[SR-Users] No audio/video transmission over different networks
miconda at gmail.com
Mon Sep 15 14:49:15 CEST 2014
the reply code indicates that the media type is not supported, thus
there has been no gatewaying between webrtc and classic rtp. Just
replacing rtpproxy with rtpengine is not enough, there are different
parameters that have to be provided.
Searching on web, I see that Carlos has published a config for it, see:
On 15/09/14 12:58, Abhishek Saini wrote:
> 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
> <mailto:sip%3Aadmin at abc.com>>;tag=bzhwwG8nT2gFwwJgIyrz.
> To: <sip:hari at abc.com <mailto:sip%3Ahari 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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