Hi! In the application that I am testing, I have a problem with frame lost, that is why the audio quality is very poor. The packet size of RTP are irregular so I think this is the reason why there are so many frame lost. How can I configure my stream/jitter Buffer to handle this? Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150730/b6a72a2e/attachment.html>
