In an earlier reply, Sebin already tested dtmf-relay rtp-nte towards the ITSP.
You can see that in the following debug, rtp-nte is sent in the offer:
Jul 31 07:21:50.875: //3258888/62DC80800004/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:5068888@192.168.1.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK1FB1FC6F2
Remote-Party-ID: "SEBIN" <sip:044086471@44086666.etisalat>;party=calling;screen=yes;privacy=off
From: "SEBIN" <sip:044086471@44086666.etisalat>;tag=797CD02B-4E0
To: <sip:5068888@192.168.1.6>
Date: Tue, 31 Jul 2018 07:21:50 GMT
Call-ID: 3A301595-93C911E8-B3EFAA70-C66C772A@44086666.etisalat
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1658617984-0000065536-0000323275-0167936010
User-Agent: Cisco-SIPGateway/IOS-15.4.3.S6a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1533021710
Contact: <sip:044086471@192.168.1.25:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1403 6924 IN IP4 192.168.1.25
s=SIP Call
c=IN IP4 192.168.1.25
t=0 0
m=audio 31932 RTP/AVP 8 101
c=IN IP4 192.168.1.25
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
However, the ITSP does not respond with rtp-nte in its answer:
Jul 31 07:21:51.436: //3258888/62DC80800004/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK1FB1FC6F2
From: "SEBIN" <sip:044086471@44086666.etisalat>;tag=797CD02B-4E0
To: <sip:5068888@192.168.1.6>;tag=mc83h58c-CC-1018
Call-ID: 3A301595-93C911E8-B3EFAA70-C66C772A@44086666.etisalat
CSeq: 101 INVITE
Timestamp: 1533021710
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:5068888@192.168.1.6:5060;transport=udp>
Content-Length: 161
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 61748586 61748586 IN IP4 192.168.1.6
s=Sip Call
c=IN IP4 192.168.1.6
t=0 0
m=audio 41270 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
In the example which works, the offer to the ITSP also contains rtp-nte:
Jul 31 14:14:21.242: //3263406/039B62000004/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:600532273@192.168.1.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK1FC5D3B19
Remote-Party-ID: "SEBIN" <sip:044086471@44086666.etisalat>;party=calling;screen=yes;privacy=off
From: "SEBIN" <sip:044086471@44086666.etisalat>;tag=7AF6794B-2143
To: <sip:600532273@192.168.1.6>
Date: Tue, 31 Jul 2018 14:14:21 GMT
Call-ID: DA8E61F8-940211E8-9BE4AA70-C66C772A@44086666.etisalat
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0060514816-0000065536-0000324171-0167936010
User-Agent: Cisco-SIPGateway/IOS-15.4.3.S6a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1533046461
Contact: <sip:044086471@192.168.1.25:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 8016 5193 IN IP4 192.168.1.25
s=SIP Call
c=IN IP4 192.168.1.25
t=0 0
m=audio 21342 RTP/AVP 8 101
c=IN IP4 192.168.1.25
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Only, the ITSP did respond with an answer containing rtp-nte, unlike in the failed call:
Jul 31 14:14:21.669: //3263406/039B62000004/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK1FC5D3B19
From: "SEBIN" <sip:044086471@44086666.etisalat>;tag=7AF6794B-2143
To: <sip:600532273@192.168.1.6>;tag=5673c3x7-CC-1018
Call-ID: DA8E61F8-940211E8-9BE4AA70-C66C772A@44086666.etisalat
CSeq: 101 INVITE
Timestamp: 1533046461
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:600532273@192.168.1.6:5060;transport=udp>
Content-Length: 217
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 62135205 62135205 IN IP4 192.168.1.6
s=Sip Call
c=IN IP4 192.168.1.6
t=0 0
m=audio 43618 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
I suspect there is a problem on the ITSP side. You can prove that this is not dial-peer configuration, by making both dial-peers: 7704 and 77800 identical. A test Sebin already did earlier, by making the change to the DTMF relay to support only rtp-nte.
Sebin, whenever you perform a test, always capture the running config, debug ccsip messages, debug voip ccapi inout, and the show call active voice.
A nice trick to filtering the data in the show call active voice command, is to do this instead:
show call active voice | in PeerAddr|PeerId|RemoteS|RemoteM|Dtmf|Coder|VAD
I also note two other issues in this thread:
1) Do note that SIP INFO DTMF Relay is not supported by CUCM. And since most ITSPs will only support rtp-nte, there is not reason to ever use it.
Reference: INFO is not supported by Unified CM.
2) The media IP coming from CUCM to the CUBE is that of CUCM itself. This indicates that some media resource is being invoked for the call, and this may not be desireable. Now, it's happening on both the working and non-working calls, so I don't think it contributes to the DTMF issue, however, it could be a source of other problems. Be sure to not selected MTP Required on the SIP trunks, don't select Early Offer Forced (Use Best Effort) on your SIP Profiles, and use No Preference for your DTMF on the SIP Trunks.
