Cisco IOS Dial-Peers: VoIP Dial Peer Configurations

In the last blog, two VoIP dial peers were created to point to two different CUCM servers performing call processing for the cluster. We will begin to discuss the dial-peer options of dial-peer 100 below: Dial-peer voice 100 voip Destination-pattern 11… Ip qos dscp cs3 signaling Dtmf-relay h245-alphanumeric No vad Codec g711ulaw Preference 1 Session target ipv4: VoIP dial-peers forward all digits by default. This behavior is very different than the default digit stripping rules of POTS dial peers. POTS dial-peer digit stripping rules mandate that only wildcard digits are forwarded by default. Since the intent of the configuration is to forward 5 digits to CUCM, no digit manipulation commands have been used in dial-peer 100 (no digit-strip, forward-digits, prefix, etc.). CUCM version 4.0 changed the default QoS signaling marking from AF31 to CS3. Although the ASIC of Cisco IP phone is responsible for marking the DSCP value for all signaling and media packets in IP telephony, the Enterprise Parameter configuration of CUCM can be changed to create different markings. The QoS SRND has been suggesting the use of CS3 for signaling for many years now and most IP phone deployments use CS3 for signaling. Cisco IOS VoIP dial-peers use the older AF31 QoS marking by default, but this can be changed as depicted by the “ip qos dscp cs3 signaling” command in dial-peer 100. Dual tone multi-frequency (DTMF) relay allows digits pressed during a conversation to be sent over Voice over IP. There are many ways to send DTMF digits and multiple methods can be configured in a prioritized list in a single dial peer. H245 is the media negotiation layer of H.323 that is responsible for negotiating codecs and opening logical channel connections. The “dtmf-relay h245-alphanumeric: command specifies that the dial-peer should send DTMF digits in the signaling path (h245) and not the media path (RTP). In the following example, both h245-alphanumeric and h245-signal are configured, but h245-alphanumeric is preferred because it appears as the first option: Dtmf-relay h245-alphanumeric h245-signal Dial-peers use the H.323 protocol by default. In future blogs, we will discuss converting VoIP dial-peers to session initiation protocol (SIP) and investigate the in band (RTP) RFC2833 dtmf-relay mechanism (dtmf-relay rtp-nte). All VoIP dial-peers run voice activity detection (VAD)/silence suppression by default. Although the “no vad” command is used to turn off this mechanism, the “show dial-peer voice” command displays VAD as silence suppression. VAD is a mechanism that turns off the transmission of packets when there is silence in a conversation (without ending the call). VAD can save bandwidth in a conversation, but has the disadvantage of clipping out the beginning and ending syllables of speech. VAD is turned on by default in VoIP dial-peers, but is off by default on Cisco IP phones controlled by both CUCM and CUCME. The bandwidth savings of VAD is normally not worth the quality issues associated with using the technology. VAD parameters can be tuned on an individual dial peer basis, but it is normally best to turn off the mechanism. VAD can be turned on in CUCM (if desired) by changing the Call Manager service parameters for Silence Suppression. It is recommended to leave VAD turned off in CUCM unless it is absolutely necessary. Cisco IOS VoIP dial-peers use the compressed G.729 codec by default. In our example, we are using dial-peer 100 to route inbound PSTN calls from the ISDN PRI circuit to CUCM. Most environments are connecting their voice gateway routers to the network via Fast Ethernet (100Mbps) or Gigabit Ethernet (1000Mbps) technology where there is little to no need for the bandwidth savings of G.729. The codec default was change to G.711ulaw, but a voice codec configuration could have been used to allow H.245 to negotiate between both G.711ulaw and G.729. The global configuration below would be used to set up the voice-class codec: Voice class codec 1 Codec preference 1 g711ulaw Codec preference 2 g729r8 Each VoIP dial-peer can now leverage the global codec voice class as shown below: Dial-peer voice 100 voip Voice-class codec 1 The session target command is used in VoIP dial-peers to point the dial-peer to an IP address, while POTS dial-peers used the port command to steer calls to a TDM interface on the router. The next blog will discuss the preference command used with dial-peer hunting and continue our conversation of VoIP dial-peers. REFERENCES Global Knowledge Cisco Unified Communications classes Cisco Press Books – CVOICE / GWGK / TUC Voice Activity Detection / Silence Suppression

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