SIP Trunking: Codecs & Call Quality

This is our 7th of 12 blogs on the subject of SIP Trunks. Subscribe in the box to the right to receive an email notification for each one in the series.

Click here to read the previous blog in this series.

SIP Trunking: How to Measure Call Quality

So, now that we have a good Internet circuit (or two), it’s time to talk about the components that make up VoIP call quality.

“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?

There is a more scientific method that is used when evaluating VoIP call quality using a scale.

Mean Opinion Scores (MOS)

MOS is a test that has been used for decades in telephone networks to measure 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

Regardless of which slogan became part of your lexicon, the fact remains that since the advent of cell phones and VoiP calls, we now regard voice quality in a different light.  Today, the most common method to use for judging call quality is to ask; “is it less than, equal to or better than an excellent cell phone call?”  Most business users consider a call that is the equal of a good cell phone call acceptable for business purposes.

What’s a Codec?

codec, which stands for coder-decoder, converts an audio signal (your voice) into digital form for transmission (VoiP) and then back into an audio signal for replay. Codecs vary in the bandwidth required per phone call and the quality of the voice transmission. 

Common VoIP Codec Protocols

G.729:  G.729 is a codec that substantially compresses voice from 64K to 8K, but still provides decent call quality. It has a MOS rating of 4.0 and is preferred in situations where bandwidth is precious, such as international calling.

G.711: G.711 is a codec that has no compression.  It is the default for quality SIP Trunk providers and is used anywhere good bandwidth is available. G.711 has a MOS rating of 4.2.  It is the most common used codec in domestic phone calls.

G.722:  G.722 is a high bit rate codec which, because it is of even better quality than 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.  This codec is not widely available on SIP Trunks as of this writing. 

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 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)


Equally if not more important than raw bandwidth is quality of service (QoS). QoS is the technology used to prioritize a specific type of data on a data network.   In this case, we are talking about prioritizing voice over all other data traffic. 

QoS is most valuable in situations where your data circuit is less than optimal or you have a high volume of usage on your circuit. A good analogy is the modern freeway.  If it’s 3 AM and there is light traffic, every vehicle in every lane travels unimpeded, just as voice would travel unimpeded on a circuit with little traffic.

However, this all changes as traffic volume increases. QoS is the equivalent of the commuter lane where voice gets special priority. However, in rush hour traffic, even the fast lane slows. QoS is a great tool to ensure high voice quality on a good data circuit, but it doesn’t fix a substandard or overworked data circuit.

Do I Need QoS:

QoS is recommended on all but the largest asynchronous circuits. QoS is not required for larger synchronous circuits that aren’t filled to capacity.  And for locations that have no or very little regular data traffic, there is no need to prioritize voice. You can’t place a priority over something that doesn’t exist.

How do I get QoS:

QoS can be provided in two ways:

Hardware QoS: QoS can be established by implementing a piece of hardware that prioritizes voice. There is an upfront cost if you don’t already have the equipment that can be set up to provide QoS. QoS is typically established in the settings of a router or firewall.

Carrier QoS: The Internet carrier can prioritize voice in its network. A minority of carriers provide this at no charge, with most charging a monthly fee. Carrier-based QoS is somewhat superior to hardware-based QoS, but more importantly, any QoS is quite superior to having no QoS.

This is an area where a quality SIP Trunk provider can provide consultation, recommendations and aid in implementing the proper solution.

In my next blog in this SIP Trunking series, we will discuss SIP Trunking on Legacy Phone Systems and if they are right for your business.  You can’t make a great decision without reading this blog!

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