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

[Paul Cadach]

Hello,

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

Netmeeting isn't support for "Empty TCS (Terminal Capability Set)" feature yet, so RTP traffic will go through Asterisk
anyway independedly on native bridge support for H.323.


We have implemented native bridge for chan_h323 but it is in very preliminary stage.


WBR,
Paul.