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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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Codecs

Codecs are used to convert an analog voice signal to digitally encoded version. Codecs vary in the sound quality, the bandwidth required, the computational requirements, etc.

Each service, program, phone, gateway, etc typically supports several different codecs, and when talking to each other, negotiate which codec they will use.

As an example, a Cisco ATA-186 supports these codecs:
  G.723.1, G.711a, G.711u, G.729a
As an example, a Cisco 7960 supports (Firmware P0S3-06-0-00):
  G.711a, G.711u, G.729a

Some codecs require payment of royalities for their use in a product or program. See Codec Patents for more information.


Some tables comparing different codecs:

Tables showing which VoIP clients support which codecs

Audio Examples of different Codecs

News

  • 2004-12-07 - SPIRIT Offers Ip-Multi Rate - Next Generation VoIP Codec; Dedicated IP-Multi Rate Codec Provides FM-Radio Speech Quality in IP Networks Businesswire
  • 2005-05-09 - Free online codec conversion tool available at asteriskguru.com.

Some potentially useful info:

Keep in mind the link layer framing that is used as well. ATM (what most of the internet backbones use) has 53 byte fixed cells. There are 5 bytes of headers. Only one packet can exist in a group of cells (a group is 1 or more cell until the whole packet is sent). This means that if you try to send a 80 byte IP packet over an ATM link it will be chopped into 2 ATM cells with 16 bytes of padding. This is only 83% efficient. By adjusing your sample size you may find that your throughput can be increased because you transfer more useful data and less padding. Make your sample size too small and you have a lot of IP overhead, make it too large and it can cause problems with call quality (think of a 30ms jitter buffer and 30ms sample sizes, in effect you have no jitter buffer because packets cant be reordered, jitter cant be controlled, etc). Its a fine balance, but something to consider. Even if you dont use ATM the internet backbones often do, so this is something that may make slightly faster transfers and better network efficiency. DSL, E1, T1, SMDS, OC1, OC3, OC12 etc generally all use ATM, so its quite common.


See also

Created by admin, Last modification by wasim on Sun 17 of Jun, 2007 [06:14 UTC]

Comments

Comments Filter

I am so new to Linux its not even funny

by David on Friday 15 of December, 2006 [16:00:06 UTC]
First of all this is a great location for information but I am LOST. I am looking for instructions on installing codecs on the Asterisk PBX I some how was able to install without a problem. I want to run GSM and G729 codecs and I understand that G729 has issues for licensing. I got a site from these pages with a list of the codes but I cannot (GET) them. Here is a list of what I need

1. How to download, install and configure the system to see the codecs.


I am down to a hand full of hair on my head, if some one does not rescue me I might start pulling other peoples hair out !! :-)

Thanks
David

Which Codec for which network?

by Ray on Friday 09 of December, 2005 [16:22:35 UTC]
I have seen a lot of work about the characterisation of Codecs in terms of bandwidth utilisation, but has there any work been done on the suitability of different codecs based on network SLA parameters. For example, bandwidth on a broadband Internet connection is essentially free up to ~512K per second, so bandwidth efficiency of one call in a truly distributed switched environment is probably not at all interesting for an end user. However, latency is more likely to be a problem.... everyone says "you have to be careful," but I have not seen any recommendations except that end to end latency should not exceed 250mS. That is not particularly useful advice.

I'd like to see a table or graph of different codec performance versus Mean Opinion Score (MOS) score for different values of:
minimum link bandwidth
network latency
jitter
packet loss

Then you could predict whether it is interesting to run voip to a particular location on a quality versus the cost decision.

Anyone?

by Ray on Friday 09 of December, 2005 [15:51:26 UTC]

by Ray on Friday 09 of December, 2005 [15:48:29 UTC]

What CODEC

by omar on Monday 24 of October, 2005 [15:28:38 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
Edit

Codec support by VoIP clients

by Anonymous on Monday 15 of November, 2004 [05:53:47 UTC]
An attempt to show which soft-phones and hardware support which codecs (additions to the table are welcome): http://compare.ozvoip.com/codecsupport.php

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

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