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Guide to VoIP Codecs and How They Affect Call Quality

A simple and complete guide to VoIP codecs.

Voice over IP (VoIP) allows businesses to communicate with customers near and far with reliable voice quality through the internet. To understand how to get the best voice call quality from your VoIP phone system, you need to pay attention to voice codecs.

So, how do VoIP codecs support VoIP call quality?

In this guide, we will go over:

  • What are VoIP Codecs?
  • What Voice Codecs are used for VoIP?
  • What is the Best Codec for VoIP?

Let’s dive in!

What is a VoIP Codec?

A VoIP codec is a technology that establishes the audio quality, bandwidth, and compression of VoIP calls. The term codec is a portmanteau of Compression and Decompression.

When placing calls with VoIP, the voice needs to be encoded and converted into data packets. During this process, data is compressed to increase transmission speed and improve caller experience with crystal-clear voice.

This is where codecs come in, as they help encode and decode voice.

Why are Codecs Important for Voice over IP?

VoIP codecs convert analog voice signals into digital data packets (compression) and then convert them back to voice at the destination (decompression).

Because of this crucial process, these codecs determine the quality of your VoIP calls. Specifically, they influence latency, packet loss, and other VoIP call issues that may occur when calls travel over the internet.

Users can adjust these voice codecs to meet different needs, such as improving voice quality or reducing bandwidth consumption. You can work with your VoIP provider to understand what codecs they use for their service and how that may impact your communication.

What Codecs are used for VoIP?

Here’s a list of common VoIP codecs:

Codec Bandwidth
(kbit/s – bit rate)
Key Points
G.711 64 kbit/s
  • Focuses on precise speech transmission
  • Two variants: μ-law (US and Japan) and A-law (Europe)
  • 8 kHz sampling frequency
  • Compression ratio 1:2 – 16-bit samples into 8 bits
  • Requires high bandwidth
  • Good for LANs
  • High MOS of 4.2 when conditions are met
  • No licensing fees
  • Best codec for VoIP-PSTN connections
G.722 48 kbit/s
56 kbit/s
64 kbit/s
  • High-definition voice codec
  • 16 kHz sampling frequency
  • Adapts to varying compressions
  • Improves audio quality
  • Lowers latency
  • Better quality and clarity
  • Free
G.723.1 5.3 kbit/s
6.3 kbit/s
  • High compression
  • High-quality audio
  • Low bandwidth requirement
  • Works with dial-up
  • Requires more processor power
G.726 16 kbit/s
24 kbit/s
32 kbit/s
40 kbit/s
  • 8kHz sampling frequency
  • Most used mode – 32 kbit/s
  • Commonly used on international phone trunks
  • Standard codec for DECT wireless phone systems
  • Improved version of G.721 and G.723
G.729 8 kbit/s
  • Excellent bandwidth utilization
  • Acceptable quality
  • Encodes audio in 10 milliseconds-long frames with 80 audio samples
  • High compression rate
  • Supports multiple calls simultaneously
  • Royalty-free
GSM 13 kbit/s
  • Global System for Mobile Communications (GSM)
  • High compression ratio
  • Free
  • Same encoding used in GSM cellphones
  • MOS of 3.7
iLBC 15 kbit/s
  • Internet Low Bit Rate Codec (iLBC)
  • Free
  • Used by many VoIP apps, including open source
  • Tackles packet loss, delay, and jitter
Speex 2.15 kbit/s
44 kbit/s
  • Free software speech codec
  • Most preferred for many VoIP apps and podcasts
  • Uses variable bit rate to reduce bandwidth usage
SILK 6 to 40 kbit/s
  • Developed by Skype
  • Available as open-source freeware
  • Base for the newest codec: Opus


What is the Best Codec for VoIP?

While there are a few different voice codecs available, you need to find the VoIP codec that works best for you.

So, which codec is better, G.711 or G.729?

This depends on your business’ bandwidth usage and capabilities, as well as call volumes. But the consensus is that the G.711 seems to offer the most reliable call quality. This codec provides uncompressed high voice quality, but also has high bandwidth usage.

G.729, on the other hand, is the low bandwidth alternative to G.711. However, it may only offer acceptable call quality.

So, it comes down to your specific business circumstances and resources. For these reasons, most VoIP providers accept G.711 and G.729.

Alternatively, G.722 also offers high voice quality, but not all VoIP providers accept this voice codec. So, make sure to ask.

Related: How Much Bandwidth is Needed for VoIP?

Choosing the Right Codec for Voice Calls

Since VoIP codecs need to compress and decompress audio traveling through your phone system, you and your provider must agree on the right codec. In other words, the codec you choose needs to support your team’s bandwidth usage as well as work effectively with your provider’s network.

This means you’ll need to speak with your desired provider to understand their requirements and evaluate that alongside your business.

United World Telecom supports the G.711 codec (both μ-law and A-law), and our call quality has an average MOS of 4.3.

Want to see if we’re a good fit for your business? Call us today at 1 (877) 898 8646 or chat with us online!

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