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

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.