For the past ten years I have worked from a home office full time. This has been the major motivation for my education in networking, and onward into VOIP technologies. Since the middle of 2005 we have not used traditional…
January 13, 2006
The Asterisk open source Voice over IP (VoIP) PBX is usually set up on a standalone PC. But Michael Graves shows how the combination of a special Asterisk distribution and a single board computer can provide a compact, quiet and low-power alternative.
Astlinux is a bundled distribution of the Asterisk open source iPBX private branch exchange (PBX) software and a Linux operating system. Originally developed by Mark Spencer at Digium, Asterisk is the leading open source software in the telephony/VoIP space. Asterisk excels at combining traditional TDM telephony capability – provided through hardware from Digium and others – with VOIPservices. These include call routing, media gateway, media server and SIP signaling capabilities.
The Asterisk user community has been growing tremendously over the past two years, especially since the v1.0 release in the fall of 2004. With that growth has come the development of new distributions that bundle suites of software tools, to ease the setup and administration of a new Asterisk system. Asterisk@Home and Xorcom Rapid are both fine examples of this sort of activity.
Astlinux was developed by Kristian Kielhofner, and intended to go in a fundamentally different direction. Astlinux provides an Asterisk installation on a Linux distribution that has been built from scratch and optimized for small format hardware platforms – it takes what is essentially an embedded systems approach to Linux and Asterisk. In this article, I’ll show you how to build an VoIP PBX using Astlinux and a Soekris Net4801 single board computer (SBC).
My experience has been that the QoS mechanisms covered previously don’t provide a complete solution to the need for assured bandwidth when using VOIP over DSL. When the connection to the ISP becomes saturated for any reason VOIP traffic can be delayed which is always a problem. When managed QoS was combined with “traffic shaping” our VOIP phone service became much more reliable. This has proven to be true even on a very busy connection to my ISP.
Like the QoS mechanisms covered previously, traffic shaping is an edge process that occurs in your router. Traffic shaping is actually a process of reserving bandwidth specifically for selected applications. That bandwidth will not be used for other forms of internet access. As before, this tends to be most critical with outbound traffic where available bandwidth is most limited. It’s also true with inbound traffic, but this tends to be less of an issue.
SOHO users like myself are unlikely to have high-end networking gear supporting their home office setups. My initial experience with VOIP over broadband involved using Vonage and a Linksys BEFSR-41 broadband router. At the time Vonage was the leading phone-over-broadband service and the BEFSR-41 was the leading SOHO router.
You hear an awful lot about “QoS, which stands for “Quality Of Service.” In point of fact it means different things to different people depending upon their perspective, telco or IP networking.
Here are a few fundamental considerations when planning a VOIP implementation using DSL.
- What is your actual available bandwidth inbound & outbound?
- How many simultaneous calls do you need to sustain?
- What voice codecs are you using? And so, how much total bandwidth do you require?
- Do you have managed QoS on your network?
- Can you also implement traffic shaping to reserve bandwidth for VOIP purposes? Especially outbound bandwidth as this is typically the most scarce.