[asterisk-bugs] [JIRA] (ASTERISK-30071) rtp: Usage of rtp_timeout on WebRTC causes failure

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30071?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259209#comment-259209 ] 

Joshua C. Colp commented on ASTERISK-30071:
-------------------------------------------

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> rtp: Usage of rtp_timeout on WebRTC causes failure
> --------------------------------------------------
>
>                 Key: ASTERISK-30071
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30071
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp, Resources/res_rtp_asterisk
>    Affects Versions: 18.12.0
>            Reporter: nappsoft
>              Labels: webrtc
>
> We recently migrated from asterisk 16.19.0 (with security patches) to asterisk 18.12.0 and PJSIP 2.12.
> We now have a problem with connections to ConfBridges over websockets as soon as the the user connects with audio and video. If the user only connects with audio or if we set rtp_timeout to 0, everything works as expected.
> We didn't have this behavior with asterisk 16.19.0.



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