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

[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