Summary:	ASTERISK-28047: chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs
Reporter:	Will (drizuid)	Labels:	pjsip
Date Opened:	2018-09-11 09:16:22	Date Closed:	2018-09-19 08:42:58
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Resources/res_pjsip_session
Versions:	15.6.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	debian 9.5, NAF's gvsip fork with oauth	Attachments:	( 0) extensions.conf ( 1) gvsip_trace1.txt ( 2) gvsip_trace2_with_codecs_disabled.txt ( 3) gvsip_trace3_max_video_streams=0.txt ( 4) pjsip.conf ( 5) rtp.conf
Description:	@jcolp sent me here to post this from the dslreports forums. When placing a call from a cisco phone registered to a Cisco Unified Communications Manager (11.5) across a sip trunk to an asterisk server out a gvsip trunk, the call fails after a reinvite contains SDP with extraneous information. The endpoint is setup to allow nothing but g711ulaw, yet without explicitly max_video_streams=0 on the endpoint, SDP contains the m=video and offers h264. This causes the call to drop after ~30seconds.
Comments:	By: Asterisk Team (asteriskteam) 2018-09-11 09:16:24.275-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Will (drizuid) 2018-09-11 09:17:24.910-0500 Calling number is listed as <my 10 digit CALLING #> or <my e164 CALLING #> (this is a Google voice number) Called number is listed as <my 10 digit cellphone>, <my 11 digit cellphone>, or <my e164 cellphone> 192.168.128.134 is the IP of my CALLING phone 192.168.128.12 is a Cisco Unified Communications Manager. It has a sip trunk to Asterisk. My CALLING phone is on the CUCM. 192.168.128.7 is the Asterisk PBX   Call flow: Phone(.134) -> CUCM(.12) -> Asterisk(.7) -> GVSIP -> PSTN -> Cellphone The call will attempt to route via sipbroker's e164 before rerouting to GVSIP. By: Joshua C. Colp (jcolp) 2018-09-18 06:01:22.337-0500 The problem is that if a session refresh is requested with a declined/removed video stream we will add a declined/removed video stream to the re-invite. The logic which doesn't do this if no compatible format exists comes after. By: Friendly Automation (friendly-automation) 2018-09-19 08:43:00.328-0500 Change 10173 merged by Joshua Colp: res_pjsip_session: Don't add declined stream if one does not exist. [https://gerrit.asterisk.org/10173|https://gerrit.asterisk.org/10173] By: Friendly Automation (friendly-automation) 2018-09-19 08:43:10.905-0500 Change 10172 merged by Joshua Colp: res_pjsip_session: Don't add declined stream if one does not exist. [https://gerrit.asterisk.org/10172|https://gerrit.asterisk.org/10172] By: Friendly Automation (friendly-automation) 2018-09-19 08:43:19.781-0500 Change 10174 merged by Joshua Colp: res_pjsip_session: Don't add declined stream if one does not exist. [https://gerrit.asterisk.org/10174|https://gerrit.asterisk.org/10174]