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Pavel Jezek wrote:
<blockquote cite="mid:470D333E.5080001@i.cz" type="cite">
<pre wrap="">
Russell Bryant wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Pavel Jezek wrote:
</pre>
<blockquote type="cite">
<pre wrap="">- not working when bridge two channles with different jb implementation
- eg. sip/h323/skinny & iax
</pre>
</blockquote>
<pre wrap="">I'm not exactly sure what you mean here. You generally don't want to use a
jitterbuffer in this situation. You only want to use it at the endpoint.
</pre>
</blockquote>
<pre wrap=""><!---->I have excessive jittery connection (wifi/cdma), endpoints (eg. ci$co
phones/gateways) can't cover this excessive jitter, I must dejitter on
asterisk and send to endpoint with smaller jitter...
I think dejittering is also needed before codec translation eg.:
sipA--(jittery
connection)--(iLBC)asterisk1(alaw)----iax----asterisk2---(alaw)sipB
I think, to be efective plc codecs algorithm, I must dejitter _before_
dojing codecs translation, so I must dejitter on asterisk1, not
asterisk2 or even endpoints</pre>
</blockquote>
it depends, on which channel translation is. For PLC it have to be in
outgoing channel (after dejjitering). <br>
<blockquote cite="mid:470D333E.5080001@i.cz" type="cite">
<pre wrap="">...
but it's not possible, because bridging channels with different jb
implementations (sip vs. iax) on asterisk1
all this will be clearly solved, when jb will be efective in incomming
direction on _incomming_ channel, as many people expected
</pre>
</blockquote>
If you want dejjitering on incoming channel, this channel must doing
recode to do correct PLC. <br>
<br>
<blockquote cite="mid:470D333E.5080001@i.cz" type="cite">
<pre wrap="">I think, kind of currently designed jb in asterisk is confusing for many
people....
</pre>
<blockquote type="cite">
<pre wrap="">
</pre>
<blockquote type="cite">
<pre wrap="">- jb is applied in incomming direction, but on _outgoing_ channel! :-\
</pre>
</blockquote>
<pre wrap="">I wrote a patch yesterday that allows this to work, even when connecting to an
Asterisk application such as Voicemail or MeetMe. See this post for more details:
<a class="moz-txt-link-freetext" href="http://russellbryant.net/blog/?p=17">http://russellbryant.net/blog/?p=17</a>
</pre>
</blockquote>
<pre wrap=""><!---->
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</pre>
</blockquote>
<br>
<br>
<pre class="moz-signature" cols="72">--
Martin Vít
LAM plus s.r.o.
<a class="moz-txt-link-freetext" href="http://www.lam.cz/">http://www.lam.cz/</a>
Tel.: 605 267 610
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