Codec Support - Voice Quality & Networking

A codec, digitises and compresses an audio signal for transmission and as the second part of the process uncompresses that same audio stream for replay.

As to the question of the appropriate codec for any one individual requirement the answer is largely driven by the quality of your network connection. Customers connecting directly to our service via a Layer2 service, or customers with high speed fibre connections can generally use the higher quality codecs like G.722. The quality of those calls will be fantastic. On the other hand if the bandwidth is pure public internet, go for the low bandwidth codecs like ILBC or G.729.

See also Ports and IPs.

MOS Scores

Mean opinion score (MOS) is a test used 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

Common VoIP Codec Protocols
  • 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: 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: 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.
Supported Codecs
  • Voice: G.711 A Law, G.711 U Law, G.722, GSM, iLBC, and G.729 with automatic transcoding between codecs.
  • Video: H.264 and H.263.
  • Fax: T.38 pass-through and termination.

  1. Log onto your Account.
  2. Select Switchboard and the phone number you wish to alter.
  3. Select Preferences.
  4. Go to Voice Quality & Networking.
  5. Set preferences.
  6. Set Codecs (recommended SysAdmins only).
  7. Set DTMF (recommended SysAdmins only).


The case for and against G.722

Calls made between two phones using the G.722 sound fantastic, so good in fact the voice quality is as if the two parties are in in the same room. Technically G.722 uses 8khz and the reason it sounds so great over VoIP networks is you are in fact hearing the consonants like ‘f’ and ‘s’ which you’ll struggle to decipher over a G.711 call.

The problem, or inherent weakness of G.722 is once it hits the PSTN and more particularly the tier one carriers that only support the narrow band G.729 and G.711 codecs. When the PSTN carriers transcode back into G.711 they drop the frequency from 8khz back to G.711’s 4kHz rendering the audio inferior to that G.711 or even G.729 ultimately risking a potential loss of voice quality.

Other settings

Not behind NAT:
It has a public IP address or port forwarding setup. When SIP Peering is enabled, NAT is always disabled.

Qualify polling options:
This is used to track the registration status.

RFC2833 Compensate Feature:
DTMF are transmitted with RTP packets just like audio on the same network connection, but are encoded separately as a different payload type than the audio stream.
With this feature, customers with reliable internet connections can generally use the higher quality codecs like G.722. The quality of those calls will be fantastic. On the other hand if the bandwidth is up and down, our recommendation is bandwidth codecs like ILBC or G.729.