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

[Lei Tang]

Hi, thanks Kristian for your answer, it make sense

2009/11/12 Kristian Kielhofner <kristian.kielhofner at gmail.com>

> 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
>
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-- 
Lei.Tang
lei.tlfly at gmail.com
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