[Freeswitch-users] RTP packet loss on outgoing streams

Stanislav Sinyagin ssinyagin at gmail.com
Sat Apr 18 00:47:15 MSD 2015


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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