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

[Kaloyan Kovachev]

 I guess common scenario is when few boxes are using IAX switching to share
their dialplan and the endpoints are even in the same network, so in this case
it is possible to migrate the IAX signaling to SIP messages and to leave the
RTP flow between the endpoints. 
 May be ... if there is a sip_migrate_port=5060 in the IAX peer config it
means we should try to migrate. We send DIAL message with IE indicating that
it is comming from canreinvite SIP peer and if the situation is the same (SIP
peer with canreinvite=yes) at the other side it sends an OPTION requesting
that IAX channel migrates (masqueraded) to SIP. SIP is trying to do native
bridging and if it succeds to move out of the RTP path at least one of the *
boxes, will not try to go back to IAX.
 It doesn't look too complex as there is already masquerade function in place,
which will do most of the job, but maybe i am wrong.

On Tue, 28 Feb 2006 12:13:13 -0600, Kevin P. Fleming wrote
> Paul Cadach wrote:
> 
> > 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.
> 
> I can't see how the complexity of supporting something like this is
> worth the  potential savings using IAX2 instead of SIP/RTP between 
> the Asterisk servers. I could be wrong, though, but this will not be 
> an easy thing to design, code and support.
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