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

Olli Heiskanen ohjelmistoarkkitehti at gmail.com
Wed Aug 13 03:35:10 CDT 2014


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.

cheers,
Olli


2014-08-12 17:40 GMT+03:00 Paul Belanger <paul.belanger at polybeacon.com>:

> On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen
> <ohjelmistoarkkitehti at gmail.com> wrote:
> > Hello,
> >
> > Thank You Paul for your reply,
> >
> > The registrations in my setup are not duplicated, the 'secret' field in
> the
> > realtime table is empty, which causes Asterisk to not authenticate
> requests
> > from my Kamailio. Kamailio handles registrations, and also routes the
> > traffic to Asterisk using dispatcher. Also, all peers have the Kamailio
> > ip:port as outbound proxy so all traffic goes through Kamailio.
> >
> That is your issue, stop using chan_sip with realtime (using data from
> kamailio).  The only SIP peer asterisk should know of is kamailio, and
> your webrtc clients should be anonymous SIP users.  This way, Asterisk
> doesn't even need to deal with websockets and RTP/SAVPF (this is what
> kamailio and rtpengine) is for.
>
> In your current setup, you are bypassing the functionality of
> rtpengine and not even leveraging it.
>
> > Looks like version 11.11 works differently, I'll try to revert back to a
> > previous version, and see if that works. I know at least the 'force_avp'
> > field is new to 11.11 so it's safe to assume there's some difference
> between
> > versions in rtp profile handling.
> >
> > It would be good to know how to handle this scenario in the new versions
> as
> > well, I'll probably need to upgrade ahead anyway.
> >
>
>
> --
> Paul Belanger | PolyBeacon, Inc.
> Jabber: paul.belanger at 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140813/141403b5/attachment.html>


More information about the asterisk-users mailing list