Summary:	ASTERISK-29122: Unable pjsip direct_media between endpoint for vp8 codec
Reporter:	Denis Islamov (diislamov)	Labels:
Date Opened:	2020-10-11 08:00:57	Date Closed:	2020-10-14 04:34:49
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	pjproject/pjsip
Versions:	16.13.0	Frequency of Occurrence
Related Issues:
Environment:	debian 10.03, asterisk 16.13.0	Attachments:	( 0) sngrep_screen.png
Description:	When you turn on the video in the rtp stream, traffic starts to pass through Asterisk. Without video, the stream goes directly. [101] type=endpoint aors=101 auth=101-auth disallow=all allow=ulaw,alaw,vp8 context=from-test callerid=101 <101> transport=transport-udp dtmf_mode=inband direct_media=yes max_audio_streams=10 max_video_streams=10 rtp_symmetric=yes rewrite_contact=yes force_rport=yes language=ru [102] type=endpoint aors=102 auth=102-auth disallow=all allow=ulaw,alaw,vp8 context=from-test callerid=102 <102> transport=transport-udp dtmf_mode=inband direct_media=yes max_audio_streams=10 max_video_streams=10 rtp_symmetric=yes rewrite_contact=yes force_rport=yes language=ru
Comments:	By: Asterisk Team (asteriskteam) 2020-10-11 08:00:58.866-0500 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Asterisk Team (asteriskteam) 2020-10-11 08:00:59.481-0500 We appreciate the difficulties you are facing, however information request type issues 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. If this issue is actually a bug please use the Bug issue type instead. 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 By: Denis Islamov (diislamov) 2020-10-11 08:02:48.817-0500 192.168.88.19 - Asterisk 192.168.88.14 - 101 peer 192.168.88.20 - 102 peer By: Asterisk Team (asteriskteam) 2020-10-11 08:16:28.038-0500 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) 2020-10-11 12:29:46.507-0500 Direct media is not supported for video streams in PJSIP. If video is present, then it is disabled. By: Denis Islamov (diislamov) 2020-10-13 12:23:02.695-0500 What other ways can I establish a direct video connection between an endpoint? By: Asterisk Team (asteriskteam) 2020-10-13 12:23:02.825-0500 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) 2020-10-13 12:35:16.371-0500 It would need to be done without any involvement of Asterisk, or such support would need to be added to Asterisk. By: Denis Islamov (diislamov) 2020-10-13 22:54:11.630-0500 Are there add-ons for Asterisk that allow you to add this functionality? Is it possible to use other solutions, for example, openSips for rtp traffic management? By: Asterisk Team (asteriskteam) 2020-10-13 22:54:12.048-0500 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) 2020-10-14 04:34:49.823-0500 I have no experience with that so can't really comment. You may have better luck using the forum at https://community.asterisk.org/ to interact with people who may have deployed such a thing.