[Freeswitch-users] RTP packet loss on outgoing streams

mbo mbodbg at gmx.net
Mon Apr 20 16:44:53 MSD 2015 

I did some more test and when I use the -B 32768 option with tcpdump I can trace up to 150 calls without packets dropped by the kernel. Finally that means as soon as you have more than 15 parallel calls on the server it’s very important to set this parameter to not get wrong results. 

Many thanks for your support

Markus


Am 17.04.2015 um 23:13 schrieb Stanislav Sinyagin <ssinyagin at gmail.com>:

> it's not dropped packets, it's the packets that didn't make it to tshark's buffer, because it was too slow and the buffer was too short. 
> 
> On Fri, Apr 17, 2015 at 10:49 PM, Brian West <brian at freeswitch.org> wrote:
> 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.
> 
> On Fri, Apr 17, 2015 at 3:47 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> you can try playing with the buffer size (-B option), but in general,
> tshark is quite CPU-intensive when it tries to analyze the UDP packet
> stream in real time. Also I'm not sure if it's multithreaded -- I
> guess it's not, so it tries to analyze all the streams in a single
> thread in real time.
> 
> I made this set of scripts to run on a low-power CPU (Atom and such):
> https://github.com/voxserv/voip_qos_probe
> What I do is run tcpdump during the call, store the capture, and then
> analyze the output with tshark in low priority.
> 
> Also I noticed that on a Xen VM on an Intel CoreDuo CPU, you need to
> increase the tcpdump's buffer size with -B option. With default
> settings, sometimes it's not even able to capture a single SIP session
> without dropping packets.
> 
> 
> 
> 
> On Fri, Apr 17, 2015 at 5:01 PM, mbo <mbodbg at gmx.net> wrote:
> > Hello,
> >
> > we need to monitor the voice quality on our server. To test this, we setup a
> > test environment with 2 freeswitches, originating calls from freeswitch 1
> > and then answer them on freeswitch 2 and play an announcement. To analyze
> > the traffic, we us tshark with the following command:
> >
> > tshark -q -f 'udp' -o rtp.heuristic_rtp:TRUE -z rtp,strems -w test-pcap
> >
> > The result is, that we see packet loss around 20% on outgoing RTP streams,
> > when we have more then 20 calls. The load average shown with the top command
> > is always < 0.3 with a 4 core CPU and CPU load is less then 10%.
> >
> > ========================= RTP Streams ========================
> >     Src IP addr  Port    Dest IP addr  Port       SSRC          Payload
> > Pkts         Lost   Max Delta(ms)  Max Jitter(ms) Mean Jitter(ms) Problems?
> >   192.168.1.134 27590   192.168.1.135 22576 0xCD7A189D      ITU-T G.722
> > 800     2 (0.2%)           59.93            0.11            0.05 X
> >   192.168.1.135 22576   192.168.1.134 27590 0x5D4E50CD      ITU-T G.722
> > 599   202 (25.2%)          160.24            0.16            0.05 X
> >   192.168.1.134 28172   192.168.1.135 29704 0xD53EC9AD      ITU-T G.722
> > 798     2 (0.2%)           59.99            0.07            0.04 X
> >   192.168.1.135 29704   192.168.1.134 28172 0x5D50942D      ITU-T G.722
> > 713    86 (10.8%)          199.98            0.08            0.04 X
> >   192.168.1.135 25794   192.168.1.134 16460 0x5D51E43D      ITU-T G.722
> > 633   165 (20.7%)          180.25            0.75            0.07 X
> >
> > Instead of tshark, we also tried to capture SIP & RTP with tcpdump with the
> > following command:
> >
> > tcpdump -i any -T rtp -vvvvv -s 0 -w test.pcap
> >
> > If we analyze this file with wireshark (TELEPHONY->RTP->Show all Streams),
> > it shows almost always 0% for packet loss. Only if we increase the load to
> > 50 calls we can see a similar scenario where inbound calls still have < 1%
> > packet loss but outgoing RTP streams have up to 20 % packet loss. We also
> > repeated the same test with freeswitch 1.4.15 with much more powerful
> > hardware with almost the same results.
> >
> > The question is what do we see here. Does tshark show us wrong result? Can
> > we more trust on the results captured with tcpdump or is there finally an
> > issue in freeswitch when sending outgoing packets? Or is the method to
> > capture on the server in general the wrong approach?
> >
> > Thanks for any comments and hints
> >
> > Markus
> >
> >
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> 
> -- 
> Brian West
> brian at freeswitch.org
> 
> 
> 
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