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

[Klaus Darilion]

Venefax schrieb:
> 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?

Take a look at the incoming/outgoing SDP.

Use tcpdump/wireshark

klaus