[asterisk-dev] Avoiding RTP flows (new topic)

[Olle E. Johansson]

23 apr 2009 kl. 10.03 skrev Venefax:

> I have complete control over the codecs. For a second leg of the  
> call, I
> only offer the same codec negotiated in the inbound leg. So maybe
> directrtp=yes is actually working in my application. Today I had 300  
> open
> calls and 20 calls per second, and the processor never went over  
> 20%. The
> question is, how do I know for sure how is the media being handled.  
> Is there
> any way to positively detect that?
> F.Alves
Well, now this is a question for asterisk-users ;-)

Try "rtp debug" in the cli.

/O