Codec

Telecoms glossary • C

Codec

Audio compression • agreed during call setup, before any sound flows

The choice that sets what a call sounds like and what it costs your line. Your phone and your provider make it for you, in milliseconds.

What is a codec?in VoIP and business telephony

1. A codec is the algorithm that turns speech into data at one end of a call and rebuilds it at the other. It fixes the trade-off between how good the call sounds and how much of your line each call uses, and your handset and provider agree on which one to use during setup, before any audio flows.

2. The name is short for coder-decoder. Only some codecs actually compress: G.711 sends the same 64 kbps of pulse-code modulation the old digital exchanges used, while G.729 squeezes the same speech into a fraction of that and gives up detail to do it.

87.2 kbpsOne G.711 call on Ethernet
31.2 kbpsThe same call on G.729
7 kHzG.722 audio bandwidth, ITU-T
50/secPackets sent per call at 20 ms

The four codecs you will actually meet

Provider portals list a dozen options. Four of them account for almost every South African business call, and the per-call figures below are Cisco’s, measured over Ethernet with headers included rather than quoting the raw payload rate.

Codec Audio bandwidth Per call on Ethernet Where it fits
G.711 3.4 kHz narrowband 87.2 kbps The safe default. Uncompressed, universally supported, passes fax and keypad tones intact.
G.722 7 kHz wideband 87.2 kbps HD voice. Same cost on the wire as G.711 for noticeably clearer audio, but both ends must support it.
G.729 3.4 kHz narrowband 31.2 kbps Constrained links. Roughly a third of the traffic, at a real cost in clarity.
Opus Up to 48 kHz Variable, 6 to 510 kbps Browser and app calling. The default in WebRTC, and it adapts its rate as conditions change.

G.722 is the one most businesses leave switched off by accident. It carries the same 64 kbps payload as G.711, so it costs nothing extra in bandwidth, and it roughly doubles the audio bandwidth the handset reproduces. See HD voice for what that sounds like in practice.

How the codec gets chosen

Nobody picks a codec on a call. The endpoints negotiate it, and the outcome is decided by the lists each side was configured with.

  1. Your phone offers a list

    The SIP INVITE carries a session description listing every codec the handset supports, in the order it prefers them.

  2. The far end answers with one

    The other side compares that list against its own and picks the first entry both support. Preference order on the calling side usually wins.

  3. Media starts

    Only then does RTP begin carrying packets, at 50 packets per second for a standard 20 ms sample.

  4. Or somebody transcodes

    If the two lists share nothing, the provider’s platform converts between them. That works, but it adds delay and a second round of quality loss, so a call that has been transcoded twice can sound worse than either codec alone.

This is why an internal call can sound better than an external one on the same handsets. Inside the platform both ends run G.722; the moment the call reaches the public network it drops to G.711 to match.

When the codec is the fault, and when it is not

Support desks reach for the codec setting far too early. It genuinely explains a narrow set of faults and none of the common ones.

It is the codec
Calls connect with total silence both ways, and the logs show no common codec offered. It is also the codec when fax fails on a line that carries speech fine, or when an automated menu misses digits because a compressed codec has mangled the in-band DTMF tones.
It is not the codec
Choppy audio in both directions is packet loss. Audio one way only is a firewall or NAT problem. A delay that has people talking over each other is latency. Switching to G.729 in any of those cases makes a quieter version of the same fault.
The compression trap
Dropping to G.729 to save a saturated line moves less data but sends the same 50 packets per second. Packet loss is counted in packets, not bytes, so a link losing 3% of packets loses 3% of a G.729 call too, and the smaller codec has less redundancy to hide it with.

Voice is cheap traffic. A busy 20-seat office on G.711 draws under 2 Mbps at peak, which is why QoS almost always beats codec changes as a fix.

Three things to put to a provider

Ask which codecs are enabled on your extensions by default, and whether G.722 is one of them. A surprising number of accounts ship with wideband disabled, which means the HD-capable handsets you paid for are running narrowband.

Then ask where transcoding happens on their platform, and whether G.729 is being applied to your account to conserve their capacity rather than yours. If a provider proposes G.729 as the answer to a quality complaint, ask them to show you the loss and jitter figures first.

About this entry

Definitions are written for South African business buyers and checked against primary sources – provider documentation, standards bodies and ICASA – not vendor marketing. Reviews are independent and sponsors are always disclosed. Read our editorial policy and scoring methodology.

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