Support / Voice / SBC / Cisco CUBE

Cisco CUBE SIP trunk to VoiceTel.

Cisco CUBE running on ISR-G2/G3, ISR4000, ASR1000, CSR1000v, or Cat8k uses dial-peers, voice class objects, and voice service voip stanzas. The skeleton below covers the VoiceTel-facing trunk; bind it to your downstream call-leg dial-peer separately.

Global voice service config

voice service voip
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
 allow-connections sip to sip
 supplementary-service media-renegotiate
 sip
  bind control source-interface GigabitEthernet0/0/0
  bind media source-interface GigabitEthernet0/0/0
  registrar server expires max 360 min 60
  early-offer forced
  midcall-signaling passthru
  pass-thru content custom-sdp
 !

Voice class codec preference

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729

SIP-UA registrar (Digest auth)

sip-ua
 credentials username <sip-username> password 7 <sip-password> realm <sip-host-from-portal>
 authentication username <sip-username> password 7 <sip-password>
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 retry register 3
 timers connect 1000
 sip-server dns:<sip-host-from-portal>
 reason-header override
 nat symmetric check-media-src
 registrar dns:<sip-host-from-portal> expires 360 tcp
 transport udp

For IP authentication (no Digest), drop the credentials and authentication lines and remove the registrar line — VoiceTel authorizes by source IP.

Outbound dial-peer (toward VoiceTel)

dial-peer voice 100 voip
 description Outbound to VoiceTel
 destination-pattern .T
 session protocol sipv2
 session target dns:<sip-host-from-portal>
 session transport udp
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 voice-class sip rel1xx supported "100rel"
 voice-class sip dtmf-relay rtp-nte
 dtmf-relay rtp-nte
 no vad

Inbound dial-peer (from VoiceTel)

dial-peer voice 200 voip
 description Inbound from VoiceTel
 session protocol sipv2
 incoming called-number .T
 voice-class codec 1
 voice-class sip rel1xx supported "100rel"
 dtmf-relay rtp-nte
 no vad

Verify

show sip-ua register status
show dial-peer voice summary
show voip rtp connections
debug ccsip messages   # for SIP trace if registration fails