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

[Rahul MathuR]

Gentle Reminder !

Thanks

Warm Regds,
Rahul

On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR <rahul.ultimate at gmail.com>
wrote:

> 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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>> sr-users at lists.sip-router.org
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>>
>>
>
>
> --
> Warm Regds.
> MathuRahul
>



-- 
Warm Regds.
MathuRahul
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