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ITU G.711

G.711 is a high bit rate (64 Kbps) ITU standard codec. It is the native language of the modern digital telephone network.

Although formally standardised in 1988, the G.711 PCM codec is the granddaddy of digital telephony. Invented by Bell Systems and introduced in the early 70's, the T1 digital trunk employed an 8-bit uncompressed Pulse Code Modulation encoding scheme with a sample rate of 8000 samples per second. This allowed for a (theoretical) maximum voice bandwith of 4000 Hz. A T1 trunk carries 24 digital PCM channels multiplexed together. The improved European E1 standard carries 30 channels.

There are two versions: A-law and U-law. U-law is indigenous to the T1 standard used in North America and Japan. The A-law is indigenous to the E1 standard used in the rest of the world. The difference is in the method the analog signal being sampled. In both schemes, the signal is not sampled linearly, but in a logarithmic fashion. A-law provides more dynamic range as opposed to U-law. The result is a less 'fuzzy' sound as sampling artifacts are better supressed.

Using G.711 for VoIP will give the best voice quality; since it uses no compression and it is the same codec used by the PSTN network and ISDN lines, it sounds just like using a regular or ISDN phone. It also has the lowest latency (lag) because there is no need for compression, which costs processing power. The downside is that it takes more bandwidth then other codecs, up to 84 Kbps including all TCP/IP overhead. However, with increasing broadband bandwith, this should not be a problem.

G.711 is supported by most VoIP providers.

ITU G series reccomendations

Created by jht2, Last modification by Eric Caron on Thu 05 of Jun, 2008 [14:42 UTC]

Comments Filter

Re: Packet dynamics of g.711

by edokter on Sunday 31 of July, 2005 [12:41:59 UTC]
G.711 sends packets with 160 bytes/samples, containing 20 ms of speech. The difference you see is not related to G.711, but related with the "Tranmit Silence" option that can be enabled by the sending party, no matter what codec is being used. This option can be found in any IP phone, soft phone or SIP gatway device.

Edit

Packet dynamics of g.711

by Anonymous on Monday 22 of November, 2004 [22:18:20 UTC]
I am a Quality Assurance Associate who is currently involved in developing a Voip gateway. I have been helping develop our gateway for some time now and have done some investigation into the 'packet dynamics' that I can see using a packet sniffer of our product verses our competitions product.

Our product uses G.711 as its primary codec. Looking at the captured data, I have noted some differences in our G.711 and our competitor’s codec (Vonage). I will just post the numbers and describe the behavior in the hopes that someone more savvy to this will know why the differences occur.

Our product will use the following no matter what occurs:
Transmit packet size: 214 bytes
Receive packet size : 214 bytes
Transmit rate : ~ 100 packets / second

Our Competitions product has variable packet sizes, and transmission rates, depending if there is sound present on the upstream, downstream, of if its silent:

Silent (on both ends):
Transmit packet size: 214 bytes
Receive packet size : 214 bytes
Transmit rate : ~ 100 packets / second

Send sound from Voip to PSTN:
Transmit packet size: 60 bytes
Receive packet size : 214 bytes
Transmit rate : ~ 100 packets / second

Receive sound from PSTN to Voip:
Transmit packet size: 214 bytes
Receive packet size : 60 bytes
Transmit rate : ~ 70 packets / second

The results show that Vonage is certainly more bandwidth conservative then ours, since their variable bit rate allows for lower overall throughput. Is this a product of their backend servers, gateways firmware, or a proprietary codec (like g.711)?

Any insight here would be appreciated.

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