[sip-comm-dev] Buffer size of sending RTPManagers in SIP calls


Hello all!

CallSessionImpl.initializeRtpManager sets a buffer size of 500ms on the
sending RTPManagers. This makes for a long latency, so I wondered if there's
a reason for it. In my short tests, a buffer size of 125ms provided good
audio quality with vastly reduced latency, but I fear that there could be
side effects that I didn't find with my tests (like choppy audio on slow
computers/different hardware?). Has anyone else experimented with the

Michael Koch