[Freeswitch-users] How to test FS rtp packet lost rate?

[Kristian Kielhofner]

The simplest way I know of is to bring up another call from a local
phone and listen to the audio.  At the same time run tcpdump/etc with
a strict filter to capture the rtp to/from that phone.  You can then
run RTP stream analysis and the like in Wireshark to identify any lost
packets.  While this obviously won't identify any/all potential lost
packets it will be a lot more practical than any of the alternatives:

- Capturing all media streams for RTP analysis
- Implementing RTCP to identify lost packets
- Commercial hardware/software

If FreeSWITCH, your machine, or your network are pushed to the max and
falling apart you're most likely going to see audio problems on your
single (captured) call.

On Wed, Nov 11, 2009 at 8:43 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
> Hi all, I'm testing a FS server using sipp, I found that sipp only show the
> retrans of sip packet, Does someone known is there a tool to test FS rtp
> packet lost rate in high concurrent call env?
>
> --
> Lei.Tang
> lei.tlfly at gmail.com
>
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-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com