It’s been about a year since we visited the excellent SipX, the SIP-based iPBX server that handles large loads and scales upward with ease, and it has changed a lot in that time. Its parent company, Pingtel, was purchased by Bluesocket in July 2007. There have been several major releases since then, and a name change. That’s right, the new broom swept away the old names of both the open-source, free-of-cost SipX and its commercial cousin, SIPxchange. Now they are sipXecs and SIPxchange ECS. ECS stands for Enterprise Communications Server, and can either be pronounced “eks” or spelled out. Either way it seems an excessive number of syllables.
Naming follies aside, sipXecs and SIPxchange ECS are excellent options to consider when you’re shopping for a new iPBX. They are SIP proxies, which means they route SIP traffic the way it is intended to be routed, which is in a distributed fashion. A SIP proxy handles only signaling, and the media stream is routed directly between endpoints. This is in contrast to Asterisk, which is a Back-to-Back User Agent (B2BUA). All traffic must route through the Asterisk server, so it becomes its own bottleneck, and you are limited by its processing power. The main limitation on sipXecs and SIPxchange ECS is your available bandwidth.
Asterisk does not natively support clustering or failover, so when you want to build a distributed Asterisk system you need to use external tools, such as the The High Availability Linux Project, or clever hacks with media gateways and routers. sipXecs and SIPxchange include native support for high-availability systems, which can be set up at installation or added later as your needs grow.
An advantage of Asterisk is its support for a number of transport protocols, such as SIP, IAX, MGCP, and H.323; sipXecs and SIPxchange only support SIP. Back in the beginning of the VoIP revolution there were no clear candidates for a dominant VoIP transport protocol. H.323 is a tenacious oldtimer that still carries a significant amount of the world’s VoIP traffic, while SIP is the upstart that appears to be on its way to becoming the dominant protocol. You’re going to need some kind of media gateway anyway to interface with the PSTN, because sipEtc. does not support TDM cards like Asterisk does, so if you need H.323 or other protocol support, your media gateway will handle it. Which makes sense for any VoIP system- separating call routing and other functions, instead of dumping the entire load on a single server, is more efficient and flexible. Using a media gateway adds redundancy and failover, takes a significant processing load off the server, and makes it a lot easier to interface with multiple vendors.
sipXecs vs. SIPxchange ECS
The feature sets and codebase are similar, but there are some differences. sipXecs is 100 percent open source, licensed under the LGPL (GNU Lesser General Public License.) There is no vendor support, so you have the usual Wiki, mailing lists, and user forums should you need a hand. sipXecs is a free software-only download. It comes in an .iso image that includes the CentOS Linux operating system, so it’s an easy one-CD installation; you may also install it on your own favorite Linux distribution. There is also a binary for FreeBSD, and work is underway for Mac OS X, Solaris, and Windows ports.
SIPxchange ECS offers vendor support and training, and is available as software-only or in various turnkey hardware bundles. It has an administrator’s manual and a certified hardware list that include media gateways, servers, and phones. It includes some commercial third-party closed-source components, such as the high-quality GIPS (Global IP Solutions) media framework. (See Resources for more detailed comparisons.)
The feature sets for both have grown significantly. The latest releases of both include:
- Plug-and-play management of a number of phones and media gateways
- Personal auto-attendants
- Improved use control panel
- Integration with Microsoft Active Directory and Exchange 2007
- Supports single or multiple sites
- Redundancy and failover
- Least-cost call routing
- “Skinnable”, which for us old people means you can customize the appearance
- Security features such as HTTPS for the Web interface and TLS-based SIP signaling
- LDAP integration
- Call center module
While the sipEtc. family are capable VoIP servers all on their own, you can also use them as call routers in front of an Asterisk or other VoIP server. Come back next week and we’ll fire up sipXecs and see for ourselves just how improved it really is.