FYI, this is very useful information so I tossed it up on the wiki on the jitterbuffer page:<br><a href="http://wiki.freeswitch.org/wiki/Jitterbuffer#Interesting_Information">http://wiki.freeswitch.org/wiki/Jitterbuffer#Interesting_Information</a><br>
<br>If anyone feels like making it look prettier by all means do so. :)<br><br>-Michael<br><br><div class="gmail_quote">On Mon, Aug 20, 2012 at 10:25 PM, 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">If the jitterbuffer is already on, calling it again will just resize<br>
it, so setting it to the same is redundant but harmless.<br>
<br>
If you are surprised by why the jitterbuffer is paused during bridge:<br>
<br>
If both sides of a bridge are RTP and both sides have a jb, its fairly<br>
useless. In fact if anything, it can worsen call quality.<br>
<br>
You should only run jitterbuffers at points of termination change of<br>
protocol. Examples, if FS was hosting a conference or IVR, if you are<br>
bridging the call to a phone for instance, you want to not use a<br>
jitterbuffer because you want to preserve the original timestamps so<br>
your phone can use its own jitterbuffer.<br>
<br>
For special examples where you are using FS jitterbuffer in front of<br>
something else that may not have one or some other special<br>
circumstance you can use the setting chris mentioned to leave it<br>
running.<br>
<div><div class="h5"><br>
</div></div></blockquote></div><br>