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

[Dennis Haney (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29699?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=256658#comment-256658 ] 

Dennis Haney commented on ASTERISK-29699:
-----------------------------------------

Thanks for your prompt reply.

Where is the option to do so when the wss tcp connection is lost?

"Timers" settings in pjsip seems to be mostly undocumented? The only mention I can find is
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip#Asterisk18Configuration_res_pjsip-endpoint_timers
What do they actually do? I don't want to just randomly hangup in the middle on someones conference call because they "talked to long".

Reconnecting with max_contacts=1 also doesn't invalidate former open channels.

rtp_timeout is still pretty useless, because I cannot figure out a way to detect if the conference contains an old dead channel, or I need to reinvite the user again.
Even if i DO reinvite the user again to the same channel, there is now 2 users of the same name and I get an AMI ConfbridgeLeave/Hangup despite still being in the conference with a live connection.
MusicOnHold also doesn't play when the same user joins twice.

> 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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