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CUBE - No audio in a call between Multi-VRF's

I have a problem with the RTP stream in a call between multi-vrf on a CUBE (2911)

Here is the config of the CUBE:

--------- VOICE SERVICE VOIP ----------

voice service voip
 address-hiding
 mode border-element
 allow-connections sip to sip
 redundancy
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 no supplementary-service sip handle-replaces
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  header-passing
  error-passthru
  midcall-signaling passthru media-change
  early-offer forced
  sip-profiles 100
  no call service stop

--------- DIAL-PEER INCOMING FROM PSTN (DEFAULT VRF) ----------

dial-peer voice 100 voip
 description --- INCOMING VoIP PEER FROM PSTN ---
 call-block translation-profile incoming SCAM-BLACKLIST
 call-block disconnect-cause incoming call-reject
 session protocol sipv2
 session transport udp
 incoming called-number .
 voice-class media 1
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad

--------- DIAL-PEER OUTGOING TO CUCM1 (DEFAULT VRF) ----------

dial-peer voice 201 voip
 description --- OUTGOING VoIP PEER TO CUCM ---
 destination-pattern <PHONENUMBER1>
 session protocol sipv2
 session target ipv4:<IP_CUCM1>
 session transport udp
 voice-class codec 1
 no voice-class sip session refresh
 voice-class sip bind control source-interface GigabitEthernet0/1.204
 voice-class sip bind media source-interface GigabitEthernet0/1.204
 dtmf-relay rtp-nte
 no vad

--------- DIAL-PEER OUTGOING TO CUCM2 (VRF 1600) ----------

dial-peer voice 1601 voip
 description --- OUTGOING VoIP PEER TO CUCM ---
 destination-pattern <PHONENUMBER2>
 session protocol sipv2
 session target ipv4:<IP_CUCM2>
 session transport udp
 voice-class codec 1
 no voice-class sip session refresh
 voice-class sip bind control source-interface GigabitEthernet0/1.1654
 voice-class sip bind media source-interface GigabitEthernet0/1.1654
 dtmf-relay rtp-nte
 no vad

When dialing from a phone on CUCM1 to a phone on CUCM2 the following RTP is set-up

11    154844     154845     18616    27000    <IP_CUBE_INTERNAL1>                               <IP_CUCM1>                              NO    CLOUD-0XXX
 callId 154844 (dir=1): called=<PHONENUMBER2> calling=<PHONENUMBER1> redirect= loopback=NO confID=-1 mode=3 rtp(tx:0/rx:485) rtcp(tx:0/rx:0) MPSS NO VRF CLOUD-0XXX           peer callId 154845: called=<PHONENUMBER2> calling=<PHONENUMBER1> redirect= , confID:-1
 , vrf = CLOUD-0XXX             1 context 0x3FA79BC0 xmitFunc 0x3656BCE0
12    154845     154844     18796    18734    <IP_CUBE_OUTSIDE-VRF(DEFAULT)>                                <IP_CUBE_OUTSIDE-VRF(DEFAULT)>                                NO    NA
 callId 154845 (dir=2): called=<PHONENUMBER2> calling=<PHONENUMBER1>redirect= loopback=NO confID=-1 mode=3 rtp(tx:485/rx:0) rtcp(tx:0/rx:0) MPSS NO VRF NA                   peer callId 154844: called=<PHONENUMBER2> calling=<PHONENUMBER1> redirect= , confID:-1
 , vrf = NA                     1 context 0x3FA899E0 xmitFunc 0x3656BCE0
13    154846     154847     18734    18796    <IP_CUBE_OUTSIDE-VRF(DEFAULT)>                               <IP_CUBE_OUTSIDE-VRF(DEFAULT)>                                NO    NA
 callId 154846 (dir=1): called=<PHONENUMBER2> calling=<PHONENUMBER1> redirect= loopback=NO confID=-1 mode=3 rtp(tx:513/rx:0) rtcp(tx:0/rx:0) MPSS NO VRF NA                   peer callId 154847: called=<PHONENUMBER2> calling=<PHONENUMBER1> redirect= , confID:-1
 , vrf = NA                     1 context 0x3FA2F7C0 xmitFunc 0x3656BCE0
14    154847     154846     18628    27466    <IP_CUBE_INTERNAL2>                               <IP_CUCM2>                              NO    CLOUD-16XX
 callId 154847 (dir=2): called=<PHONENUMBER2> calling=<PHONENUMBER1> redirect= loopback=NO confID=-1 mode=3 rtp(tx:0/rx:513) rtcp(tx:0/rx:0) MPSS NO VRF CLOUD-16XX           peer callId 154846: called=<PHONENUMBER2> calling=<PHONENUMBER1> redirect= , confID:-1
 , vrf = CLOUD-16XX             1 context 0x3D97B740 xmitFunc 0x3656BCE0

So signaling of the call via SIP is working, but when we answer the call, RTP is not routed between the two VRF's (default and 16XX)

Any advice is welcome !!!!!!

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