[asterisk-dev] Jitter Buffer and PLC

[Martin Vít]

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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