[asterisk-bugs] [JIRA] (ASTERISK-29699) ConfBridge doesn't detect disconnect from webrtc

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29699?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=256660#comment-256660 ] 

Joshua C. Colp commented on ASTERISK-29699:
-------------------------------------------

There is no option currently to terminate channels if an underlying TCP/TLS/WSS connection is lost. Session timers send a periodic SIP request to check that the SIP signaling is alive, if it's not then the channel is hung up. I don't understand why RTP timeout doesn't work for you - it hangs up the channel (the same as if there were an option to hang up the channel if the connection was lost) when media stops flowing for a period of time. It sounds like you're expecting some kind of event or notification that a channel has "died". That doesn't exist currently for anything. All options just hang the channel up.

I'd suggest raising this on the community forum instead[1] as this really seems to be centered around your usage and the options available.

[1] https://community.asterisk.org/

> ConfBridge doesn't detect disconnect from webrtc
> ------------------------------------------------
>
>                 Key: ASTERISK-29699
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29699
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Core/Bridging
>    Affects Versions: 18.7.1
>         Environment: Linux asteriskcloudpjsip 4.19.0-16-amd64 #1 SMP Debian 4.19.181-1 (2021-03-19) x86_64 GNU/Linux
>            Reporter: Dennis Haney
>              Labels: webrtc
>
> It seems that ConfBridge doesn't detect when the user has crashed/disconnected or otherwise closed the wss connection.
>   == WebSocket connection from 'me:1247' closed
>     -- Removed contact 'sips:mylogin at me:1247;transport=ws;rtcweb-breaker=no' from AOR 'mylogin at someuniquevalueforyoumachine.mydomain.com' due to shutdown
>   == Contact mylogin at someuniquevalueforyoumachine.mydomain.com/sips:mylogin at me:1247;transport=ws;rtcweb-breaker=no has been deleted
> Checking state of the confbridge user was in at the time:
> asteriskcloudpjsip*CLI> confbridge list 6f22d552eec54391951519ef4b9665bf
> Channel                        Flags  User Profile     Bridge Profile   Menu             CallerID
> ============================== ====== ================ ================ ================ ================
> PJSIP/mylogin.someuniquevalueforyoumachine.mydomain.com-00000001 M      interviewer      default_bridge   default_menu     <unknown>
> asteriskcloudpjsip*CLI> pjsip show channels
>   Channel:  <ChannelId........................................>  <State.....>  <Time.....>
>       Exten: <DialedExten.............>  CLCID: <ConnectedLineCID.......>
> ==========================================================================================
>   Channel: PJSIP/mylogin.someuniquevalueforyoumachine.mydomain Up            00:14:28
>       Exten: 1                           CLCID: "someuniquevalueforyoumachine.mydomain.com" <>
> This never gets cleaned up.
> asteriskcloudpjsip*CLI> pjsip show channelstats like my
>                                              ...........Receive......... .........Transmit..........
>  BridgeId ChannelId ........ UpTime.. Codec.   Count    Lost Pct  Jitter   Count    Lost Pct  Jitter RTT....
>  ===========================================================================================================
>  f3bd17e0 mylogin.someuni 00:14:57 opus      140       0    0   0.000  44841       0    0   0.003   0.000
> Objects found: 1
> Furthermore, Transmit never detects it is sending packets into the void and marks them as dropped.



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