[asterisk-dev] Native Bridge for H.323?

[Ken Chan]

Hello,

I am wondering has anyone implemented (or want to implement) the
"Native Bridge" function for the H.323 channel driver 
(i.e. "h323_native_bridge" function in
asterisk/channels/h323/ast_h323.cpp file)?

It is on the TODO List but any idea when it will be implemented?

Currently, when the call is between SIP Phone and Netmeeting (H.323),
all the RTP packets are going through Asterisk.  

a)  Is there any method I can avoid the RTP packets 
going through Asterisk?

b)  Does anyone know how to implement this feature?  I don't mind
doing it if someone is willing to share more information or
show me some tricks how to implement this part.  Any design document
I can follow?

Thanks
ken

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