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Which VoIP/SIP Codec Should You Use?

How To Describe the Quality of a SIP Trunking Call

“So clear, you can hear a pin drop!”

Are you old enough to remember the very successful tag line for Sprint Communications’ long distance service back in the ancient 1990’s?

“Can you hear me now?”

Or…Did you grow up remembering the equally (if not more) successful phrase that Verizon Wireless made part of our cultural reality in the new millennium?

Regardless of which slogan became your favorite lexicon, the fact remains that since the advent of cell phones and VoIP calls, we now regard voice quality on quite a different scale.

Today, the most common and simple method to use for judging call quality is to ask; “is it less than, equal to or better than an excellent cell phone call?” 

On the other hand, a more professional method that is used when evaluating the quality of VoIP call quality is a scale called the mean opinion score.

Mean Opinion Scores (MOS)

Mean opinion score (MOS) is a test that has been used for decades in telephony networks to obtain the human user’s experience of the quality of a phone call.

The MOS is assigned by a group of listeners using the following values:

5 – Excellent
4 – Good
3 – Fair
2 – Poor
1 – Bad

Now, that we have a way to compare voice over IP call quality, let’s take a look at the magical (and very geeky) technology that can be used in VOIP phone calls.

Codec Types

What’s a 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 essence of VoIP. 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.

Little Known Fact – You can assign a different codec to individual phones. Your staff can use medium quality/low bandwidth G.729 codec while the boss and legal department uses the superior quality/heavy bandwidth G.722 codec.

Common VoIP Codec Protocols

G.729  G.729 is a codec that has low bandwidth requirements but provides good audio quality. This is the most commonly used codec in VoIP calling and has a MOS rating of 4.0

G.711 G.711 is a codec that was introduced by ITU in 1972 for use in digital telephony. With only a 1:2 compression and a 64K bitrate for each direction (128K plus some overhead), it is best used where there is a lot of bandwidth available. G.711 has a MOS rating of 4.2

G.722 G.722 is a high bit rate (48/56/64Kbps) ITU standard codec which, because it is of even better quality of the traditional public switched telephone network (PSTN), it can be used for a variety of higher quality speech applications. This standard also requires an adequate amount of bandwidth and usually rates a 5.0 on the MOS scale.

Summary

There are many codecs out there, some like the G.711 are royalty-free, others require licensing (which is often included in the gateways). Some that we haven’t mentioned here drive the wholesale movement of voice traffic among the carriers and are used for specialty applications. And to make it a bit more confusing, they all contain variations within their own specification.

How to Decide?

The codecs that provide the best quality consume the most data bandwidth, thus there is a trade-off that you need to consider. The easiest way is to ascertain, on a phone by phone basis, whether you want the voice conversation to be:

  • Slightly less than the quality of an excellent cell phone call (G.729)
  • Equal to the quality of an analog land-line today (G.711)
  • Better than the public switched telephone network for voice critical applications (G.722)

Need some help in reviewing your SIP Trunking needs? For more information or any help you need, please call us on (+44) 0808-117-6736.

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