[asterisk-users] strategy for measuring conference audio delay

[George Williams]

Hi,

I have need to measure the end-to-end audio delay in the MeetMe conference
application.

Currently, I have written a python program that connects to an Asterisk
MeetMe conference via SIP, and pumps RTP packets into the conference.

Another instance of the program dials into the same conference and receives
the mixed RTP stream.

I figure I can have both python programs running on the same machine -
essentially creating an echo test setup.  Then, all I have to do is measure
the time delay between when I send the audio stream to when I receive it.

I think I'm being a bonehead, but the trick appears to be in what kind of
RTP packets to generate and how to analyze the mixed audio stream coming
back from Asterisk.  Preferably, I would send one RTP packet that gets mixed
in a predictable way and then I can look for it in the receive stream.

Anyone have any suggestions on how I can do this, given my setup?

Thanx!
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