[asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration

Olli Heiskanen ohjelmistoarkkitehti at gmail.com
Sun Aug 10 13:27:31 CDT 2014


Thanks Daniel for your reply.

Sorry for having been a bit obscure, it is my intention to have all clients
able to call each other, regardless of which ua client software they use. I
think I've realized what's going on. My goal is to use rtpengine to bridge
between rtp profiles when they are different. But according to sip.js
instruction, I set up my clients in a way that Asterisk took the place of
rtpengine and changed the rtp profiles along the way based on the realtime
table values. That got me confused but now I know at least what the problem
is so I can fix it. This setup works in a way that I can make calls between
websocket and sip clients, but the problem with it is that I need different
values in the realtime table, according to which rtp profile the client

Doing this I made a wrong turn in my project, I'll need to have "universal"
setup for each peer so the user can use a websocket client or a sip client
to register and use an account. I'll still need to figure out which
settings to use and which not to use, so the rtp gets handled by rtpengine,
not Asterisk. But that's a question for the Asterisk list.

The problem about Asterisk setting the rtp profile as UDP/TLS/RTP/SAVPF was
fixed using a peer setting in the realtime table, now Asterisk accepts
RTP/SAVPF I can have calls flowing as soon as I can get rtpengine to
cooperate with me.

I wonder, is there UDP/TLS/RTP/SAVPF handling in rtpengine/kamailio? I may
have to add some kind of handling to this if I have to revert back to my
previous settings.


2014-08-05 16:49 GMT+03:00 Daniel-Constantin Mierla <miconda at gmail.com>:

> On 01/08/14 10:56, Olli Heiskanen wrote:
>> Hi,
>> I got ahead with my setup, this post helped me much:
>> http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=
>> 66fdf8cc4be5d955ba584e989a23442f
>> At least the avpf setting had to be removed from sip.conf and put in the
>> realtime db table, defined per client. I left the encryption setting in
>> sip.conf. I had some problems calling from SIP client to another, then had
>> to define avpf=no for those clients. Personally I don't like to use
>> different settings to different clients, is there a way around this?
>> With this setup I can make calls between SIP clients but not ws clients.
>> My client (now I use sip.js) fails to parse the sdp - including the
>> apparently correct rtp profile UDP/TLS/RTP/SAVPF - and sends back 488,
>> which makes the call fail. I'd like to hear opinions from you guys which
>> would be the correct place to handle this? My setup has Asterisk Kamailio
>> realtime integration, and I use dispatcher in Kamailio to route calls to
>> Asterisk. Kamailio sounds like the logical place, but I'd rather find a way
>> to not change the rtp profile along the way, at least until the clients can
>> support that one.
> To understand properly, you don't want to use rtpenging for
> srtp(webrtc)-rtp(classic sip) gatewaying?
> If yes, maybe you can partition the users (classic-sip and webrtc-sip),
> then use two asterisk instances with routing via kamailio.
> Cheers,
> Daniel
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
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