<div dir="ltr">Hi,<br><br>I have need to measure the end-to-end audio delay in the MeetMe conference application.<br><br>Currently, I have written a python program that connects to an Asterisk MeetMe conference via SIP, and pumps RTP packets into the conference.<br>
<br>Another instance of the program dials into the same conference and receives the mixed RTP stream.<br><br>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.<br>
<br>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.<br>
<br>Anyone have any suggestions on how I can do this, given my setup?<br><br>Thanx!<br><br><br><br><br></div>