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
