G.729 reduces the network bandwidth used by each VoIP call, without sacrificing call quality. Audio compressed with G.729 uses only 1/8 the bandwidth of standard G.711 audio, which allows for more calls to be carried without increasing network capacity and allows voice to travel on limited-bandwidth connections that would otherwise not support VoIP.

Cut bandwidth requirements without sacrificing call quality.  Standard G.711 calls take 64kbit/s per call.  The G.729 codec compresses the payload to 8kbit/s, giving you up to eight (8) times the capacity on the same connection.  Ideal for use in limited bandwidth scenarios (ADSL connections, international VoIP service, satellite connections, etc.).

Up To 140 Calls On A Single T1

A practical example is the number of calls that may be carried across a standard 1.5 megabit/s T1 link. When using uncompressed G.711 audio, one can expect 18 concurrent calls across a T1. When using G.729 compression and Digium's IAX2 Trunking, instead of SIP, signaling protocol, one can expect about 140 concurrent calls across the same link.

Digium's implementation of the G.729 codec allows Asterisk software to convert audio between G.729 and any other supported codec. Many IP telephones and VoIP gateways include support for G.729. With the Digium G.729 Codec for Asterisk, those devices can now exchange calls with Asterisk directly at a fraction of the bandwidth of standard G.711.

Without the capability to transcode G.729, Asterisk software can only pass-through G.729 data between endpoints. This means that basic station to station calling can be made to work, but the advanced PBX features of Asterisk such as Call Conferences, DTMF digit collection, Call Recording and more will not work without Digium's licensed G.729 Codec. NOTE: After downloading the G.729 Codec, you can then buy the license key  from the Digital Techniques Web Store.

Download G.729 Codec