asterisk users - Oct 2007 - G.729 codec between avaya and asterisk

Dear all
   
          i have asterisk connected with avaya through E1 back-2-back now when i
configure my sip client with g.729 codec then i m not able to put call from
asterisk to avaya and when i user g.711 it is working fine so i dont know why i
need G.729 on E1 Trunk it is TDM technologies then why my call fail in g.729
case
   
   
  [sip_phone]------[asterisk]-----E1----[Avaya]----[analog_phone]
   
  Asterisk sip client configure with g.711 alaw/ulaw
  Avaya phone client configure g.711 alaw/ulaw
   
  suggest how do it implement g.729 on this case what change i have to done on
both part

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Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish
patel:
> Dear all
>  
>         i have asterisk connected with avaya through E1 back-2-back
> now when i configure my sip client with g.729 codec then i m not able
> to put call from asterisk to avaya and when i user g.711 it is working
> fine so i dont know why i need G.729 on E1 Trunk it is TDM
> technologies then why my call fail in g.729 case 
Hi Satish,

Neither do I know why you _need_ G.729. Are there any specific reasons
why you do not want to use G711 in the sip client, which is "working
fine"? (Nota bene: there are some more codecs supported by asterisk,
some of which may be also supported by your sip phone)

Your E1 trunk obviously is G711-only - this is to be expected, because
the G711 wave samples are those which go over the wire (as time-division
multiplexed bitstream).

Together with the information from
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing

namely
** G.729 requires a license per channel unless it is used
** in pass-thru mode.

which exactly matches your setup (by the way that was the first google
match for "g729 asterisk") we can guess that you did not buy the
license
which would be necessary for asterisk to transcode G729/G711.
> [sip_phone]------[asterisk]-----E1----[Avaya]----[analog_phone]
>  
> Asterisk sip client configure with g.711 alaw/ulaw
> Avaya phone client configure g.711 alaw/ulaw
>  
> suggest how do it implement g.729 on this case what change i have to
> done on both part
Avaya / E1 stays as is, sip client stays as is, your credit card data is
transferred to digium, and their license goes into the appropriate file
on your asterisk machine hard drive.

Others may have real world experience with those steps, but that is what
I read on this mailing list.

YMMV,
Anselm

there is no special requiremnt to use g.729 but day to day my sip client
incressing thats why some time i got breaking voice or voice quality not much
better i think in LAN there is lots of brodcat on lan

if i purches g.729 transcoder license for asterisk to convert g.729 to g.711
then  it will work or not

but why i need codec on trunk ????

Anselm Martin Hoffmeister <anselm at hoffmeister-online.de> wrote: Am
Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish
patel:
> Dear all
>  
>         i have asterisk connected with avaya through E1 back-2-back
> now when i configure my sip client with g.729 codec then i m not able
> to put call from asterisk to avaya and when i user g.711 it is working
> fine so i dont know why i need G.729 on E1 Trunk it is TDM
> technologies then why my call fail in g.729 case 
Hi Satish,

Neither do I know why you _need_ G.729. Are there any specific reasons
why you do not want to use G711 in the sip client, which is "working
fine"? (Nota bene: there are some more codecs supported by asterisk,
some of which may be also supported by your sip phone)

Your E1 trunk obviously is G711-only - this is to be expected, because
the G711 wave samples are those which go over the wire (as time-division
multiplexed bitstream).

Together with the information from
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing

namely
** G.729 requires a license per channel unless it is used
** in pass-thru mode.

which exactly matches your setup (by the way that was the first google
match for "g729 asterisk") we can guess that you did not buy the
license
which would be necessary for asterisk to transcode G729/G711.
> [sip_phone]------[asterisk]-----E1----[Avaya]----[analog_phone]
>  
> Asterisk sip client configure with g.711 alaw/ulaw
> Avaya phone client configure g.711 alaw/ulaw
>  
> suggest how do it implement g.729 on this case what change i have to
> done on both part
Avaya / E1 stays as is, sip client stays as is, your credit card data is
transferred to digium, and their license goes into the appropriate file
on your asterisk machine hard drive.

Others may have real world experience with those steps, but that is what
I read on this mailing list.

YMMV,
Anselm



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Am Dienstag, den 23.10.2007, 22:21 -0700 schrieb satish
patel:
> there is no special requiremnt to use g.729 but day to day my sip
> client incressing thats why some time i got breaking voice or voice
> quality not much better i think in LAN there is lots of brodcat on
> lan 
If your LAN is congested and a lot of single packet delay happens, you
should improve the LAN. You cannot run a LAN at 99% saturation with
VoIP, it will just not work, with packet drop rates and delays making
phone calls more of a earth-to-moon radio experience ("Houston *crackle*
*crackle* have *crackle* problem").

If _all_ that traffic is VoIP, G729 might help a bit, but I would not
expect it to get around all your bandwidth problems. Try to improve the
network first.

One interesting aspect of g729 might be that your sip client phones that
live behind a DSL line might profit from the smaller bandwidth
requirement on their side.
> if i purches g.729 transcoder license for asterisk to convert g.729 to
> g.711 then  it will work or not
I _think_ it will work (btw this is, as of some website I found, the
"main revenue stream" of Digium, so they will be interested in having
it
working). Others with real-world experience could tell you.
> but why i need codec on trunk ????
Codec stands for coding-decoding (or something similar). If you imagine
the "original signal" as voice and sound, meaning variations in air
pressure around the membrane of the telephone handpiece microphone, then
every digital representation is a kind of "coding". This even refers
to
8-bit-wave, which is the most obvious way of encoding: It merely writes
down the voltage level at the microphone input in the range -128 to +127
(IIRC, correct me if I am wrong). Accordingly, 16-bit wave has the
higher precision of -32768 to +32767.

G711 is - again, if I remember correctly - an adaptation of these bytes
to a logarithmic scales, bearing in mind the idea that small changes in
the higher ranges are treated differently from small changes in the
near-0-region. Something like the fiction bytestream value 0 1 2 3
representing the scale 0 4 6 7 of microphone values, instead of linear
data. Please research this yourself if you are interested in details.

G711 is the standard (and usually, the only available) codec for
ISDN/T1/E1... Europeans and US Americans established two different kinds
of G711 (?-law and a-law) which seem to be functionally similar.

BR
Anselm