Displaying 20 results from an estimated 5000 matches similar to: "ADPCM - vs - G.726"
2004 Apr 05
2
ADPCM 4-bit, 6 kHz
I found some posts regarding this issue dating of September 2003, but no
real answer.
The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I
need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help
migration.
Is there an existing format/codec for this? If not, can I make myself a
shared object in /usr/lib/asterisk/modules? Is this easy??? :-(
Thanks,
Yves
2005 Jan 17
2
iaxtel - -- Format for call is ADPCM
When I try to call iaxtel it goes to codec ADPCM even though I have
define in iax.conf gsm
Call accepted by 69.73.19.178 (format ADPCM)
-- Format for call is ADPCM
My settings:
[general]
port=4569
register => xxxx:xxxx@iaxtel.com
bandwidth=high
jitterbuffer=no
tos=lowdelay
[voipjet]
type=peer
host= xxx.xxx.xxx.xx
secret= xxx
auth=md5
notransfer=yes
context=incoming
disallow=all ;
2008 Mar 27
1
ADPCM codec and IAXy device
Hi All;
I need to buy one IAXy device, but I discovered that
it supports only g711 and ADPCM codec, so I was wonder
that it does not support g729 or GSM?!
Anyway, what is that ADPCM and how much it consumes
bandwitdh? Also, asterisk support such codec? What its
name in the configuration?
Any advise?
Regards
Bilal
2003 Sep 16
3
Adpcm, 6KHz codec
Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get
this codec?
Thank you.
Alex Zarubin
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2007 Jun 19
2
RTP/RTSP streaming of GSM or ADPCM audio
Greetings:
It would be nice if Icecast supported RTSP; however I would
appreciate any suggestions for a small RTSP/RTP solution to
encode 8kHz mono audio in GSM or ADPCM and service multiple
unicast client connections. The ideal would be a black-box
hardware solution with an audio input and ethernet interface
similar to broadcast studio IP audio links or the network
audio capabilities of certain
2004 Nov 29
1
IAXy and ADPCM codec problem
Hi everyone,
I'm using the IAXy boxes and i'm having some trouble when I use it with
the ADPCM codec.
When I use the ADPCM codec only one person (out of the two of the
conversation) is able to hear the other, but when I switch to ULAW codec
everybody can hear the other.
The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds
choppy, the ADPCM codec sounds good but only
2007 Jun 19
1
RTP/RTSP streaming of GSM or ADPCM audio
Thomas B. Ruecker wrote:
> Michael Grigoni wrote:
>
>>Greetings:
>>
>>It would be nice if Icecast supported RTSP;
>
> It probably never will
>
>>however I would
>>appreciate any suggestions for a small RTSP/RTP solution to
>>encode 8kHz mono audio in GSM or ADPCM and service multiple
>>unicast client connections.
>
> why not use
2004 Sep 10
2
Re: Lossless AMI ADPCM
I'm copying the flac-dev list to see if anyone has any
feedback also...
--- Juhana Sadeharju <kouhia@nic.funet.fi> wrote:
> Hello again. I had time to check the paper out. I have filled the
> steps given in the paper with formulae, and then written a piece of
> C code. It is not complete code, but could be a reasonable start.
> Maybe there is one typo in the paper -- I have
2005 Sep 28
1
Asterisk sound files, audio bandwidth, and sound quality
Hello, everybody:
I'm developing an application using Asterisk and a TDM-400 card.
I understand the concept of the difference between GSM and WAV files
when using Asterisk, but I'm not happy with the sound quality with the
GSM compression. It's merely *acceptable* for a telephone call, but for
anything else, it leaves something to be desired.
Case in point -- if you compare the
2004 Mar 30
1
G726 not working ?
Hi,
I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of
this morning 3/30/04 of asterisk, zap and libpri.
The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced".
When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I
can see:
[format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
==
2004 Jan 26
0
ADPCM support with RECORD FILE
I want to record audio in ADPCM format. According to the "show codecs"
output of Asterisk, it looks like it supports adpcm. But I do not know what
to tell the "RECORD FILE" directive in my AGI script.
The RECORD FILE command usually has this form:
RECORD FILE <filename> <format> <timeout> [BEEP]
It records fine in WAV or GSM if I enter "wav" or
2004 Nov 29
0
Subject: IAXy and ADPCM codec problem.
Hi everyone,
I'm using the IAXy boxes and i'm having some trouble when I use it with
the ADPCM codec.
When I use the ADPCM codec only one person (out of the two of the
conversation) is able to hear the other, but when I switch to ULAW codec
everybody can hear the other.
The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds
choppy, the ADPCM codec sounds good but only
2003 Sep 18
2
Adpcm quality
Please, try
exten => 99,1,Wait,1
exten => 99,2,Record,/tmp/pcmfile:pcm
exten => 99,3,Wait,1
exten => 99,4,Playback,/tmp/pcmfile
exten => 99,5,Wait,1
exten => 99,6,Record,/tmp/voxfile:vox
exten => 99,7,Wait,1
exten => 99,8,Playback,/tmp/voxfile
(put your own extension).
Pcm recording is OK, playback is OK.
Adpcm recording is noticeably worse. Adpcm playback is very
2009 Dec 30
2
Skype for Asterisk
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is working fine.
case 2: This skype account (rexesbposolutions) has been assigned with a
online virtual
2004 Sep 10
0
Re: Lossless AMI ADPCM
>From: Josh Coalson <j_coalson@yahoo.com>
>
>I'm copying the flac-dev list to see if anyone has any
>feedback also...
I'm supposed to be there myself since yesterday but have not got
the first digest yet.
>First, the results they show are for compression of data
>that has already been lossily quantized to fewer bits per
>sample, e.g. u-Law and A-Law are
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
2012 Nov 18
1
[LLVMdev] Basic Block Frequency counting in LLVM 2.9
Dear All,
I'm using LLVM2.9 for profiling basic block frequency.
I'm using following commands.
rdpatel55 at ubuntu:~$ llvm-gcc -emit-llvm -O0 -c -o adpcm.bc adpcm.c
rdpatel55 at ubuntu:~$ llvm-gcc -emit-llvm -O0 -c -o rawcaudio.bc rawcaudio.c
rdpatel55 at ubuntu:~$ llvm-link -o main.bc rawcaudio.bc adpcm.bc
rdpatel55 at ubuntu:~$ opt -q -f -insert-edge-profiling -o main.inst main.bc
2004 Dec 12
1
patton smartnode integration
Any have any success using a patton smartnode 4118/js/eiu fxs gateway
with asterisk? We we're able to get the unit to register with
asterisk, but when trying to place a call, no codec was compatible,
even though I had all of the following enabled on the patton ...
# G.711 A-Law/?-Law (64kbps)
# G.726 (ADPCM 40, 32, 24, 16 kpbs)
# G.723.1 (5.3 or 6.3 kbps)
# G.729ab (8kbps)
the link to this
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09, Marcelo Terres wrote:
> I suppose that you enable the video support on sip.conf, right?
>
> Regards,
> Marcelo H. Terres <mhterres at gmail.com>
> IM: mhterres at jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
>