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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30071?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259386#comment-259386 ] 

Asterisk Team commented on ASTERISK-30071:
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Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> 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
>            Assignee: 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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