Abstract: This article describes the bandwidth and call concurrency of Asterisk PBX as a means to determine VoIP operators realistic maximum number of concurrent calls. By gathering the maximum number of concurrent calls, the required number of Asterisk PBX servers may be calculated. This article assumes the dimensioning figures documented by Digium and others for the number of concurrent calls and assumes the VoIP operator will be using a voice codec that compares with the quality of the POTS.
General Bandwidth Requirements
The g711 ulaw/alaw codec is the only codec that matches the voice quality offered by the traditional POTS. The g711 codec uses 64 Kbps of bandwidth for the digitized voice signal. Once the overhead of the Internet protocol is added, it realistically uses 87.2 Kbps. This amount of bandwidth is used in both directions. A minimum of a 128 Kbps broadband connection is required to successfully utilize this codec.
| Related Side-note |
|
Internet Bandwidth Requirements
A high-end server running Asterisk PBX can handle around 100 concurrent calls or 200 concurrent channels. A single call comprises of at least 2 channels, but may use more depending on the dialplan logic employed. If we use this figure and multiply it by the bandwidth usage, we realize a total of 17,440 Kbps - Each call has two channels, each channel is 87.2 Kbs, and we have 200 channels.
A T1 has 1,544 Kbps of available bandwidth. This means we can safely place 17.7 channels onto it. A T3 in the US has 44,736 Kbps of available bandwidth. If we assume that only a single channel utilizes the Internet, because the other channel is utilizing TDM/Zap, we could place 513 calls on a T3. This equates to 2.56 Asterisk systems completely saturating a T3. However, if we assume that both channels utilize the Internet, then we could only place 256.5 calls on a T3.
An OC-3 has 155,520 Kbps of available bandwidth. This equates to 1,783.48 channels and 8.9 Asterisk systems.
In the examples described above, you must remember that this includes no other traffic, such as DNS, email, HTTP, etc. The realistic amount is far less unless your traffic is comprised of only VoIP traffic.
Computing Requirement
The figure of 200 concurrent channels is based on a dual Xeon 2.8 Ghz system with 1 GB of RAM performing no echo cancellation and no codec transcoding. A server that is performing echo cancellation and/or codec transcoding will perform far fewer concurrent channels. In addition, a server with Zaptel hardware will also detrimentally effect the maximum number of concurrent channels.
Additional call concurrency may be realized by employing Sun Microsystems hardware and the Solaris operating system providing that features available only to the Linux operating system are not needed. Features that are not presently supported on the Solaris operating system are Zaptel kernel drivers which control TDM based hardware and the Meetme conferencing application of Asterisk PBX. For more information about Solaris verses Linux and the Asterisk PBX, please see the authors additional article listed in the references.
Conclusion
The majority of VoIP operators can probably suffice with a single Asterisk PBX system on a high-end server, due to the nature of high price tags associated with Internet connections and the wide range of traffic traversing their Internet connection.
In summary, using Asterisk PBX to offer residential calling features, one can expect to service around 80 to 100 concurrent calls. Using the 1 person to 10 people guideline, a typically residential customer base only has 10% of the customers concurrently using the phone. This means a single Asterisk PBX is capable of handling around 1,000 residential customers. However; this number does vary on the type of features and services offered.
Changelog
Updated on Mar. 20, 2008 - Corrected the wording for channel/call above. Reported by Joel Bresler.
References
VoIP Calculator
http://www.agilevoice.com/f/calculator/voip.htm
Bandwidth Consumption of VoIP Codecs
http://www.voip-info.org/wiki/index.php?page=Bandwidth+consumption
Solaris 10 verses Linux: Asterisk PBX
http://www.thrallingpenguin.com/articles/asterisk-solaris.htm
Quality of Service for Voice over IP
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00800d6b73.shtml
Asterisk Dimensioning
http://www.voip-info.org/wiki/index.php?page=Asterisk+dimensioning
Sun Microsystems
http://www.sun.com/
Asterisk PBX
http://www.asterisk.org/
About the Author
Joseph Benden, Sr. is the owner of Thralling Penguin LLC. Thralling Penguin designs, develops, and extends software technologies for the most demanding business applications, as well as offering VoIP Consulting services.