[asterisk-dev] RTP streams between H323 and SIP

[Johansson Olle E]

27 jun 2006 kl. 04.22 skrev Paul Cadach:

> Hello,
>
> Johansson Olle E wrote:
>>> Are you sure? All channels that supply 'ast_rtp_bridge' as their
>>> bridge method should already support it.
>>
>> That was news to me. Cool if that's the case!
>> Never seen or heard about it before.
>>
>> I stand corrected.
>
> I just acknowledge we have H.323 native bridge code that support  
> RTP "move" (like re-invites for SIP) and all works fine
> between SIP and H.323 channels.
Cool. When do we see that in the bug tracker!

/O