Podrška #14586
Zatvoren
asterisk codecs
100%
Povezani tiketi 3 (0 otvoreno — 3 zatvorenih)
Izmjenjeno od Ernad Husremović prije skoro 17 godina
izgleda da je za lowbandwidth najbolje g.729 codec, ali on nije free
Installing the G.729 Codec for Asterisk
The G.279 codec is used to reduce the bandwidth necessary to process voice calls. Instead of 64Kbps of data for a voice call, G.729 stuffs the call into 8Kbps. What MP3 did for music, G.729 does for voice calls. To install the G.729 codec, you first need to download the version that matches the processor in your Asterisk box. There are codecs available for both Linux and FreeBSD systems here. You’ll also need to download the registration utility. If you’re using Asterisk@Home, you’ll need the glibc_2.3 utility available here. If you don’t know what version of glibc is running on your Asterisk server, go to a command prompt and type ldd –version. Note: There should be two dashes before the word “version.” Now that you’ve downloaded codec_g729a.so, you’ll need to copy it to /usr/lib/asterisk/modules on your Asterisk server while logged in as root. Next, copy the register program to any convenient place on your Asterisk server, e.g. /tmp will do. Modify the permissions for the register program so that it is executable: chmod a+x register. Now pay your $10 and wait for your registration key to be emailed to you. When you get the key, go to your Asterisk server and issue the following command from the directory where you placed the register program: ./register G729-1234ABCD substituting your actual key for G729-1234ABCD. Your Asterisk server must have Internet access to complete the registration process. Once you get a message that the registration was successful, restart Asterisk, and you’re in business: amportal stop then amportal start. Finally, note that the G.729 registration is locked to the MAC addresses of the network cards in your Asterisk server. If you change NICs, you’ll need to reregister the G.729 codec. You get two bites at the apple without contacting Digium for a new code.
Izmjenjeno od Ernad Husremović prije skoro 17 godina
interesantno, ali siemens izgleda nema gsm codec ?
kada na siemensu tražim optimu postavke codec-a za lowbandwidth, dobijem ovaj poredak- g.729
- g.726
- g.711 alaw
- g.711 ulaw
- g.722
znači seljedeći je g.726 u smislu malog bandwidth-a, da vidimo je li on free
Izmjenjeno od Ernad Husremović prije skoro 17 godina
Asterisk Codecs¶
Asterisk supports the following narrow-band and wideband (HD audio) codecs:- G.711 ulaw (as used in US)
- G.711 alaw (as used in Europe)
- G.722 - 16 kHz wideband codec; passthrough, playback and recording in Asterisk 1.4; full support incl. transcoding in Asterisk 1.6, a backport for 1.4 is available, or use this possibly more up-to-date patch
- G.723.1 - pass-thru for people who need a license , free for other people
- G.726 - 32kbps in Asterisk 1.0.3, 16/24/32/40kbps in CVS HEAD; flawed until Asterisk 1.4 which corrected the implementation and introduced g726aal2 for backwards compatibility with Asterisk 1.2 installations
- G.729 - may require a license unless using pass-thru, free version available for use in countries without patents or for educational use only
- GSM
- iLBC
- LPC10 (not recommended!)
- Speex - configurable 4-48kbps, VBR, ABR, etc. see bug 2536. For Asterisk 1.4. there is patch 10519 available that adds wideband support for the OpenWengo software client
Use this commands in the Asterisk CLI for a detailed listing of the actual capabilities:
show codecs ** show translation show translation recalc 10
Izmjenjeno od Ernad Husremović prije skoro 17 godina
šta kaže naš ifold
ifold*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 DESC
--------------------------------------------------------------------------------
1 (1 << 0) (0x1) audio g723 (G.723.1)
2 (1 << 1) (0x2) audio gsm (GSM)
4 (1 << 2) (0x4) audio ulaw (G.711 u-law)
8 (1 << 3) (0x8) audio alaw (G.711 A-law)
16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2)
32 (1 << 5) (0x20) audio adpcm (ADPCM)
64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM)
128 (1 << 7) (0x80) audio lpc10 (LPC10)
256 (1 << 8) (0x100) audio g729 (G.729A)
512 (1 << 9) (0x200) audio speex (SpeeX)
1024 (1 << 10) (0x400) audio ilbc (iLBC)
2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)
4096 (1 << 12) (0x1000) audio g722 (G722)
65536 (1 << 16) (0x10000) image jpeg (JPEG image)
131072 (1 << 17) (0x20000) image png (PNG image)
262144 (1 << 18) (0x40000) video h261 (H.261 Video)
524288 (1 << 19) (0x80000) video h263 (H.263 Video)
1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)
Izmjenjeno od Ernad Husremović prije skoro 17 godina
- 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)
Izmjenjeno od Ernad Husremović prije skoro 17 godina
a zašto ne bi probali g729 ???
asterisk kaže da je i on dostupan:
256 (1 << 8) (0x100) audio g729 (G.729A)
znači hajmo probati prvo g729 pa onda g726
Izmjenjeno od Ernad Husremović prije skoro 17 godina
How to use the G.729 codec in pass-thru mode in Asterisk
"Pass-thru" means that if you were, for example, using two phones which both have inbuilt support for g729 codec. You can let them talk to each other in g729 format, without asterisk having to transcode (which requires a license)
G.729 may require a license per channel unless it is used in pass-thru mode.
If no license is required, you can download the Open Source implementation or binaries for Linux and FreeBSD.
Contrary to what was on this page before, access to Voicemail requires only a few configuration tweaks in pass-thru mode:- set format=g729 (and only format=g729) in voicemail.conf
- set maxsilence=0 in voicemail.conf
Detecting silence in Asterisk means that you need access to the stream in slin, which requires transcoding. This setting will disable that detection.
In order to ensure that it is used in this mode, pay attention to the configuration:
no T,t or monitor in the dialplan, application Dial
Configuration for the SIP channel
sip.conf needs the following:
[general] disallow=all allow=g729 allow=ulaw allow=alaw
Izmjenjeno od Ernad Husremović prije skoro 17 godina
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
Credits Intel originally offered a sample G.729 and G.723.1 implementation for Linux. Daniel Pocock ( daniel at readytechnology.co.uk ) converted Intel's code into an Asterisk module. Arkadi Shishlov and others supplied compiler optimizations, and bug fixes. Support this project Would you like to see this project continue? Click here to read some important information about our funding. Introduction This code let's Asterisk use the G.729 and G.723.1 protocols for voice compression when communicating with other devices. The code produces a Asterisk modules, codec_g729.so and codec_g723.so, that you put in your Asterisk modules directory. The code is provided as a patch which will convert Intel's sample application into an Asterisk codec module.
Izmjenjeno od Ernad Husremović prije skoro 17 godina
G.729 and G.723.1 codecs x86 (and x86_64) Linux and FreeBSD binaries for Asterisk open source PBX
DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm.
Supportthe project
- Sources
- Binaries
- Notes and Troubleshooting
- Getting help
Sources
To compile the codecs you need Intel IPP libraries installed. Currently only Asterisk 1.4, 1.6 and TRUNK are supported. Support for Asterisk 1.2 and Callweaver is planned, for now use the binaries. Use "g723 debug" and "g729 debug" commands to print statistics about received frame sizes, can aid in debugging audio problems. You need to bump Asterisk verbosity level to 3 to see the numbers.- asterisk-g72x-1.0-beta5.tar.bz2
- choose codec binary appropriate for your Asterisk version and CPU type, use x86_64 for 64-bit mode, scroll to the end of the list for FreeBSD binaries
- delete old codec_g729/723*.so files (if any) from /usr/lib/asterisk/modules directory
- copy new codec_g729/723*.so files into /usr/lib/asterisk/modules directory
- restart Asterisk
- check the codec is loaded with 'core show translation recalc 10' on Asterisk console ('show translation' in Asterisk 1.2)
- G.723.1 send rate is configured in Asterisk codecs.conf file (Linux Asterisk 1.2, 1.4, 1.6, TRUNK and Callweaver, FreeBSD 7.x Asterisk 1.4 binaries only):
[g723] ; 6.3Kbps stream, default sendrate=63 ; 5.3Kbps ;sendrate=53
This option is for outgoing voice stream only. It does not affect incoming stream that should be decoded automatically whatever the bitrate is.
Izmjenjeno od Ernad Husremović prije skoro 17 godina
Izmjenjeno od Ernad Husremović prije skoro 17 godina
root@ifold:~# ls /usr/lib/asterisk/modules/codec*
/usr/lib/asterisk/modules/codec_adpcm.so /usr/lib/asterisk/modules/codec_alaw.so /usr/lib/asterisk/modules/codec_a_mu.so /usr/lib/asterisk/modules/codec_g726.so /usr/lib/asterisk/modules/codec_gsm.so /usr/lib/asterisk/modules/codec_lpc10.so /usr/lib/asterisk/modules/codec_ulaw.so
Izmjenjeno od Ernad Husremović prije skoro 17 godina
Izmjenjeno od Ernad Husremović prije skoro 17 godina
ja mene budale ovo je g723 codec :(
Izmjenjeno od Ernad Husremović prije skoro 17 godina
Izmjenjeno od Ernad Husremović prije skoro 17 godina
pokušaću ovo na ifold-u:
http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-pentium4.so
evo kako izgleda stanje codec-a prije instalacije:
ifold*CLI> core show translation recalc 10
Recalculating Codec Translation (number of sample seconds: 10)
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 9 - - - 2 -
ulaw - 4 - 1 2 2 1 9 - - - 2 -
alaw - 4 1 - 2 2 1 9 - - - 2 -
g726aal2 - 4 2 2 - 2 1 9 - - - 1 -
adpcm - 4 2 2 2 - 1 9 - - - 2 -
slin - 3 1 1 1 1 - 8 - - - 1 -
lpc10 - 5 3 3 3 3 2 - - - - 3 -
g729 - - - - - - - - - - - - - <<<<<<<<<<<<<<<<<<<<<< g729 trenutno supica tach
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - 4 2 2 1 2 1 9 - - - - -
g722 - - - - - - - - - - - - -
Izmjenjeno od Ernad Husremović prije skoro 17 godina
root@ifold:~# wget http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-pentium4.so
--15:39:53-- http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-pentium4.so
=> `codec_g729-ast14-gcc4-glibc-pentium4.so'
Resolving asterisk.hosting.lv... 213.21.217.135
Connecting to asterisk.hosting.lv|213.21.217.135|:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: 397,332 (388K) [application/octet-stream]
100%[====================================>] 397,332 232.53K/s
15:39:54 (231.91 KB/s) - `codec_g729-ast14-gcc4-glibc-pentium4.so' saved [397332/397332]
root@ifold:~# cp codec_g729-ast14-gcc4-glibc-pentium4.so /usr/lib/asterisk/modules/codec_g729.so
Izmjenjeno od Ernad Husremović prije skoro 17 godina
ifold*CLI> core show translation recalc 10
ifold*CLI>
Recalculating Codec Translation (number of sample seconds: 10)
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 9 17 - - 2 -
ulaw - 4 - 1 2 2 1 9 17 - - 2 -
alaw - 4 1 - 2 2 1 9 17 - - 2 -
g726aal2 - 4 2 2 - 2 1 9 17 - - 1 -
adpcm - 4 2 2 2 - 1 9 17 - - 2 -
slin - 3 1 1 1 1 - 8 16 - - 1 -
lpc10 - 5 3 3 3 3 2 - 18 - - 3 -
g729 - 6 4 4 4 4 3 11 - - - 4 - <<<<<<<<<<<<<<<< izgleda da ga je učitao
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - 4 2 2 1 2 1 9 17 - - - -
g722 - - - - - - - - - - - - -
Izmjenjeno od Saša Vranić prije skoro 17 godina
odradio i evo
root@rmlh-1:/usr/lib/asterisk/modules# ls -l codec_g* -rwxr-xr-x 1 root root 155495 2008-06-03 19:38 codec_g726.so -rwxrwxrwx 1 root root 2087480 2008-06-19 15:40 codec_g729.so -rwxr-xr-x 1 root root 637172 2008-06-03 19:38 codec_gsm.so
Izmjenjeno od Saša Vranić prije skoro 17 godina
evo sad i kod mene
rmlh-1*CLI> core show translation recalc 10
Recalculating Codec Translation (number of sample seconds: 10)
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 5 4 - - 2 -
ulaw - 2 - 1 2 2 1 5 4 - - 2 -
alaw - 2 1 - 2 2 1 5 4 - - 2 -
g726aal2 - 2 2 2 - 2 1 5 4 - - 1 -
adpcm - 2 2 2 2 - 1 5 4 - - 2 -
slin - 1 1 1 1 1 - 4 3 - - 1 -
lpc10 - 2 2 2 2 2 1 - 4 - - 2 -
g729 - 2 2 2 2 2 1 5 - - - 2 -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - 2 2 2 1 2 1 5 4 - - - -
g722 - - - - - - - - - - - - -
Izmjenjeno od Saša Vranić prije skoro 17 godina
podesio sam na vranici 51 ext. da koristi g729 codec, pa treba testirati
Izmjenjeno od Ernad Husremović prije više od 16 godina
- Status promijenjeno iz Dodijeljeno u Zatvoreno
- % završeno promijenjeno iz 0 u 100
ovome i QoS-u se baš treba posveititi, ali ovo što je urađeno do sada - urađeno.
nakon raznoraznih permutacija, kod vzeljke je instaliran grandstream video phone, i sad smo testirali officesa-vranici i vzeljka kaže da ima echo ali je puno bolja situacija nego li sa "tajvancem"