Summary:	ASTERISK-29963: res_rtp_asterisk: mapping->ssrc_invalid on unidirectional videostream after confbridge reinvite
Reporter:	Erik Bergschöld (Bergschold)	Labels:	webrtc
Date Opened:	2022-03-10 04:33:57.000-0600	Date Closed:
Priority:	Minor	Regression?
Status:	Open/New	Components:	Resources/res_rtp_asterisk
Versions:	18.8.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	Alma Linux 8, Chrome WebRTC with JsSIP	Attachments:	( 0) sip.txt
Description:	When a client joins a Confbridge and there is another participant in the conference already then the client get an INVITE from asterisk looking like this (see attached file) Asterisk will send the other participants video stream on the second video with ssrc=12683906 and everything is working with video and voice coming from and to both participants. The problem is when the client loses a packet and sends a NACK request to Asterisk, then asterisk is unable to map that rtcp instance to the sendonly stream since the mapping is marked as ssrc_valid=0. Asterisk will then fail to map to the correct instance and try to act on the voice instance instead and write res_rtp_asterisk.c:6551 ast_rtcp_interpret: (0x7ff0c8083bc0) RTCP before handle NACK request, retransmissions are not enabled ignore this message! I can see by adding some extra logging in asterisk that the function ast_rtp_bundle sets the mapping.ssrc_valid=child_rtp->themssrc_valid wich in this case is 0 right after asterisk processes the sdp. It's possible to get retransmissions to work by changing this line in __rtp_find_instance_by_ssrc if (mapping->ssrc_valid && mapping_ssrc == ssrc) { to if (mapping_ssrc == ssrc) {
Comments:	By: Asterisk Team (asteriskteam) 2022-03-10 04:33:58.150-0600 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. 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: Joshua C. Colp (jcolp) 2022-03-10 05:14:03.574-0600 I attached the SIP trace, there is no need to embed it in the description.