[SR-Users] Webrtc: Don't catch 488 between JSSIP and SIP UA's

[LAA]

Apologize. Previous message was too long.
L.


El 02/06/2014 20:25, "LAA" <ornitorrinco7424 at gmail.com> escribió:

> Hi all,
>
> Another guy strugling his mind trying to get a configuration to enable
> calls between WebRTC UA (JSSIP) to standard SIP UA (Twinkle or SjPhone)
> I've been working with the examples that were shared by Carlos Ruiz Diaz
> and Peter Dunkley (thanks to both).
>
> http://www.slideshare.net/crocodilertc/webrtc-websockets
> http://caruizdiaz.com/2014/02/26/webrtc-kamailio/
>
> Kamailio is not running behind NAT. I'm using rtpproxy-ng module with
> Kamailio 4.1.3, and Rtpengine.
>
> I share a link with my current configuration, wich is based in Peters
> example, with websocket support from websocket.cfg example.
>
> -  Calls between SIP standard UA's are working OK. I have some endpoint
> behind nat.
> -  Calls between JSIP UA's are working OK. So, websocket support is
> running.
> -  Calls from JSIP and Twinkle are NOT WORKING OK. sip UA send's back a
> 488 response, and Kamailio send it back to JSSIP (Incompatible SDP).
> -  Calls from Twinkle to Jsip are NOT WORKING OK: Kamailio sends an INVITE
> to JSIP, and it returns an error. And Kamailio sends 488 to Twinkle.
>
>
> It seems as if Kamailio is not catching 488. I share a snippet of my
> config, and links to tcpdump captures:
>
> https://www.dropbox.com/s/i7c9ty57oauujc4/fromws0.pcap
> https://www.dropbox.com/s/q3q30pgzvdoswts/kamailio.cfg
> https://www.dropbox.com/s/rqtjwcbgg1foaoq/tows0.pcap
>
> What am I missing?
>
>
> Best regards.
>
> Luis.
>
>
>
>
>
>
>
>
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