Hi Raja, It is high jitter value for LAN indeed. Jitter is usually caused by network, sound device lag/burst, CPU load spike, and, as you mentioned, perhaps there is bug. You said that RTT sometimes jumped to 1000ms, it will just show that there may be network problem. About bug possibility, I would just say it is much less possible since audio quality tests have been done quite a lot and jitter in LAN has never raised an issue. I am afraid I couldn't see other clue from the info you've given. Regards, nanang On 04/07/2008, raja <raja at solnettechnologies.com> wrote: > > > Hi All, > > We have two pjsip endpoints talking to each other through asterisk, in > asterisk we had set canreinvite to yes so that the media is not routed > through asterisk.We are using ulaw codec for rtp media. > > we were continuously monitoring the media quality by collecting rtcp > statistics by doing dd(dump detailed) command in pjsua, in this I am > consistently seeing RX MAX-Jitter as 1000ms for all calls.I am wondering > if there may be some issue with thread locking or systems timer or > interrupt in pjsip that sometimes occurs resulting in the 1000 ms > addition.And also RTT values varies between 2-3ms as we might expect on > a LAN but sometimes jumps to 1000ms.I couldn't understand why the Rx- > Max-Jitter value always stays at 1000ms, Is this anything to do with > codec part?.please help me in this. > > Thannks in Advance > > with regards > raja > > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080706/90b35e31/attachment.html
