Apologize. Previous message was too long. L.
El 02/06/2014 20:25, "LAA" <ornitorrinco7...@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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