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

Olli Heiskanen ohjelmistoarkkitehti at gmail.com
Tue Aug 12 06:56:52 CDT 2014

Okay, tried reverting to Asterisk 11.10.2. I didn't change the realtime
table yet, but now when calling from websocket client to another websocket
client, cli says:

WARNING[30620][C-00000000]: chan_sip.c:11056 process_sdp_a_dtls:
Unsupported fingerprint hash type 'sha-2' received on dialog
WARNING[30620][C-00000000]: chan_sip.c:10509 process_sdp: Rejecting secure
audio stream without encryption details: audio 10640 RTP/SAVPF 111 103 104
0 8 106 105 13 126

This many times, until the forking capacity of Kamailio has been reached
and call fails. The clients are running on chrome, and calls have worked
before... I wonder if I should revert further back and/or change or remove
some realtime table fields?


2014-08-12 11:17 GMT+03:00 Olli Heiskanen <ohjelmistoarkkitehti at gmail.com>:

> 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.
> 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.
> Thanks,
> Olli
> 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
>> clients.
>> > For clients I have sip.js based clients on chrome, Zoipers and a
>> Grandstream
>> > phone. Challenge here is I'd like to have Kamailio and rtpengine to
>> handle
>> > the bridging between different rtp profiles but Asterisk changes them
>> in the
>> > 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,
>> and
>> > 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
>> avpf
>> > 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
>> rtpengine.
>> >
>> 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:
>> https://twitter.com/pabelanger
>> --
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