Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" sip:admin@abc.com;tag=bzhwwG8nT2gFwwJgIyrz. To: sip:hari@abc.com;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek