Question

G.711 a/u-law Vs G.729 a/ab

Asked by: mysticaljoey

Hi,

i'm deciding on using G.711 a/u-law or G.729 a/ab on my voip system, if you are to choose 1 only, which one will you choose? and why?

regards

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Asked On
2006-02-09 at 04:44:46ID21729978
Tags

vs

Topic

Voice Over IP

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Answers

 

by: grbladesPosted on 2006-02-09 at 06:15:28ID: 15912713

G711 is bandwidth is not an issue as it does not compress the sound as much and is higher quality.

 

by: mysticaljoeyPosted on 2006-02-09 at 18:33:32ID: 15918986

but most carrier does not support g711 as their primary codec is that true?

 

by: grbladesPosted on 2006-02-10 at 01:11:37ID: 15920515

I would have expected most to support it.

However if you are sending the calls over the internet then a higher compression codec would usually be better unless you have lots of bandwidth available or will only have a small number of simultaneous calls.

 

by: catranisPosted on 2006-02-13 at 07:26:10ID: 15941942

i would make this decision based on your hardware and bandwidth resources. If you are on an IP connection that is congested your probably better off with G729. But you need to make sure you have enough processor power on your IP-PBX as well. If that's what your using. G711ulaw is uncompressed voice. It's basically a digital version of analog voice and it uses more bandwidth then a standard phone call. Including transport overhead and signalling your looking at about 80 kbps per g711 channel. g729 will be closer to 20 kbps.

 

by: grbladesPosted on 2006-02-13 at 07:55:35ID: 15942197

Technically G711ulaw is compressed. Normal voice is 8khz 12 bit. ulaw uses logrithmic (spelling?) compression of the amplitude down to 8 bits.
This reduces the amount of data required as catranis said above and means it can normally still represent DTMF tones and low speed modem data calls (faxes etc...).

 

by: catranisPosted on 2006-02-13 at 08:41:20ID: 15942676

Per Wikipedia

"The basic digital circuit in the PSTN is a 64-kilobits-per-second channel, originally designed by Bell Labs, called a "DS0" or Digital Signal 0. To carry a typical phone call from a calling party to a called party, the audio sound is digitized at an 8 kHz sample rate using 8-bit pulse code modulation. The call is then transmitted from one end to another via telephone exchanges. The call is switched using a signalling protocol (SS7) between the telephone exchanges under an overall routing strategy."


grblades is probably referncing POTS characteristics.

 

by: grbladesPosted on 2006-02-13 at 10:03:16ID: 15943410

Yes I was referencing more as to how the digital handsets work rather than an official standard as what quality the handset manufacturer digitises the voice before encoding is purely up to them.
The main point is that although DS0 and ulaw are both 8bit 8khz sampling with a 64kbit bandwidth, DS0 uses standard digital encoding while ulaw encodes the ampliture in a logrithmic way. This improves the quality of quiet sounds but reduces the accuracy of loud ones so the overall quality is slightly better. It does however mean that it stops modems and faxes working properly.

It may seem like a minor point but I thought I would mention it as otherwise some people buy an analogue phone adapter and configure it to use ulaw and think that they will be able to use it  to link their modems to the new phone system. Even faxes running at 14.4 with ECM enabled can have problems.

 

by: JgearingerPosted on 2006-02-13 at 13:44:45ID: 15945449

Sorry to hop in on this so late boys,

To start here are the basics:
G.711a is a low compression codec that transmits at 64kbps with low complexity and an average MOS of 4.5

G.729a is a low to medium compression codec that transmits at 28kbps medium complexity and an average MOS score of 4
    The bigest problem with G.729a is the transmition of the DTMF tone.  One in every 12 tones will be misinterpreted on average.  You can avoid this buy either using AVT or another DTMF relay ??RFC2388??? that transmits the tone at the FXO.

If you are using this as a PBX install with out and remote offices it makes complete sense to use g.711a, a standard office's layer 2 switch will be able to carry 100Mbs which is more than enough to handle 1428 concurent calls if the PBX proccessor/liscense can handle that.  If there are remote offices you are connecting I would go with the lower bandwidth codect due to the excessive cost of bandwidth on secure connections.  

If it is on an uplink to a VoIP trunk from a carrier I would contact the carrier to see if they supprt the DTMF relay.  If not I would deffinately use G.711a because this will carry the most VoIP traffic.  

 

by: JgearingerPosted on 2006-02-13 at 13:50:07ID: 15945512

Comment from mysticaljoey
Date: 02/09/2006 06:33PM PST
 Author Comment  

but most carrier does not support g711 as their primary codec is that true?


The gateway determines the what codec will be used.  Most carriers do not terminate thier own calls.  Most have multible agreements.  and each gateway can define which codecs it will accept/prefer.  Level 3 supports multible codecs but sends them out in a specific order.:  G.711a, G.729a, G.723.1.  The FXS depending on make will ahndle this differently.  You can always make a call in your prefered codec but, only cisco as far as i know can control how it recieves the codec.  Also remember if you limit yourself to one codec and call on NET and the other PBX uses a different codec "only" you will get a fast busy.

 

by: grbladesPosted on 2006-03-09 at 09:54:28ID: 16146975

I recomend a splt between the 3 of us who answered as we all provided different information.

20120131-EE-VQP-002

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