Codec selection

Codec is a portmanteau of coder-decoder. A codec is an algorithm for encoding an audio or video signal for storage or transmission over a digital communications network. Codecs are responsible for compression of the data stream and may also take some responsibility for error correction, packet loss concealment and silence suppression.

Each device or softphone typically has one or more codec algorithms included. Some software, such as the Asterisk PBX, can support additional codecs with the help of modules or plugins.

The list of codecs supported by a product depends on the age of the product, patents and the type of products it is intended to interact with. Open source solutions generally avoid patented codecs, although unofficial implementations of them can be found online, such as the popular G.729 codec for Asterisk.

Table 4.1. Common codecs

NameTypeBitrate (kbps)PatentedComments
G.711 (alaw, ulaw)audio64NWidely supported in phones, WebRTC browsers, ISDN gateways and virtually everywhere else. Quality of traditional phone calls.
G.722audio64NSupported in most modern software and high quality desk phones. Transmits higher quality wideband audio in the same 64kbps bandwidth used by G.711
Opusaudio6 - 510NSupport in more modern software, WebRTC browsers and some very recent desk phones.
G.729audio8YA low bitrate codec supported in a lot of older VoIP phones and related hardware. Voice quality less than a standard telephone call.
G.723.1audio5.3, 6.3YAn ultra-low bitrate codec support in some older VoIP phones and related hardware. Voices sound very bad.
GSM full rateaudio13NThe original codec for GSM mobile telephony. Supported in many older open source products and some VoIP hardware.
iLBCaudio15NA codec developed for Internet use before Opus. Supported in many older open source products but not widely supported in VoIP hardware.
speexaudio2-44NA codec developed for Internet use before Opus. Supported in many older open source products but not widely supported in VoIP hardware.
VP8videodependsNMandatory part of WebRTC, supported in browsers. Bitrate depends on resolution and frame rate.
H.264videodependsYMandatory part of WebRTC, supported in browsers and in many existing video conferencing hardware products.

The person configuring the software or device can typically select which codecs are permitted and also specify which codecs are preferred over others.

When a call is made, the endpoints negotiate to select a codec that is supported at both ends. If both endpoints have no codecs in common, the call is not possible and the user may see a message telling them the call failed. If both endpoints have more than one codec in common, the exact codec selected in this negotiation algorithm depends on the relative priorities specified by the administrators of each endpoint.

Some software has the ability to dynamically change the list of permitted codecs. For example, some mobile apps will only enable high-bandwidth codecs when they detect the mobile device is using wifi or a 4G/LTE network.

More modern codecs support a variable bit rate that can be changed automatically during a call to adapt to poor network conditions. The Opus codec used for WebRTC has this capability.


Enable as many codecs as possible to maximize the chance of connection. This reduces the number of calls where the endpoints fail to find a common set of codecs.

Disable those codecs that won't possibly work given the available bandwidth. For example, in a remote location with a 128kbit DSL broadband connection it is usually necessary to disable 64kbit codecs like G.711 and G.722 and use codecs that have lower audio quality such as Opus, GSM and G.729.

However, in a location with good bandwidth, don't disable the low-bit rate/low-quality codecs. They will still be needed when calling a user who doesn't have great bandwidth.

Order the remaining codecs based on the quality, with the best quality codec first. G.711 is typically present for compatibility but other codecs like G.722 offer better quality for the same amount of bandwidth, so rate G.722 ahead of G.711 and G.722 will be used whenever possible.

Example 1: a mobile app: enable, in order of priority starting with the most preferred: Opus, Silk and GSM codecs.

Example 2: a soft PBX in an office accessed by local users and mobile users: enable, in order of priority starting with the most preferred: Opus, G.722, G.711, Silk, iLBC, GSM, G.729.