11 Feb
2015
11 Feb
'15
5:28 p.m.
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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