On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen <ohjelmistoarkkite...@gmail.com> wrote: > Hi, > > Wow, thanks Paul, realizing the problem makes a lot of sense. > > So I setup Kamailio as a peer, but if I disable chan_sip module completely, > I can't do it in sip.conf like I'd otherwise assume to do. I tried to > rebuild Asterisk without chan_sip, but I guess that's not quite the way to > go? Asterisk stopped sending back any sip messages so either there is a > configuration means on how to do this or I'm doing something wrong with my > current setup. My next thought was to compile Asterisk normally and set > rtcachefriends to no, that did not work either, when dialing the cli stated: > app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' > (cause 20 - Subscriber absent) > which I guess says Asterisk does not know where to send the message. > > The inner workings of Asterisk is a bit beyond me, if you don't mind giving > advice on how to proceed I'd be most grateful. > I think you are still mis-understanding me. I'll try to be clearer.
>From the POV of asterisk, you do still need chan_sip, however the only peer asterisk needs to be away of it Kamailio. All other peers will be stored within kamailio. This was the reason for my comment about realtime sip, you don't need it. Then, within kamailio, you'll need to invoke rtpengine using (rtpproxy-ng with kamailio 4.1) to rewrite the sdp for the invite to asterisk. You'll use the rtpproxy_offer and rtp_answer functions to remove ICE when calls originate from webrtc clients. Since you are not using a websocket in asterisk, it will just be a SIP over udp, the need for ICE and SAVPF is not needed. What you are trying to do is pretty complicated, it took me about 2 weeks to get everything setup properly. There is good information[1] on the web, you just need to google for it. [1] http://www.slideshare.net/crocodilertc/webrtc-websockets -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
