[asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk
ohjelmistoarkkitehti at gmail.com
Tue Aug 12 03:17:22 CDT 2014
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.
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.
2014-08-12 1:56 GMT+03:00 Paul Belanger <paul.belanger at polybeacon.com>:
> On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen
> <ohjelmistoarkkitehti at gmail.com> wrote:
> > Hello,
> > I'm trying to get calls working between websocket clients and sip
> > For clients I have sip.js based clients on chrome, Zoipers and a
> > phone. Challenge here is I'd like to have Kamailio and rtpengine to
> > the bridging between different rtp profiles but Asterisk changes them in
> > sdp bodies along the way. I'm using Asterisk 11.11.0.
> > Is there a way to configure Asterisk to ignore the rtp profile but allow
> > calls to pass with either of those profiles (even though clients might
> > answer with 488 which would be caught and handled by Kamailio and
> > rtpengine)? In my setup I have Asterisk Kamailio realtime integration,
> > the second goal is to be able to add peers to the db table with similar
> > data, as in no different values based on what kind of client wants to
> > register. I'd like to allow the user to register using which ever client
> > they choose (in this case one of the 3 I mentioned).
> > Previously I had problems like 'rejecting secure audio stream without
> > encryption details', no audio or BYE messages sent immediately after call
> > has begun etc, but according to sip.js documentation
> > (http://sipjs.com/guides/server-configuration/asterisk/) the settings
> > and force_avp affect the way Asterisk handles the rtp profiles and now my
> > calls do work ok but I'd need to move the rtp profile handling to
> We are successfully using kamailio / rtpengine with websockets and
> asterisk 1.8. First question is why are you duplicating registrations
> within asterisk? Secondly, why are you using websockets in asterisk?
> Without knowing more about your use case, I'll tell you how we did it.
> Like I said, kamailio is responsible for our SIP/ws subscribers and
> registrations. Once within kamailio we simply dispatch traffic to
> asterisk via SIP/udp. RTP is handled by rtpengine (using rtproxy-ng)
> and that is basically it.
> No special configuration is needed for asterisk (in fact 1.8 has no
> support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a
> kamailio peer and away you go.
> Paul Belanger | PolyBeacon, Inc.
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> Github: https://github.com/pabelanger | Twitter:
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