Red Phone with Question Marks

What’s a G.711 Voice Codec and Why Should You Care?

Red-Phone-with-Question-MarksOne of the challenging things about writing about SIP trunking and unified communications is finding the right balance between providing too much technical information and not enough. While deploying and using SIP trunks is fairly straightforward and can be accomplished by someone with no telecommunications background, it is useful to have some technical context when selecting the service that is right for your business. You don’t have to be an expert, but knowing a little bit about the G.711 voice codec will help you make a more informed decision.

What is a Voice Codec?

A codec, which stands for coder-decoder, converts an audio signal (your voice) into compressed digital form for transmission (VoIP) and then back into an uncompressed audio signal for replay. It’s the secret sauce of VoIP.

Different codecs have different levels of compression. The highly compressed signals require less internet bandwidth, while less compression is associated with better voice quality.

free-sip-trunk-for-testing

Why Choose G.711?

Although there are many voice codecs out there, two are by far the most popular, G.711 and G.229. G.711 provides uncompressed high quality voice. G.729, on the other hand is compressed so that it uses less bandwidth at the sacrifice of quality.

Back when high speed bandwidth was rare and expensive, it might have made sense to favor “good enough” calls to reduce bandwidth requirements. Now that high bandwidth connections are inexpensive and available to almost every business, SIP buyers no longer need to worry about this tradeoff. By choosing a SIP trunking provider, like SIP.US, that leverages the G.711 codec, businesses can get excellent voice quality without too much concern over bandwidth.

How Much Bandwidth Do I Need?

The G.711 voice codec consumes 85kbps of internet bandwidth up and down. Let’s look at a low end example. If you have a DSL connection of 512kbps up and 3M down, the limiting factor will be the upstream limit of 512kbps. If you divide 512 by 85, you get 6, which is the maximum number of calls that this particular connection will support. Most modern broadband internet connections are much faster and can support dozens of calls at one time.

So, when considering voice codecs, there is some tradeoff between quality and bandwidth. For most customers, the bandwidth concerns do not outweigh the business need of high quality voice calls and G.711 is the right choice.

free-sip-trunk-for-testing

Free SIP Trunk in 60 Seconds

Related Posts

SIP and VoIP work hand in hand

Choosing the Right SIP Trunk Provider: Features, Benefits, and Value

July 22, 2024

The telecommunications industry has undergone a dramatic transformation, with traditional phone systems giving way to…

Read More

Cloud Telephony Demystified: Understanding Its Features and Benefits

July 16, 2024

Discover the benefits, features, and workings of cloud telephony solutions. Enhance efficiency and communication in your business today.

Read More

Mastering Cloud SIP Trunking: A Guide for Modern Businesses

July 16, 2024

Master cloud based SIP trunking with our comprehensive guide. Discover benefits, features, and implementation tips for modern businesses.

Read More