Re: [SR-Users] Kamailio + mediaproxy-ng for webrtc to sip calls

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 <mailto:shari_786pk@hotmail.com> Email: shaheryarkh(a)googlemail.com <mailto:shaheryarkh@googlemail.com> _______________________________________________ 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