[Asterisk-Users] sip to h323 ... direct RTP?

Kevin P. Fleming kpfleming at digium.com
Fri Jun 23 00:53:23 MST 2006


----- Jeremy McNamara <jj at nufone.net> wrote:
> The problem is 're-inviting' in H.323-jive is very much a non-trivial
> task.

Ahh, OK, then this is a protocol limitation more than an implementation issue. Never mind :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.




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