PVC. Use frame-relay interface-dlci to configure voice on this interface. voice-encap tells the IOS that the DLCI will be used to transport voice. Following voice-encap is the segment size. Last, the Frame Relay map class is stated.
interface serial 0 encapsulation frame-relay frame-relay traffic-shaping interface serial 0.1 point-to-point ip address 10.1.1.1 255.255.255.0 frame-relay interface-dlci 101 voice-encap 80 class FRtype1 interface serial 0.2 point-to-point ip address 10.1.2.1 255.255.255.0 frame-relay interface-dlci 102 voice-encap 80 class Frtype1 interface serial 0.3 point-to-point ip address 10.1.3.1 255.255.255.0 frame-relay interface-dlci 103 voice-encap 160 class Frtype2
Configure the Frame Relay map class:
Configure BECN to be a part of traffic shaping for further throttling of the circuit.
State the CIR to allow the IOS to properly throttle the DLCI.
BC stands for burst control. This is the excess bandwidth allowed for bursting.
Map-class frame-relay Frtype1 frame-relay adaptive-shaping becn frame-relay cir 128000 frame-relay bc 1000 map-class frame-relay Frtype2 frame-relay adaptive-shaping becn frame-relay cir 256000 frame-relay bc 1000
Configure Frame Relay dial peers:
dial-peer voice 101 VOFR destination-pattern 5... session-target serial 0 101 dial-peer voice 102 VOFR destination-pattern 5... session-target serial 0 102 dial-peer voice 103 VOFR destination-pattern 5... session-target serial 0 103
Configuring Voice over ATM is very similar to configuring Voice over Frame Relay. VoATM is diminishing in support as the reach of ATM is diminishing. The following example shows a basic configuration for VoATM with a destination number of 1234 and a VPI/VCI defined by the carrier as 30/100. The ATM PVC is configured on interface ATM 1/1.
router(config)#dial-peer voice 1234 voatm
router(config-dial-peer)#destination-pattern 1234
router(config-dial-peer)#session target ATM 1/1 pvc 30/100
As well as allowing configuring dedicated voice ports to support VoIP, Cisco routers can also be configured to support voice calls over dedicated digital connections such as ISDN BRI and PRI links.
The following code is a sample configuration for connecting a Cisco router to an ISDN BRI carrier service or a BRI port on a PABX:
router(config)# isdn switch-type switch-type
router(config)# interface bri slot|port
router(config-if)# no ip address
router(config-if)# isdn incoming-voice voice
router(config-if)# shutdown
router(config-if)# isdn layer1-emulate {user | network}
router(config-if)# no shutdown
router(config-if)# [no] line-power
router(config-if)# isdn protocol-emulate {user | network}
router(config-if)#^Z
There is a little more work required when configuring PRI connections to a router. At a minimum, you must define the following:
Interface type
Signaling interface
Line encoding
Framing format
Participating channels
A sample configuration follows:
router#config terminal router(config)#isdn switch-type basic-ni1 router(config)#controller t1 1/0 router(config-controller)#framing esf router(config-controller)#clock source internal router(config-controller)#linecode b8zs router(config-controller)#pri-group timeslots 1-24 router(config-controller)#^Z
Q.931 is used for the setup and teardown of the connection in the ISDN circuit. It is located at the Network Layer in the protocol stack. The basic steps to configure ISDN PRI with Q.931 support are as follows:
router(config)#isdn switch-type primary-net5
router(config)# controller {T1 | E1} slot/port
router(config-controller)# pri-group timeslots range
router(config-controller)# exit
router(config)# interface serial0/0:n
router(config)#isdn protocol-emulate {network | user}
router(config-if)# [no] line-power
router(config-if)# isdn incoming-voice voice
router(config-if)# ^Z
QSIG configuration on a Cisco router allows the router to appear to a PABX as though it is in fact the PSTN.
The two commands used for configuring QSIG with an ISDN PRI circuits are:
router(config)#isdn switch-type primary-qsig
router(config-if)#isdn protocol-emulate {user | network}
It is a trivial task to configure CAS support on a Cisco router:
router(config)#controller E1 0/0 router(config-controller)#framing crc-4 router(config-controller)#clock source line primary router(config-controller)#linecode hdb3 router(config-controller)#cas-group 0 timeslots 1-31 type e&m-fgb dtmf dnis
The various CAS signaling types are listed in the following table:
Table 9. CAS Signaling Types
| Signaling Type | Description |
| e&m-fgb | E&M Type II FGB |
| e&m-fgd | E&M Type II FGD |
| e&m-immediate-start | E&M Immediate Start |
| fxs-ground-start | FXS Ground Start |
| fxs-loop-start | FXS Loop Start |
| p7 | P7 switch |
| R2-analog | R2 ITU Q411 |
| R2-digital | R2 ITU Q421 |
| R2-pulse | R2 ITU Supplement 7 |
| sas-ground-start | SAS Ground Start |
| sas-loop-start | SAS Loop Start |
As with CAS, configuring Common Channel Signaling support is also quite straightforward:
router(config)#isdn switch-type primary-dms100 router(config)#controller T1 0/0 router(config-controller)#framing esf router(config-controller)#clock source line primary router(config-controller)#linecode b8zs router(config-controller)#pri-group timeslots 1-24 router(config-controller)#exit router(config-dial-peer)#dial-peer voice 100 pots router(config-dial-peer)#destination-pattern 10.. router(config-dial-peer)#direct-inward-dial router(config-dial-peer)#port 0:D
[IE-Mult-WP2-F04]
[2002-12-24-01]
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