Hi!
We recently started having issues with no audio for external inbound and outbound calls.
A heads up I'm not that experienced with troubleshooting these kinds of issues so sorry if I don't understand how things work fully. I'm trying my luck here before we contact ask TAC.
Before this no audio issue we had an issue that some calls were dropped after 15 minutes. We "hopefully" did a workaround for that problem(no reports of that issue yet) when we sat the cube sessions expire timer to max. Couple of days later we are getting reports of no audio on calls.
We got an idea that the sessions are hanging due to the timer and that's why we don't get audio.
Call flow SIP phone <--> CUCM <--> CUBE <--> ITSP
Going to add a SIP trace from RTMT as an attachement soon.
This case the person tested it on his own phone. First couple of calls everything worked as it should. After the 3th or 4th call no audio.
Here is the first invite from SIP phone to CUCM. Calling 20224 Called 092888888
SENDER: [SEPCC70ED350FC4] 10.140.7.2
GUID: cc70ed35-0fc4001e-7c320351-5a43ad0e@10.140.7.2
MSG_LABEL: INVITE
RECEIVER: 10.20.200.3
MAC_ADDRESS: SEPCC70ED350FC4
MSGTAG: 124316808
MSG_TYPE: UCM_CTRACE
CORRELATIONID: 4,100,14,159790.4980^10.140.7.2^*
TIMESTAMP: 2021/01/13 14:47:44.015
Detailed Sip Message
INVITE sip:092888888@SS-20-0004;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.140.7.2:50141;branch=z9hG4bK675f8940
From: " Name Of Caller" < sip:20224@SS-20-0004> ;tag=cc70ed350fc4121f26009557-77720333
To: < sip:092888888@SS-20-0004>
Call-ID: cc70ed35-0fc4001e-7c320351-5a43ad0e@10.140.7.2
Max-Forwards: 70
Session-ID: 03c3922100105000a000cc70ed350fc4;remote=00000000000000000000000000000000
Date: Wed, 13 Jan 2021 13:47:38 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8851/12.8.1
Contact: < sip:34834717-8812-c239-40d0-e5b396173679@10.140.7.2:50141;transport=tcp> ;+u.sip!devicename.ccm.cisco.com=" SEPCC70ED350FC4"
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: " Name of caller" < sip:20224@SS-20-0004> ;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 711
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 8691 0 IN IP4 10.140.7.2
s=SIP Call
b=AS:4064
t=0 0
m=audio 25096 RTP/AVP 114 9 124 113 115 0 8 116 18 101
c=IN IP4 10.140.7.2
b=TIAS:64000
a=rtpmap:114 opus/48000/2
a=fmtp:114 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=64000;stereo=0;sprop-stereo=0;usedtx=0
a=rtpmap:9 G722/8000
a=rtpmap:124 ISAC/16000
a=rtpmap:113 AMR-WB/16000
a=fmtp:113 octet-align=0;mode-change-capability=2
a=rtpmap:115 AMR-WB/16000
a=fmtp:115 octet-align=1;mode-change-capability=2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
Appreciate any help I can get!
Thank you.
Earl
