[asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

Paul Belanger paul.belanger at polybeacon.com
Wed Aug 13 12:07:53 CDT 2014


On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen
<ohjelmistoarkkitehti at 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.belanger at polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger



More information about the asterisk-users mailing list