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

Daniel-Constantin Mierla miconda at gmail.com
Tue Aug 5 08:49:45 CDT 2014

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.


Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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