Gentle Reminder ! Thanks
Warm Regds, Rahul On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR <[email protected]> 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 <[email protected]> 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 <[email protected]> >> 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 >>> [email protected] >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> [email protected] >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > -- > Warm Regds. > MathuRahul > -- Warm Regds. MathuRahul
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [email protected] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
