[asterisk-dev] [Video / SIP] Codec negotiation passthrough ?

Nicolas Bourbaki ncl.bourbaki at gmail.com
Thu Sep 2 10:30:38 CDT 2010


2010/9/2 Simon Perreault <simon.perreault at viagenie.ca>

> On 2010-09-02 10:54, Nicolas Bourbaki wrote:
> > my goal is to have, for any video call a well defined path for RTP flow,
> > which can be :
> > - Client A --> "RTP proxy 1" --> "RTP proxy 2" --> "RTP proxy N" -->
> > client B
> > with "RTP proxy" 1, 2, 3, ... in the same LAN segment (in fact not
> > really but it's the best and easiest way to explain our objectives)
> >
> > OpenSIPs + rtpproxy doesn't work very well (can't pass through multiple
> > proxy :s)
>
> Multiple OpenSIPs then?
>


I've tried, but does'nt works. RTP packet flow is the following :
from A to B : client A --> rtpproxy1 --> client B
from B to A : client B --> rtpproxyN --> client A



>
> Simon
> --
> NAT64/DNS64 open-source --> http://ecdysis.viagenie.ca
> STUN/TURN server        --> http://numb.viagenie.ca
> vCard 4.0               --> http://www.vcarddav.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20100902/d4026630/attachment.htm 


More information about the asterisk-dev mailing list