[asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
miconda at gmail.com
Tue Aug 5 08:49:45 CDT 2014
On 01/08/14 10:56, Olli Heiskanen wrote:
> I got ahead with my setup, this post helped me much:
> 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.
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