[SR-Users] Kamailio sends SIP messages to old websocket after reconnection during call
Daniel-Constantin Mierla
miconda at gmail.com
Fri May 18 14:55:47 CEST 2018
Hello,
are you doing lookup(location) for the BYE that is attempted to be
routed on the old connection?
Cheers,
Daniel
On 17.05.18 17:45, Idris AVCI wrote:
> Hi,
>
> We use kamailio as a backend for JSSIP based WebRTC UAs.
> We use SIP over Weboscket as transport and everythings work fine
> generally.
>
> Some websocket connections to kamailio drop during an active call and
> JSSIP reconnects immediately. New websocket connection has a different
> port than the dropped one but kamailio tries to send to old websocket
> port which is not active anymore.
>
> We use GRUU and configured all the related parameters in registrar,
> dialog, tm, websocket, tls, xhttp and userloc modules both for
> versions 4.4 and 5.1 but couldn't find a solution yet.
>
> For a typical call below are the ordered events:
>
> 1. UA register to kamailio over wss (Code: WS1)
> 2. Kamailio send an INVITE to the UA (Code: Call1)
> 3. UA answer and media starts
> 4. UA drops WS1 to kamailio and reconnect (Code: WS2)
> 5. UA has an active websocket (WS2) to kamailio
> 6. Kamailio tries to send BYE message of Call1 to WS1 which is no
> longer active and complains that websocket could not be found
> 7. Kamailio send an INVTE to the UA (Code: Call2) over WS2
> 8. UA answer and media starts
> 9. Kamailio send BYE message of Call2 to UA over WS2 and everything
> work as expected (No websocket drop)
>
> Error log:
>
> WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS
> connection (id: 0) for WebSocket could not be found
>
>
> I think we need a mechanism that will update the dialog information of
> Call1 to use WS2 after reconnection.
>
> Any help greatly appreciated.
>
> Thanks.
>
>
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--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
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