14 Feb
2015
14 Feb
'15
7:50 a.m.
Thank you Marc.
On Thu, Feb 12, 2015 at 11:51 PM, Marc Soda <msoda(a)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(a)gmail.com>
wrote:
--
Warm Regds.
MathuRahul
Gentle Reminder !
Thanks
Warm Regds,
Rahul
On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR <rahul.ultimate(a)gmail.com>
wrote:
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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(a)coredial.com> wrote:
--
Warm Regds.
MathuRahul
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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(a)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@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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>
>
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--
Warm Regds.
MathuRahul