<div dir="ltr">What i'm getting at is you should never have dropped packets on the host itself like that, If you do then you probably have WinNic's aka RealTek garbage.</div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Apr 17, 2015 at 3:47 PM, Stanislav Sinyagin <span dir="ltr"><<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">you can try playing with the buffer size (-B option), but in general,<br>
tshark is quite CPU-intensive when it tries to analyze the UDP packet<br>
stream in real time. Also I'm not sure if it's multithreaded -- I<br>
guess it's not, so it tries to analyze all the streams in a single<br>
thread in real time.<br>
<br>
I made this set of scripts to run on a low-power CPU (Atom and such):<br>
<a href="https://github.com/voxserv/voip_qos_probe" target="_blank">https://github.com/voxserv/voip_qos_probe</a><br>
What I do is run tcpdump during the call, store the capture, and then<br>
analyze the output with tshark in low priority.<br>
<br>
Also I noticed that on a Xen VM on an Intel CoreDuo CPU, you need to<br>
increase the tcpdump's buffer size with -B option. With default<br>
settings, sometimes it's not even able to capture a single SIP session<br>
without dropping packets.<br>
<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
<br>
On Fri, Apr 17, 2015 at 5:01 PM, mbo <<a href="mailto:mbodbg@gmx.net">mbodbg@gmx.net</a>> wrote:<br>
> Hello,<br>
><br>
> we need to monitor the voice quality on our server. To test this, we setup a<br>
> test environment with 2 freeswitches, originating calls from freeswitch 1<br>
> and then answer them on freeswitch 2 and play an announcement. To analyze<br>
> the traffic, we us tshark with the following command:<br>
><br>
> tshark -q -f 'udp' -o rtp.heuristic_rtp:TRUE -z rtp,strems -w test-pcap<br>
><br>
> The result is, that we see packet loss around 20% on outgoing RTP streams,<br>
> when we have more then 20 calls. The load average shown with the top command<br>
> is always < 0.3 with a 4 core CPU and CPU load is less then 10%.<br>
><br>
> ========================= RTP Streams ========================<br>
> Src IP addr Port Dest IP addr Port SSRC Payload<br>
> Pkts Lost Max Delta(ms) Max Jitter(ms) Mean Jitter(ms) Problems?<br>
> 192.168.1.134 27590 192.168.1.135 22576 0xCD7A189D ITU-T G.722<br>
> 800 2 (0.2%) 59.93 0.11 0.05 X<br>
> 192.168.1.135 22576 192.168.1.134 27590 0x5D4E50CD ITU-T G.722<br>
> 599 202 (25.2%) 160.24 0.16 0.05 X<br>
> 192.168.1.134 28172 192.168.1.135 29704 0xD53EC9AD ITU-T G.722<br>
> 798 2 (0.2%) 59.99 0.07 0.04 X<br>
> 192.168.1.135 29704 192.168.1.134 28172 0x5D50942D ITU-T G.722<br>
> 713 86 (10.8%) 199.98 0.08 0.04 X<br>
> 192.168.1.135 25794 192.168.1.134 16460 0x5D51E43D ITU-T G.722<br>
> 633 165 (20.7%) 180.25 0.75 0.07 X<br>
><br>
> Instead of tshark, we also tried to capture SIP & RTP with tcpdump with the<br>
> following command:<br>
><br>
> tcpdump -i any -T rtp -vvvvv -s 0 -w test.pcap<br>
><br>
> If we analyze this file with wireshark (TELEPHONY->RTP->Show all Streams),<br>
> it shows almost always 0% for packet loss. Only if we increase the load to<br>
> 50 calls we can see a similar scenario where inbound calls still have < 1%<br>
> packet loss but outgoing RTP streams have up to 20 % packet loss. We also<br>
> repeated the same test with freeswitch 1.4.15 with much more powerful<br>
> hardware with almost the same results.<br>
><br>
> The question is what do we see here. Does tshark show us wrong result? Can<br>
> we more trust on the results captured with tcpdump or is there finally an<br>
> issue in freeswitch when sending outgoing packets? Or is the method to<br>
> capture on the server in general the wrong approach?<br>
><br>
> Thanks for any comments and hints<br>
><br>
> Markus<br>
><br>
><br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">
<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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