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

[Jeremy McNamara]

Kevin P. Fleming wrote:
> I believe this is incorrect; all the RTP-using channel drivers supply 'ast_rtp_bridge' as their native bridge method, so assuming they also implement the 'set_rtp_peer' method, then an RTP native bridge between dissimilar channels should work fine. If the channel driver(s) also support sending the RTP peer address to the endpoint (as chan_sip does with reinvite), then a direct media path should also be possible.
> 


The problem is 're-inviting' in H.323-jive is very much a non-trivial task.


Jeremy McNamara