[asterisk-dev] Feature: IAX2-controlled RTP native bridges

Paul Cadach paul at odt.east.telecom.kz
Tue Feb 28 10:51:32 MST 2006


Kevin P. Fleming wrote:
> > IMHO that would be nice to have a way to provide native briding for such type of calls:
> > a) signalling: ep1->Asterisk1, Asterisk2->ep2
> > b) voice: ep1->ep2 directly.
> This sounds extremely complicated, and I doubt the utility of it when
> you could just use SIP between the boxes and it will work already.
> What is the value of using IAX2 between the boxes in this case, when the
> media is going to be pulled away from them anyway?

ep1 and/or ep2 could or could not allow RTP transfers. When native RTP transfer isn't possible better is to use IAX
between Asterisk1 and Asterisk2 instead of SIP with RTP. Also, transfers of RTP streams between endpoints didn't
requires additional UDP sockets, so it offloads Asterisk boxes a little.


More information about the asterisk-dev mailing list