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

[Muhammad Shahzad]

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 at hotmail.com
Email: shaheryarkh at googlemail.com
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