We have this very odd issue:
Incoming call from external number to internal extension that is forwarded to another external number is connecting, but has no audio.
Phone A (any external number, but in this example +359889610844) dials +496152930142; this goes to our internal number Phone B. If Phone B doesn’t answer in 5 seconds (usually higher value, but we set it to 5 for the test) it is forwarded to Phone C (again any external number, but in the example +359885890203).
While Phone B is ringing Phone A hears the ringback tone. As soon as the 5 seconds expire and the call is sent to Phone C, Phone A stops hearing anything. Phone C can still answer the call, but both Phone A and Phone C don’t hear anything.
If Phone A is internal number and dials the other internal number Phone B, then the call reaches the external Phone C with no problem and call is successful with two-way audio.
We reviewed debugs from CUBE and here is what we see:
INVITE sip:+firstname.lastname@example.org:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 184.108.40.206:5060;branch=z9hG4bK6711e1970bb02b8a77b6edf2802e011d.6122ffbd
To: +496152930142 <sip:+email@example.com;transport=udp;user=phone>
CSeq: 317357 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REGISTER, UPDATE
o=- 1197381055 1197381056 IN IP4 220.127.116.11
c=IN IP4 18.104.22.168
m=audio 16476 RTP/AVP 8 116 0 108 102 18 4 109
INVITE sip:+firstname.lastname@example.org:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.178.2:5060;branch=z9hG4bKC8E1025B6
Date: Tue, 17 Sep 2019 07:34:36 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
o=CiscoSystemsSIP-GW-UserAgent 4242 3444 IN IP4 192.168.178.2
m=audio 22006 RTP/AVP 8 0 101
c=IN IP4 192.168.178.2
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 192.168.178.2:5060;received=22.214.171.124;branch=z9hG4bKC8E11144A
CSeq: 102 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE
o=- 3560920190817093400 1500643355 IN IP4 126.96.36.199
c=IN IP4 188.8.131.52
m=audio 11370 RTP/AVP 8 101
At this point the caller stops hearing anything.
From here on the two phones wait for 5-10 seconds of the silence and they hang up. The call is disconnected normally.
From signalling point of view everything looks just fine for me. The call reaches the destination, it is answered, then disconnected all normally. But the media portion is lost somewhere along the way.
What I sent as questions to the ITSP is:
I am suspecting that the issue might be with one of those things, but I am not sure.
Does anybody have a clue for something in the CUBE, which might be causing the issue? I am attaching the running config and debugs ccsip messages and voice ccapi.
Any help will be greatly appreciated!
It's surprising that nobody who posted here took a few moments out of their day to share their solution. Wish everyone here could be down voted.
This was the answer in my case:
The ITSP's firewall wouldn't setup RTP until we initiated RTP. In the case of a forwarded call, you're in a catch-22 situation. The customer will never send RTP audio to the ITSP since they will never receive any RTP audio from the ITSP for the initial inbound call. You can do a packet capture on the customer interface facing the ITSP to verify this, and send it to your ITSP and have them correct their firewall to handle this situation.
The workarounds are to enable an MTP in CUCM if using CUCM, but this is best to be avoided since now you will have media for all calls flowing through CUCM instead of directly to the gateway, and using MTPs can create another set of issues. The second option is to configure STUN for firewall traversal and have it send some bogus RTP packets.
As you seem to be using Deutsche Telecom I think that this post might be of help. https://community.cisco.com/t5/unified-communications/cube-config-sip-trunk-to-deutsche-telekom/m-p/4447706/highlight/true#M167984