Voice Quality & Network

Voice quality allows you to set parameters for your calling quality.

  1. Log into your Account.
  2. Select Switchboard.
  3. Select your desired phone number.
  4. Select Preferences.
  5. Voice Quality & Networking.
  6. Set preferences:
  7. Set Codecs (recommended SysAdmins only).
  8. Set DTMF (recommended SysAdmins only).
  9. Click Save settings to update.

There is also the ability to specify the codecs you wish to support. These are some options:

  • G.711 a-law codec supported (Excellent quality).
  • G.711 u-law codec supported (Excellent quality).
  • G.722 wideband codec supported (The best quality).
  • GSM codec supported (Good quality).
  • iLBC codec supported- MUST be 30ms/13.33kbps variant (Good quality).
  • G.729 codec supported (Good quality).
  • H.263 video codec supported.
  • H.264 video codec supported.
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 is 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 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.