[asterisk-users] Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2

Robert McGilvray rmcgilvr at sscinc.com
Thu Mar 17 14:01:33 CDT 2016


I did some more troubleshooting eliminating G722 just in case there was an issue with transcoding / MTP which has resulted in a slightly different SDP but resume still doesn't work.

Full sip dialog: https://drive.google.com/open?id=0B6XOeEMvID0vZVdOV2NzM3NJZGM

Initial call setup appears to be correct. CUCM sends an early offer to Asterisk with SDP, Asterisk responds with a 200 OK with SDP and a=sendrecv.

When I place the call on HOLD the CUCM sends a delayed offer INVITE to Asterisk:

Asterisk responds with a 200 OK containing the SDP with a=sendrecv
CUCM ACKS with an SDP containing a=sendonly

When I resume the call the CUCM sends a delayed offer INVITE to Asterisk:

                Asterisk responds with a 200 OK containing the SDP with a=recvonly
                CUCM ACKS with an SDP containing a=sendonly

I may be missing or interpreting something incorrectly but that does not right for a RESUME scenario.

Per RFC32645 the CUCM is responding in one of the two ways permitted:


"If a stream is offered as sendonly, the corresponding stream MUST be
   marked as recvonly or inactive in the answer.  If a media stream is
   listed as recvonly in the offer, the answer MUST be marked as
   sendonly or inactive in the answer.
"

Is this a bug or am I wrong in my interpretation of the dialog?

Thanks!

Robert McGilvray
o: 914 293 3584

From: Robert McGilvray
Sent: Thursday, March 17, 2016 12:55 PM
To: 'asterisk-users at lists.digium.com'
Subject: Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2

Hello,

We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe and chan_sip for conferences. I have been testing the new versions of Asterisk with PJSIP and ConfBridge but have run into an issue which is preventing us from moving forward. Everything works fine until a call is placed on hold, after resuming the call the user cannot hear audio from the bridge. The same thing works perfectly with 1.8.20.1.

The scenario is:                 Cisco 8845 SIP (G722) -> CUCM 8.6.2 (also tested 10.5, same issue) - > SIP trunk to Asterisk 13.7.2 PJSIP with delayed offer ->  ConfBridge.

Tcpdump reveals Asterisk is sending the RTP to the endpoint so I suspect we're dealing with a bug / interop issue with the culprit possibly being a=inactive lines in the SDP.

I've included a link (on drive) to two separate SIP traces, one using ngrep and the other is the output of pjsip logging and the relevant sections of my pjsip.conf

https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM&usp=sharing

Can anyone offer some insight?

Regards,

BobM
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