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

[Rahul MathuR]

Thanks guys !

I did further investigation of the Chrome logs and found that... (this is
really interesting), even though I disabled Video; still JSsip was sending
video information in the m & a lines.
The fact that I was trying to call PSTN number made it mandatory to set
video port to '0' in 183 and 200. However, JSsip was not happy with that
and cribbed about codec-formats not being present, ergo "Bad Media
Description".

Marc,
Could you please share your config so that I'd be sure my kamailio &
rtpengine side is in proper shape.


P.S. I am attaching mine here.

On Wed, Feb 11, 2015 at 8:58 PM, Marc Soda <msoda at coredial.com> wrote:

> We are in the middle of designing a similar solution with Kamailio and
> rtpengine and after some initial problems things are going really well.  I
> can tell you that we ended up going with SIPjs over JSSip and it handled a
> lot of the weird browser specific issues we were having.
>
> I'm not sure about the media description error, however, the crypto error
> is probably not a real issue.  Richard explained it here:
>
> http://lists.sip-router.org/pipermail/sr-users/2014-December/086271.html
>
> I corrected the other issues I was having and that one seemed to resolve
> itself.
>
> Hope that helps,
> Marc
>
> On Tue, Feb 10, 2015 at 12:01 PM, 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
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
>
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> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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>
>


-- 
Warm Regds.
MathuRahul
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