G.729 for Asterisk IPBX - Guidelines for Implementation

G.729 is one of the most popular voice compression protocols used in IP telephony. This is due to the fact that G.729 reduces the bandwidth usage for a standard telephone call from 80 kilobits per second (kbps) to 24 kpbs, with minimal voice quality degradation.

While in the past the G.729 algorithm ran on dedicated Digital Signal Processors (DSP), today’s processors can easily run the G.729 algorithm effectively while addressing other tasks as well. This document describes how to add the open G.729 codec to Xorcom’s XR1000, XR2000 and XR3000 stand-alone IP-PBX appliances:

XR1000 IP-PBX for Small Office / Home Office

XR2000 IP-PBX for Small/Medium Business

XR3000 IP-PBX for Enterprise

Conversion Times Based on Compression Method

G.729 requires CPU resources. The tables at the end of this document show the conversion time (using different compression methods) that can be achieved by each member of Xorcom’s IP-PBX family.

Methods for Obtaining the G.729 Codec

There are two ways to obtain the G.729 codec for an Asterisk-based IP-PBX:

1. Buy the licenses from Digium
2. Install the Open Source version

Both codec versions provide the same performance, but in case of the Open Source version you undertake the responsibility for payment of the license fee. The legal aspects of using the Open Source version of the G.729 codec are discussed in the following article: http://voip-info.org/wiki/view/Asterisk+G.729+Licensing

Installing the Open Source G.729 Codec

To install the Open Source implementation of G.729 codec on the Xorcom XR1000/2000/3000 perform the following actions:

1. Download the codec binary file from http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-pentium4.so
2. Copy it to the /usr/lib/asterisk/modules folder.
3. Restart the Asterisk: /etc/init.d/asterisk restart or amportal restart

Estimating the Number of G.729 Channels Required

If you choose to purchase the G.729 license from Digium you will need to compute the number of G.729 channels required by your configuration. You can estimate the required value by using the following information:

• a call between two SIP extensions usually requires two G.729 channels, unless the pass-thru mode is used (http://voip-info.org/wiki/view/Asterisk+G.729+pass-thru), in which case it doesn’t require any G.729 channel.
• a call between a SIP extension and a Zaptel/DAHDI extension/trunk requires one G.729 channel.
• a call to Voice Mail or another Asterisk service where IVR messages must be played requires one G.729 channel.

Typical Translation Time Between Formats: XR3000

The following table contains the values of typical translation time between formats (in milliseconds) for one second of data source format (rows) to destination format (columns) in the Xorcom XR3000 IP-PBX appliance:

Typical Translation Time Between Formats: XR2000

The following table contains the values of typical translation time between formats (in milliseconds) for one second of data source format (rows) to destination format (columns) in the Xorcom XR2000 IP-PBX appliance:

Typical Translation Time Between Formats: XR1000

The following table contains the values of typical translation time between formats (in milliseconds) for one second of data source format (rows) to destination format (columns) in the Xorcom XR1000 IP-PBX appliance:

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