login | register
Fri 04 of Jul, 2008 [20:12 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.
 
Google Ads
Last Changes
Shoutbox
  • Samuel, Thu 03 of Jul, 2008 [13:41 UTC]: ok thank you
  • Mats Karlsson, Thu 03 of Jul, 2008 [13:37 UTC]: Nice Samuel, will look forward to rad it.
  • bwl_fernstudent, Thu 03 of Jul, 2008 [09:08 UTC]: Your blog shows some usefull code
  • Samuel, Thu 03 of Jul, 2008 [08:04 UTC]: I'll translate it, for sure
  • Mats Karlsson, Wed 02 of Jul, 2008 [20:46 UTC]: LOL, in french! Translate it to English and I will read it.
  • Samuel, Wed 02 of Jul, 2008 [08:07 UTC]: Hello, i wrote a blog about Asterisk, speaking about installation,programming and more http://sambranche.blogspot.com/
  • Nick Barnes, Tue 01 of Jul, 2008 [17:46 UTC]: Steve - Asterisk doesn't 'fit into linux' - it's an application which runs on top of Linux.
  • Steve, Mon 30 of Jun, 2008 [18:07 UTC]: anyone know where I can find a block diagram of how asterisk fits into linux. my f'ing bosses want me to draw something up.. ugh.
  • akbar, Fri 27 of Jun, 2008 [10:37 UTC]: marley_boyz@yahoo.com how to configure call forward, call back, call pick up using TDM and asterisk 1.2.13... please help me.. thx...
  • Matthew Williams, Tue 24 of Jun, 2008 [22:37 UTC]: We are looking for Tier II VoIP Support Technicians in St Louis. Send resumes to mwilliams AT voxitas DOT com.
Server Stats
  • Execution time: 0.91s
  • Memory usage: 2.63MB
  • Database queries: 30
  • GZIP: Disabled
  • Server load: 0.62

Asterisk codecs

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

    • show codecs Screen output
The 'show codecs' command is deprecated and will be removed in a future release. Please use 'core show codecs' instead.

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)



To tell which codec is being used for a specific call use one of the following CLI commands:

 sip show channels
 iax2 show channels

To use with allow and disallow, here is the association table:

 G.711 ulaw = ยต-law (US standard)
 G.711 alaw = A-law (European standard)
 G.722 = g722 (don't confuse this with g722.1 or g722.2)
 G.723.1 = g723.1 (pass-thru only)
 G.726 = g726
 G.729 = g729
 GSM = gsm
 iLBC = ilbc
 LPC10 = lpc10
 Speex = speex
 ADPCM = adpcm

A typical use might be:

 disallow=all
 allow=alaw
 allow=ulaw


File name extensions

Extensions for various encoded files in Asterisk
  • wav:
  • pcm:
  • gsm:

Packetization

Various clients support variable sample periods / packetization. Asterisk 1.2 and earlier only supports 20ms packetization in RTP-based protocols like SIP and MGCP, so you should configure your client to use this. However, iLBC with its 30 ms packets also works with Asterisk 1.2. 1.4 and later include support for variable packetization, either settable in the config or set automatically according to the SDP.

See also:


Created by flavour, Last modification by KieranMullen on Thu 05 of Jun, 2008 [15:31 UTC]

Comments

Comments Filter

variables for codecs?

by Dirk-Michael Brosig on Thursday 14 of December, 2006 [09:55:37 UTC]
If there variables for the used audio and video codecs for use in extension.conf? P.e. ...

exten => 100,n,GotoIf($"${CODEC_VIDEO}" = "h.264"?200:300) // play h.264 video otherwise play h.263
exten => 100,200,Playback(video264)
exten => 100,300,Playback(video263)

Packetization

by Sergey Basmanov on Wednesday 16 of August, 2006 [11:07:14 UTC]
Beta version of packetization code is available here: http://bugs.digium.com/view.php?id=5162
With this patch, packetization can be set for each codec for SIP peer/user, as well as match to remote packetization values if present in SDP.

What CODEC am I using?

by omar on Monday 24 of October, 2005 [15:27:35 UTC]
From my voice provider I have to choose a codec g.726, and I need to know what g.726 codec I am using.
How can I know if I am using
g.726 - 40
g.726 - 32
g.726 - 16
etc.
or ASTERISK only support g.726 - 32

Re: Quality Voice

by chammoud on Friday 15 of April, 2005 [09:54:23 UTC]
it could be due to many factors. Internet bandwidth one of them. if you are using iax test different codecs and see how that cahnge the quality of the sound. also in iax.conf try a call with jitterbuffer=yes and no or comment it. whaen you make a call perform this command: iax2 show netstats and write down your results.
be a member os asterisk-lists and post your problem with much more inputs so people can help you

Quality Voice

by hameds on Tuesday 08 of March, 2005 [14:21:31 UTC]
(:razz:)

Hi... i need your help .... the quality voice from my asterisk id very low how to change this!!!

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2008 VOIP-Info.org LLC

Powered by bitweaver