[asterisk-users] jitterbuffer and PLC

nakaji at 02.246.ne.jp nakaji at 02.246.ne.jp
Fri Jan 15 05:56:38 CST 2010


Hi, I have a question about jitterbuffer and PLC.

I use Asterisk 1.6.2.0 and 1.6.0.20 or older.
I use uLaw.

My system map:
=============================================================================

 [ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ]

=============================================================================

I use two asterisk server. 
'asterisk 2' do "Playback(some voice file)"
'asterisk 1' do "jitter".
'asterisk 1' and 'asterisk 2' has trunked by sip or iax2.

X-lite call 3003 to asterisk 1,
 and asterisk 1 call 3000 to asterisk 2.
 
On the map, 
 at "LOSS" I caused packet loss 5%, 
 and at "A" and "B" I captured packet.

I thought number of packets at "B" would be same number from "asterisk 2".
And I thought number of packets at "A" would be 95% from "asterisk 2".  

But the result is different!

At "A" and "B" is the same number as 95%.

No PLC ,No interpolations has made.
But on CLI,messages was "like that jitter and PLC work right" .

Like this:
=======================================================
-- Local/3000 at extd-651d;1 answered SIP/"id"-00000001
    -- fixed jitterbuffer created on channel Local/3000 at extd-651d;1
VvvvvvvvLvvvvvvvvvvvvvvvLvvvvLvvvvvv
vvvvvvvvvvvvvvvvvvvvvvLvvvvvvLvvvvvvvvvvvvvvvvvvv
vvvvvvvvvvLvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvLvvvv
vvvvLvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv


vvvvvvvvvvvvvLLvvvvvvLvvvvvvvvvvvvvvvvvvvvvvvvvvv
vvvvvvvvvvvvvvLvvvvvvvvvvvvvvvvvvLvvvvvvvvvLvvvvv
vvvvvvvvvvvvvLLLLLLLLLL    -- Hungup 'IAX2/"ip-address":4569-3645'
  == Spawn extension (extd, 3000, 2) exited non-zero on 'Local/3000 at extd-651d;2'
    -- fixed jitterbuffer destroyed on channel Local/3000 at extd-651d;1
  == Spawn extension (extd, 3003, 2) exited non-zero on 'SIP/"id"-00000001'
=======================================================
 

Why ?  why no interpolations ?
when sip trunk and iax trunk,same result.
 
I don't know how to do. So please help me.

How to do to work correct.
Or Asterisk has not yet have jitter and PLC ,hasn't it?



In 'asterisk 1' ,
==================================================================
 write on sip.conf =>  jbenable=yes  ,and ,  jbimpl=adaptive
 write on iax.conf =>  jitterbuffer=yes ,and, trunktimestamps=yes
 write on codecs.conf =>  genericplc => true
 and on extensions.conf

  when use sip trunk =>  
-------------------------------
exten => 3003,1,Dial(Local/3000 at extd/nj)
exten => 3000,1,Set(CALLERID(num)="some id")
exten => 3000,2,Dial(SIP/${EXTEN}@"some exten in sip.conf",120,T)
exten => 3000,3,Congestion
--------------------------------

  when use iax trunk =>  
-------------------------------
exten => 3003,1,Dial(Local/3000 at extd/nj)
exten => 3000,1,Set(CALLERID(num)="some id")
exten => 3000,2,Dial(IAX2/"id:pass"@"asterisk 2 ip-address"/${EXTEN},120,T)
exten => 3000,3,Congestion
--------------------------------

==================================================================


In 'asterisk 2' ,
==================================================================
write on extensions.conf
 =>
---------------------------
exten => 3000,1,Answer()
exten => 3000,2,Wait(1)
exten => 3000,3,Playback(aaa)
exten => 3000,4,Congestion
---------------------------

==================================================================






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