you don't buffer outgoing rtp. jitter is introduced on the wire after it leaves the box so buffering outgoing streams would only delay audio and not add any value.
On Wednesday, June 3, 2015, Kamil Nigmatullin <kamil.nigmatullin@gmail.com> wrote:Dear All,We have developed our own sip application that works with Freeswitch. So, on incoming rtp stream we can ajust jitter buffer on client side, but on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a way to switch Jitter Buffer only for outgoing stream in Freeswitch
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