<div dir="ltr"><div>What I mean, I have this scheme: [Android Phone on 3G] <---> [Freeswitch] <---->[DID Provider]. When Android user calls DID Provider, I want to apply Jitter Buffer on Freeswitch so that voice (stream) coming from Android become smoother. On the same time, I want that the stream from DID provider of the same call was not affected by Jitter Buffer and will be delivered to Android Phone unchanged. <br><br></div>Thanks<br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-03 21:09 GMT+06:00 Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">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.<div class="HOEnZb"><div class="h5"><span></span><br><br>On Wednesday, June 3, 2015, Kamil Nigmatullin <<a href="mailto:kamil.nigmatullin@gmail.com" target="_blank">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 <a href="tel:%28707%29%202517003" value="+17072517003" target="_blank">(707) 2517003</a><br>Skype: kamil.nigmatullin</div></div>
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