Nanang, Thank you for your response. Turned out that our problem was related to a bad SIP message. But thanks for the info on jitter buffer, will come in handy sometime. Regards, Vikram. Nanang Izzuddin wrote: > Hi, > > Please follow the inline comments > > On Thu, Apr 8, 2010 at 4:22 AM, Vikram Ragukumar > <vragukumar at signalogic.com> wrote: >> Hello, >> >> We are making VoIP calls using a softphone that uses the pjsip stack. Calls >> experience high jitter. >> >> In case there is an under run of the jitter buffer (either jitter buffer is >> not enabled or buffer size is insufficient) >> > > Currently the jitter buffer size is set to 500ms, this may be > insufficient on such high jitter network. And insufficient size will > cause jitter buffer discarding frames, which leads to jitter buffer > under run indeed. > >> 1) Would there be an impact on the transmission of RTP from the softphone? > > No, jitter buffer is only used in the receiving direction. > >> 2) Would there be an error message that is indicative of a jitter buffer >> under run ? If yes, what is the expected error message ? > > Kind of "jitter buffer empty" message in the log, lots of them in the > whole call session. > > BR, > nanang > > >> Thanks and Regards, >> Vikram. >> >> _______________________________________________ >> 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 >> > > _______________________________________________ > 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
