[Asterisk-Users] Asterisk and RTP flow

[Michael Manousos]

Tebaldi Marco wrote:
> Hi all,
>  
> i' using asterisk with chan_oh323 to test the SIP/H.323 interoperability. All working well...
>  
> Now i would like to know if is it possible that * doesn't control the RTP traffic.
>  
> I would tht the end point control the RTP flow...
>  
> Any idea, is it possible???

No, this is unimplemented.

>  
> Thanks 
> Marco


Michael.