[asterisk-dev] Jitter Buffer and PLC

[Martin Vít]

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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