[asterisk-dev] Why does chan_sip disable early bridging???

[Olle E Johansson]

3 okt 2007 kl. 10.48 skrev Sergio Garcia:

>
>
> Thanks for the clarification.
> Probably I'm again wrong, but I assume that the main problem is  
> knowing
> if the remote peer has compatible codecs in order to keep asterisk  
> in the
> media loop or not. Could it be solved if we implement the option of
> sending INVITEs without SDP and sending it in the ACK?
>
I think we do support it now. But the problem is not Asterisk sending  
the INVITE
but another device.

/O