Summary:	ASTERISK-27795: chan_sip: one way / no audio with srtp
Reporter:	Florian Kaiser (fmkaiser)	Labels:	pjsip
Date Opened:	2018-04-08 05:36:50	Date Closed:	2018-05-03 10:30:31
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/SRTP
Versions:	15.3.0	Frequency of Occurrence	Constant
Related Issues:	is related to ASTERISK-27604 chan_sip: turning on srtp results in one way audio in Asterisk 15 is related to ASTERISK-27800 One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP
Environment:		Attachments:
Description:	Same behavior as described in ASTERISK-27604: One-way audio between srtp and non-srtp clients. No audio between two srtp clients. No problem with just one client (e.g. echotest). Analysis of tcpdump (available on request) reveals that for the srtp stream Asterisk -> client, the wrong - local - encryption key is being used. That explains the behaviour: Client can't decode incoming stream -> no audio. Git bisect narrows it down to commit 065c300 / ASTERISK-27118.
Comments:	By: Asterisk Team (asteriskteam) 2018-04-08 05:36:51.959-0500 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: Asterisk Team (asteriskteam) 2018-04-08 05:36:52.751-0500 The module you are reporting the issue against is no longer supported as a core module but 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 By: Kevin Harwell (kharwell) 2018-04-09 15:19:43.339-0500 I've somewhat confirmed this with a little bit different results. With Asterisk 15 running at commit just prior to 065c300 I saw the following: srtp->dialpan (playback) - heard audio srtp->non srtp endpoint - two way audio srtp->srtp endpoint - two way audio However, once I updated to commit 065c300 I then saw the below: srtp->dialpan (playback) - heard audio srtp->non srtp endpoint - two way audio srtp->srtp endpoint - one way audio So there does appear to be a problem. By: Kevin Harwell (kharwell) 2018-04-09 15:38:01.395-0500 Also tested with chan_pjsip and there does not appear to be any problem when using that channel driver. By: Addix Internet Services GmbH (addix) 2018-04-19 03:37:38.478-0500 Can reproduce this issue here also. Thats a big show-stopper for us in the transistion to asterisk 15 as we liked the idea of having both sip stacks to make a slow transition of the users between the sip stacks. By: Addix Internet Services GmbH (addix) 2018-05-03 01:20:00.658-0500 Thanks, patch from gerrit 8886 solved that for us. By: Friendly Automation (friendly-automation) 2018-05-03 10:30:32.484-0500 Change 8886 merged by Jenkins2: res_rtp_asterisk: Always update SRTP on local SSRC change. [https://gerrit.asterisk.org/8886|https://gerrit.asterisk.org/8886] By: Friendly Automation (friendly-automation) 2018-05-03 10:33:30.132-0500 Change 8885 merged by Jenkins2: res_rtp_asterisk: Always update SRTP on local SSRC change. [https://gerrit.asterisk.org/8885|https://gerrit.asterisk.org/8885]