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

[Rahul MathuR]

Thank you Marc.

On Thu, Feb 12, 2015 at 11:51 PM, Marc Soda <msoda at coredial.com> wrote:

> Our config is based on the example config and the WebRTC bits are based on Carlos'.
>  I've attached the relevant parts.  It's pretty heavily customized to our
> specific environment.  The main differences are the way that we detect a
> video call, how we route to our backend servers, and that we send video
> calls directly to a registered peer and not the the backend Asterisk
> servers.
>
> On Thu, Feb 12, 2015 at 12:34 PM, Rahul MathuR <rahul.ultimate at gmail.com>
> wrote:
>
>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>>
>>> --
>>> Warm Regds.
>>> MathuRahul
>>>
>>
>>
>>
>> --
>> Warm Regds.
>> MathuRahul
>>
>> _______________________________________________
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>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
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>
>


-- 
Warm Regds.
MathuRahul
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