Hi, thanks Kristian for your answer, it make sense 2009/11/12 Kristian Kielhofner <[email protected]>
> 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 <[email protected]> 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 > > [email protected] > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > [email protected] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Lei.Tang [email protected]
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