Any thoughts on this gents ?
On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate <rahul.ultim...@gmail.com> wrote: > Kamailio is just acting as a proxy and protocol modifier so to say. It is > workin with rtpengine from sipwise to handle media as evident from he logs. > This architectue uses a TURN server and the browser is chrome with > latest updates. > > The only thing whih I haven't done is enable TLS in kamailio and create > certs. (which I'm not completely sure how to do).. > Also, does it necessitates to have Apache ruuning https on 443 ? > > Thanks in advance > > > Sent from Samsung Mobile > > > -------- Original message -------- > From: Gonzalo Gasca Meza > Date:27/01/2015 4:07 AM (GMT+05:30) > To: "Kamailio (SER) - Users Mailing List" > Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy > > Are you terminating media in Kamailio or just handling WS communication? > If yes which version of Kamailio and rtp-proxy ? > Have you tried passing media directly between Browser and Kamailio with > any TURN server? > > Are you using latest Chrome version or FF ? > > A working sample config using the following architecture: > > https://github.com/spicyramen/llamato/tree/LlamatoReg > > signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip > udp--> * > media: sipml5 > ------------------------------------------------------------------------> * > > > > > On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR <rahul.ultim...@gmail.com> > wrote: > >> Hi Richard, >> >> Thanks for spending some cycles on it. >> >> It is OpenSSL 1.0.1e-fips 11 Feb 2013 >> >> On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs <rfu...@sipwise.com> >> wrote: >> >>> On 26/01/15 02:21 PM, Rahul MathuR wrote: >>> >>>> Hello, >>>> >>>> I am totally struck at a point while implementing Kamailio as proxy for >>>> WebRTC enabled UAC (Jssip). I am using Google's TURN server >>>> (rfc5766-turn-server for ICE/STUN). I am able to get to the point where >>>> the SIP server sends 183 session in progress to kamailio but after that >>>> I can only see - >>>> "STUN: using this candidate" >>>> "Successful STUN binding request from .." >>>> "SRTP output wanted, but no crypto suite was negotiated" >>>> >>> >>> This is fairly strange: >>> >>> Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port >>>> 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection >>>> profile negotiated >>>> Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port >>>> 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection >>>> profile negotiated >>>> >>> >>> Are you running a very old OpenSSL version by any chance? >>> >>> cheers >>> >>> >>> _______________________________________________ >>> 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 >>> >> >> >> >> -- >> 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 >> >> > -- Warm Regds. MathuRahul
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