Hi Raja, AFAIK, RTT calculation is based on RTCP packets, nothing to do with the codecs. So first, please make sure RTCP traffic is received correctly, e.g: using wireshark/net sniffing tool. Regards, nanang 2008/6/24 raja <raja at solnettechnologies.com>: > Hi all, > > I am running pjsip(0.8.0) calls between US and india through asterisk.I > am also consecutively monitoring rtcp informations in these > pjsi[ endpoints. > > In this I am always getting all the orig and term RTT as 0.000ms and > also all orig jitter as 0.000ms and only get term max jitter as > 1015.000ms. > > I am using PCMU codec for the calls,I am not sure why the orig/term RTT > are always shown as 0.000ms. please help me in explaining why RTT values > are zero.And also how the Orig jitter values are also 0.000ms. > > Also note that other codecs like sppex,ilbc are giving non-zero values. > > 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/20080625/e8ea27e8/attachment.html
