A codec (coder-decoder) turns the caller's voice into digital data for the network, and back into sound at the other end. The codec decides how much bandwidth a call uses and how natural the voice sounds. When a call is set up, both sides list the codecs they support in their SIP messages and agree on one they share.
The three codecs found on most business lines:
- G.711: uncompressed telephone-quality audio at 64 kbit/s. It is the quality reference and almost every operator supports it, but it uses the most bandwidth: about 87 kbit/s per call once network overhead is added.
- G.729: compresses the voice to 8 kbit/s, about 31 kbit/s with overhead. Useful when bandwidth is short, at a small cost in audio quality.
- Opus: a modern open codec that adapts its bitrate to the connection and can carry wideband audio, so voices sound clearer than on a classic phone line. Browsers use it for WebRTC calls.
When the two ends have no codec in common, the platform transcodes, converting from one codec to the other, which adds a little delay and processing. To size a connection, multiply the bandwidth per call by the number of simultaneous calls and keep a margin for other traffic.
OVOKY's SIP trunk supports G.711, G.729 and Opus.