Summary:	ASTERISK-26858: chan_sip: SDP does not contain declined video stream in session timers
Reporter:	Dragomir Haralambiev (goup)	Labels:
Date Opened:	2017-03-12 13:34:10	Date Closed:
Priority:	Major	Regression?
Status:	Open/New	Components:	Channels/chan_sip/WebSocket
Versions:	14.3.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	CentOS 7.3.1611,	Attachments:	( 0) http.conf ( 1) sip.conf ( 2) tryit.jssip.net.log
Description:	Hello, I try to test tryit.jssip.net with Asterisk 14.3.0. 1060 (WebRTC) call 1061 (SIP) using chan_sip. Call is connected, after one min sound is stopped in asterisk log is appear: Got SIP response 500 "JsSIP Internal Error" back from 192.168.1.104:5060 If using SIPML5 all forking fine. What can I do to solve this problem? Here full asterisk log: == DTLS ECDH initialized (automatic), faster PFS enabled == Using SIP RTP CoS mark 5 -- Executing [1061@webrtc:1] Dial("SIP/1060-00000019", "SIP/1061") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1061 -- SIP/1061-0000001a is ringing > 0x7fbf14009110 -- Probation passed - setting RTP source address to 192.168.1.152:55334 -- SIP/1061-0000001a answered SIP/1060-00000019 -- Channel SIP/1061-0000001a joined 'simple_bridge' basic-bridge -- Channel SIP/1060-00000019 joined 'simple_bridge' basic-bridge > 0x7fbf14009110 -- Probation passed - setting RTP source address to 192.168.1.152:55334 > 0x7fbec403a9e0 -- Probation passed - setting RTP source address to 192.168.1.104:53248 -- Got SIP response 500 "JsSIP Internal Error" back from 192.168.1.104:5060 > 0x7fbec403a9e0 -- Probation passed - setting RTP source address to 192.168.1.104:53248 [Mar 10 21:08:11] WARNING[4801]: netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
Comments:	By: Asterisk Team (asteriskteam) 2017-03-12 13:34:11.205-0500 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: Dragomir Haralambiev (goup) 2017-03-12 13:39:21.897-0500 Here is attached Asterisk sip.conf and http.conf. tryit.jssip.net.log is Google Chrome log. By: Joshua C. Colp (jcolp) 2017-03-12 14:01:18.247-0500 We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum. The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors. Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines If after seeking support this turns out to be an issue in Asterisk itself this issue can be reopened. By: Dragomir Haralambiev (goup) 2017-03-12 17:25:18.816-0500 Thanks for your quick reply. When I report to JsSip (https://groups.google.com/forum/#!topic/jssip/MdO0-Y7-6fY) I receive follow answer: ==== The re-INVITE send by Asterisk (SIP re-invite Session-Timers) is wrong because it lacks the m=video line that was previously negotiated. This is a clear bug in Asterisk because it violates SDP rules on renegotiation. You should report it to Asterisk. ===== JsSip recommends referring to Asterisk , Asterisk recommends asking JsSip and problem is not resolved. Maybe it is good idea to migrate from Asterisk to Freeswitch or Opensips? By: Asterisk Team (asteriskteam) 2017-03-12 17:25:19.182-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Joshua C. Colp (jcolp) 2017-03-12 17:51:07.316-0500 The chan_sip module is community supported so there is no guarantee on when the underlying problem will be resolved, I have accepted the issue though.