Hello,

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:
- https://github.com/caruizdiaz/kamailio-ws

Cheers,
Daniel

On 15/09/14 12:58, Abhishek Saini wrote:

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

-- 
Daniel-Constantin Mierla
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