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@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@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
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