[SR-Users] No audio/video transmission over different networks

Abhishek Saini abhishek.saini at enukesoftware.com
Mon Sep 15 12:58:48 CEST 2014


I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng
package on my ubuntu box. As suggested here:

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
Via: SIP/2.0/WS
From: "admin" <sip:admin at abc.com>;tag=bzhwwG8nT2gFwwJgIyrz.
To: <sip:hari at 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.

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