<div dir="ltr">If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end.<div>That is why its not defaulted to being possible but you asked how to do it.</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <span dir="ltr"><<a href="mailto:rajil.s@gmail.com" target="_blank">rajil.s@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><br></div>The wiki page (<a href="https://wiki.freeswitch.org/wiki/Jitterbuffer" target="_blank">https://wiki.freeswitch.org/wiki/Jitterbuffer</a>) says, <br><pre>If both sides of a bridge are RTP and both sides have a jb, its fairly
useless. In fact if anything, it can worsen call quality.<br><br></pre><pre>So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?<br></pre></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On 15 February 2016 at 17:10, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">The variable must be set on the leg before the channel engages media.<div>you can test if its on with </div><div><br></div><div>uuid_jitterbuffer <uuid> debug:10</div><div><br></div><div><br></div><div>if you want it on both legs set the following global vars in vars.xml</div><div><br></div><div>
<p><span>rtp_jitter_buffer</span><span>_during_bridge=true</span></p><p><span><span style="font-size:12.8px">jitterbuffer_msec=60</span><br></span></p><p><span><span style="font-size:12.8px"><br></span></span></p><p><br></p></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <span dir="ltr"><<a href="mailto:rajil.s@gmail.com" target="_blank">rajil.s@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>Hello,<br>
<br>
I have two FS instances connected to each other over VPN. The VPN<br>
sometimes has poor connectivity and I was wondering whether<br>
jitterbuffer can help to improve the call quality. I am using the OPUS<br>
codec between the FS instances since it is supposed to behave better<br>
in lower bandwidth. The ATA on both ends are Linksys SPA3102.<br>
<br>
My connections is as follows:<br>
<br>
ATA_A---->FS_A---->vpn---->FS_B---->ATA_B<br>
<br>
<br>
When user A is calling to B where should i specify jitter buffer<br>
amongst the following:<br>
<br>
1. The bridge between Freeswitch instance FS_A and FS_B<br>
<action application="bridge"<br>
data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/><br>
<br>
2. The bridge between Freeswitch instance FS_B and ATA_B<br>
<action application="bridge"<br>
data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/><br>
<br>
Thanks,<br>
Rajil<br>
<br></div></div>
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