Voice Gateways and Cisco Unified Communications Manager

Various gateway protocols are supported with Cisco Unified Communications Manager (CUCM). The gateway protocols include H.323, MGCP (Media Gateway Control Protocol), SIP (Session Initiation Protocol), and SCCP (Skinny Client Control Protocol). Protocol selection should be based on various criteria including the following:

  • • Cisco IOS proficiency level
  • • Hardware Selection
  • • Cisco IOS access level
  • • Telephony feature support
  • • Fault Tolerance Requirements
  • • IOS Revision

This blog entry will focus on the Cisco IOS proficiency level required to configure an H.323 or SIP dial peer. A high IOS proficiency level is required with H.323 and SIP gateway protocols because gateway requires extensive Cisco IOS dial-peer configuration. The dial-plan for call routing is configured on both the CUCM server and the gateway router in the case of both H.323 and SIP. H.323

  • Dial-peer voice 100 voip
  • Description Dial-peer to Cisco Unified Communication Manager 6.1
  • Preference 1
  • Destination-pattern 11…
  • Dtmf-relay h245-alphanumeric h245-signal
  • Session target ipv4:10.1.1.100
  • Incoming called-number .T
  • Codec g711ulaw
  • No vad
  • Ip qos dscp cs3 signaling

SIP

  • Dial-peer voice 100 voip
  • Description Dial-peer to Cisco Unified Communication Manager 6.1
  • Preference 1
  • Destination-pattern 11…
  • Dtmf-relay rtp-nte
  • Session protocol sipv2
  • Session target ipv4:10.1.1.100
  • Incoming called-number .T
  • Codec g711ulaw
  • No vad
  • Ip qos dscp cs3 signaling
Both of the above dial peers will route any 5 digit phone calls beginning with the digits 11 (destination-pattern) to the CUCM server at the 10.1.1.100 IP address (session target). The dial-peer destination-patterns are used for outbound call routing after matching the necessary dialed digits. Dual-Tone Multi-Frequency (DTMF) digits will be converted and passed via H.245 alphanumeric or signal for the H.323 dial-peer while the SIP dial-peer is using the named telephony events (NTE) capabilities of RFC2833 (DTMF-Relay). This VoIP dial-peer will also be used to answer inbound calls consisting of any digit strings passed to the router as long as one dialed digit is received (incoming called-number .T). The default audio codec used in a VoIP dial-peer is the compressed G.729 audio codec. Our VoIP dial-peers have been configured to use the G.711 uncompressed audio codec because CUCM is accessible over LAN interfaces. Voice Activity Detection (VAD) is a silence suppression mechanism which is turned on by default for VoIP dial peers. VAD saves bandwidth by not sending packets when there is silence in a conversation. The disadvantage of VAD is the voice clipping at the beginning and ending of voice samples when VAD is turned on and off. It is best practice to disable VAD unless routing calls over low bandwidth, high cost circuits. The ip qos cs3 signaling command was used to ensure the end-to-end QoS design follows the cs3 marking for signaling and not the older marking of AF31. Post to this blog the different aspects of dial peer configuration that you would add, change, or remove in your dial peer configurations. Post any questions you have regarding the configurations as well.

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