I'm tracing the code to understand how it works. I'm not far enough but:
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 :-)
Martin Vít wrote:
Hello,
i've done some tests how the jitterbuffer works with SIP (asterisk
1.4b2) and tests shows that PLC is not working.
Configuration:
E1 Multiplexer with analog phone
Asterisk A:
E1 wcte11xp
version: 1.4b2
Asterisk B:
version: 1.2.11
ztdummy for timing
Simple testing method:
Call from analog phone via TDM -> E1 asterisk A -> SIP or IAX2
-> asterisk B (musiconhold)
testing codecs between asterisk A and B: alaw, g726
Jitter simulation:
asterisk B:
modprobe sch_netem
tc qdisc add dev eth0 root netem delay 0ms 0ms
tc qdisc change dev eth0 root netem delay 0ms 300ms
Results:
module reload codec_alaw shows that PLC is true.
Jitter is working for both IAX2 and SIP channels but without PLC.
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)
I've tried jitter buffer patch for 1.2 asterisk (from backports) and
SIP PLC is not working too.
Sample audio:
Random Jitter (0-300ms) and correct PLC http://www.lam.cz/iax2.wav
Random Jitter (0-300ms) without PLC http://www.lam.cz/sip.wav
So my question is: do i have something wrong or it is bug? Any
suggestion will be appreciative.
Festr
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--
Martin Vít
LAM plus s.r.o.
http://www.vasesit.cz/
mobil: 605 267 610
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