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.<span></span><br><br>On Wednesday, June 3, 2015, Kamil Nigmatullin <<a href="mailto:kamil.nigmatullin@gmail.com">kamil.nigmatullin@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Dear All,<div><br></div><div>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</div><div><br></div><div>Thanks <br clear="all"><div><br></div>-- <br><div><div dir="ltr">Kamil Nigmatullin<br>Tel: 77272323748<br>mob: 7 (707) 2517003<br>Skype: kamil.nigmatullin</div></div>
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