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I'm tracing the code to understand how it works. I'm not far enough but:<br>
<br>
at main/translate.c the function "plc_fillin" is called. But there is
no difference if i disable it. I'm in the end for now because of big
complexity :-)<br>
<br>
Martin Vít wrote:
<blockquote cite="mid4536774B.3060906@lam.cz" type="cite">
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Hello, <br>
<br>
i've done some tests how the jitterbuffer works with SIP (asterisk
1.4b2) and tests shows that PLC is not working. <br>
<br>
<b>Configuration:</b><br>
<br>
E1 Multiplexer with analog phone<br>
<br>
Asterisk A: <br>
E1 wcte11xp<br>
version: 1.4b2 <br>
<br>
Asterisk B:<br>
version: 1.2.11 <br>
ztdummy for timing<br>
<br>
<b>Simple testing method:</b><br>
<br>
Call from analog phone via TDM -> E1 asterisk A -> SIP or IAX2
-> asterisk B (musiconhold)<br>
<br>
testing codecs between asterisk A and B: alaw, g726<br>
<br>
<br>
<b>Jitter simulation:<br>
<br>
</b><i>asterisk B:</i><b><br>
</b>modprobe sch_netem<br>
tc qdisc add dev eth0 root netem delay 0ms 0ms<br>
tc qdisc change dev eth0 root netem delay 0ms 300ms<br>
<br>
<b>Results:<br>
<br>
</b>module reload codec_alaw shows that PLC is true. <br>
<br>
Jitter is working for both IAX2 and SIP channels but without PLC. <br>
I've switched back to 1.2.11 and IAX2 PLC was correct (for alaw
chan_zap has to be patched to force codec to slinear, so transcoding
can do PLC)<br>
I've tried jitter buffer patch for 1.2 asterisk (from backports) and
SIP PLC is not working too. <br>
<br>
Sample audio:<br>
Random Jitter (0-300ms) and correct PLC <b><a
class="moz-txt-link-freetext" href="http://www.lam.cz/iax2.wav">http://www.lam.cz/iax2.wav</a></b><br>
Random Jitter (0-300ms) without PLC <b><a class="moz-txt-link-freetext"
href="http://www.lam.cz/sip.wav">http://www.lam.cz/sip.wav</a></b><br>
<br>
So my question is: do i have something wrong or it is bug? Any
suggestion will be appreciative. <br>
<br>
Festr<br>
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</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.vasesit.cz/">http://www.vasesit.cz/</a>
mobil: 605 267 610
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