[Sipp-users] Monitoring RTP packet loss

[Eric Lenington]

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

[Rob Day]

Hi Eric,

SIPp doesn't currently support many statistics on RTP packets (though
it's something I might consider adding in the future). Until then, in
your situation, what I'd probably do is to capture all the RTP packets
with Wireshark/tcpdump, then write a small program using Python and
scapy (or similar tools, like rubypcap or jNetPcap, if you prefer
other languages) to retrieve the packets, separate them out into
calls, process the streams to calculate packet loss/delay/ordering
issues, and generate some sort of statistical report.

If you end up writing (or finding) a tool like that, do consider
emailing the list to let others know - I think it's something that
others would find useful, and I'd be happy to link to it from the SIPp
docs.

It might also be worth making a sample of your RTP capture available,
so that anyone else on the list who knows of such a tool (or might
want to write one) can see if it's suitable.

Hope that helps,
Rob

Post by Eric Lenington
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
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