Maybe you could include you config also? On 10 February 2015 at 15:01, Rahul MathuR <rahul.ultim...@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 > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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