[Sipp-users] Monitoring RTP packet loss

[Eric L. <er...@te...>]

Hello,

I'm relatively new to SIPp. I'm trying to set up a call load stress test
that not only tests the SIP signaling, but also checks for RTP packet loss.
So far, I have set up a simple scenario that makes calls and sends RTP
(from an audio file), both with and without RTP echo from the server end
(both tests are relevant in my environment--one-way and two-way audio).
This works, but I can't determine if there's a way to get statistics on the
RTP streams. Specifically, packet loss and ideally also jitter, latency,
and/or out-of-order packets (the streams are UDP). I can capture the
session and use Wireshark to review each call and its associated RTP
stream, but unless I'm missing something in Wireshark, there's no
"aggregated" way to look at those details, and looking at each individual
call in a session of 1000s is not a practical solution. If anyone else has
tried to do this, I'd be grateful for any ideas. And if SIPp isn't the
right tool for this, what else should I consider? (By the way, I had
originally done some basic tests UDP load tests using iperf, but that's not
quite the same as monitoring actual RTP call traffic).

Regards,
Eric