wrote:

Hello, there were some config snippets on mailing lists (maybe sr-dev) about this topic. Peter just published his slides at last event he presented on this topic: - http://www.slideshare.net/crocodilertc/webrtc-websockets Last part has also snippets of Kamailio config. Cheers, Daniel On 09/12/13 14:04, Muhammad Shahzad wrote: Hi, According to documentation, using kamailio's rtpproxy-ng module with mediaproxy-ng service, it is possible to make webrtc to sip calls and vice versa, However i am stuck since morning to make JSSIP (in chrome) to phonerlite (in Windows 8) calls. There is not working example or sample code anywhere either. So i was wondering if anyone has actually tries that successfully and would care to share some samples for us. So, far i tried "+SP" flags for phonerlite to JSSIP calls and "-sp" for JSSIP to phonerlite calls in "rtpproxy_manage" method. Apparently both calls connects but then drop after a few seconds of ACK. Which indicate the problem is likely to be on mediaproxy-ng end rather then kamailio.. Thank you. -- Mit freundlichen Grüßen Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users