Codec (Audio Codec)

The word codec combines coder and decoder. In VoIP it refers to the method used to turn microphone audio into a stream of bytes small enough to send over a network, and back again. Different codecs trade quality against bandwidth and processing cost.

During call setup each side lists the codecs it supports in SDP, and the answer picks one. If the lists do not overlap, the call fails with a 488 Not Acceptable Here response or connects with no audio. This is why administrators match codec settings between softphones, PBX and trunk provider.

Common codecs in SIP include G.711, which sends uncompressed telephone-quality audio at 64 kbit/s in ulaw (PCMU) and alaw (PCMA) variants; G.722, which offers wider audio bandwidth at the same bitrate; G.729, which compresses to 8 kbit/s but was patent-encumbered for years and is still often missing; and Opus, defined in RFC 6716, which adapts from very low to high bitrates and handles packet loss well. RFC 3551 assigns the standard payload type numbers for the older codecs.

SIPdia supports Opus, G.711 ulaw and G.711 alaw. It does not include G.722 or G.729. Use the bandwidth calculator to see what each codec costs on your connection.