[asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk
paul.belanger at polybeacon.com
Tue Aug 12 09:40:47 CDT 2014
On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen
<ohjelmistoarkkitehti at gmail.com> wrote:
> 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)
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