[SR-Users] WebRTC to PSTN call, proxied through Kamailio

[Ben Langfeld]

Maybe you could include you config also?

On 10 February 2015 at 15:01, Rahul MathuR <rahul.ultimate at gmail.com> wrote:

> Hello gents,
>
> I was trying my hands on getting a successful RTCweb call (JSsip, since
> Peter Dunkley mentioned that he's been using JSsip for most of the testing
> scenarios..) to PSTN, making my kamailio as proxy + protocol converter (sip
> over web-sockets to sip over udp).
> And yes, I've referred Carlos' config; the main problem is I get 'Bad
> Media Description' error in Google Chromium (Version 40.0.2214.111 m) &
> my SIP server even sends 200 OK, but my phone doesn't ring. To make it
> worse, I can see rtpengine throwing this error -
> "SRTCP output wanted, but no crypto suite was negotiated"
>
> BTW, I have -
> [root at localhost log]# openssl version
> OpenSSL 1.0.1j 15 Oct 2014
>
> I even tried building kamailio & rtpengine using this openssl but in-vain.
> One thing that baffles me is that, apparently kamailio has started
> receiving RTP packets (perhaps early media) but the mobile phone hasn't
> ringed :-(
>
> I am attaching all possible logs & seek some guidance from the array of
> experts in this list.
>
> Files attached:
> a) tcpdump on ext. interface
> b) tcpdump on loopback
> c) syslogs
> d) Chromium JS logs
>
> UAC (14.98.55.38), Kamailio (125.99.186.126), SIP Server
> (157.238.178.153), Media Server (199.27.244.6)
>
>
>
> --
> Warm Regds.
> MathuRahul
>
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