Summary:	ASTERISK-28731: Directmedia Reinvites have SDP with codecs from configuration not negotiation
Reporter:	Christian Berger (Christian_Berger)	Labels:
Date Opened:	2020-02-10 05:28:33.000-0600	Date Closed:
Priority:	Major	Regression?
Status:	Open/New	Components:	Channels/chan_sip/CodecHandling
Versions:	13.31.0	Frequency of Occurrence	Constant
Related Issues:
Environment:		Attachments:	( 0) ast_a_config.tar.gz ( 1) ast_b_config.tar.gz ( 2) ast_c_config.tar.gz ( 3) clean.pcapng
Description:	When Asterisk sends out Re-INVITES to 2 connected parties in order to get out of the media stream, the SDP it offers does not reflect the negotiated SDP, but the one configured in sip.conf I have written an overview about the issue here: https://community.asterisk.org/t/chan-sip-direct-rtp-codec-negotiation-problem/82425 In short the situation is like this: I have made a minimalistic setup in order to examine an unreleated bug. In this setup I have 3 Asterisk instances (A,B and C) running on 3 separate computers. Asterisk A is configured to use RTP-Events, Asterisk C is configured to only use Inband DTMF. Asterisk B is configured to accept both and to Re-Invite both parties in order to get out of the media stream when it is possible. When a call comes from A to B, B plays a message and forwards it to C. The A-B leg will include RTP-Events, the B-C leg will not include RTP-Events. However shortly after this, Asterisk Re-Invites A and C to get out of the media stream. In these Re-Invites it will offer RTP-Events to both sides, making A believe that C can accept RTP-Events when it fact cannot do that.
Comments:	By: Asterisk Team (asteriskteam) 2020-02-10 05:28:33.819-0600 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: Christian Berger (Christian_Berger) 2020-02-10 05:40:32.969-0600 This shows the Setup. Asterisk A has the IP 192.168.11.65 Asterisk B has the IP 192.168.11.68 Asterisk C has the IP 192.168.11.64 By: George Joseph (gjoseph) 2020-02-10 08:31:07.860-0600 The chan_sip channel driver is in 'extended' support status and is supported only by community members.  Your issue is in the queue. Your patience is appreciated as a community developer may work the issue when time and resources become available. Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1] If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way. [1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2]: http://www.asterisk.org/community/discuss [3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties