Last week we began our iPBX comparisons with Fonality and AstLinux. Today we’re wrapping up with products from Pingtel and Digium.
One important note about PBXtra and trixbox Pro: Both are designed to be your
LAN’s primary DNS server. If you want your existing DNS server to remain as
your primary, you’ll have to set up DNS forwarding. (Read
about it here.)
SipX and SIPxchange
SipX is the free-of-cost open source SIP-based iPBX from Pingtel. SIPxchange
is their commercial edition. They are very similar, with a few
differences. The main ones are that SIPxchange includes a call center module,
and that it uses the closed-source commercial GIPS media framework instead of
open-source codecs. Neither edition supports the multitude of VoIP protocols
(IAX, H.323, MGCP, SCCP, UNISTIM, etc.) that Asterisk supports; just SIP.
SIP has become the popular VoIP protocol: Virtually all VoIP networks, endpoints
and gateways support SIP. IAX is a native Asterisk protocol that is not nearly
as well supported as SIP. H.323 is the grandmommy of VoIP protocols, and is
used mainly by big commercial carriers. It is well-supported in most VoIP gear.
SIP is similar to H.323 in the way it works, but is less complex and not nearly
as heavyweight.
SipX and SIPxchange claim to deliver superior call quality and support greater
call volumes by implementing the SIP protocol the way it’s designed to be used.
SIP separates the signaling stream from the media stream. The signaling stream
handles the job of registering and authentication, and then the audio stream
can travel by the most efficient route between endpoints, even bypassing your
server entirely. Asterisk makes itself the endpoint, so all call traffic must
go through it. Thus, it functions as its own bottleneck. (If you want a snazzy
VoIP term for this, it is “back-to-back user agent”, or B2BUA.) The server does
not do any media transcoding, so that’s a CPU-intensive chore the server doesn’t
have to handle. Since transcoding is almost always necessary, the easy way to
handle this is to pair your SipX/SIPxchange server with a media gateway like
the Cisco 37xx series, or the Audio Codes MP-108. (Visit Pingtel.com
to find a list of certified devices.)
SIPxchange has various levels of vendor support and training. It comes in
three editions: Enterprise Communications Server, CallManager, and SIPxNano.
Enterprise Communications Server is pretty much the same as SipX, but it includes
a call center module. CallManager is a management interface for a distributed
VoIP network with multiple servers or locations. SIPxNano, as the name implies,
is a wee appliance that puts Enterprise Communications Server (minus the call
center module) on a stylishly small server. This is intended for small shops
with up to 30 users.
SipX is free of cost and Free software, licensed under the LGPL. It offers
only community support, which is pretty good, plus a lot of documentation.
It’s a free download that includes the operating system, so it’s an easy one-CD
install. Administration is via Web interface.
Both products auto-provision a number of phones, such as Polycom, snom, LG-Nortel,
Cisco, and Grandstream. Auto-provisioning phones is a huge time saver, so always
look for this.
SipX also includes a DNS server. See VoIPowering
Your Office: Recovering SipX Passwords and DNS Done Right for more information
on integrating it into your network.
Asterisk itself
And now we come to the original iPBX and the beginning of the VoIP revolution, Asterisk. Asterisk is currently available in four editions: AsteriskNOW and plain-vanilla Asterisk, both of which are free of cost and open source, Asterisk Business Edition, and the Asterisk Appliance.
Plain old Asterisk is for gnarly geeks who want to customize it to their own specifications, or who want to keep up with the latest releases, or who want to install it on their own choice of operating system. Once you learn your way around Asterisk’s vast herds of configuration files you can make changes and updates pretty fast, and it’s easier to copy text files than to re-enter gobs of stuff in graphical forms.
AsteriskNOW is a software appliance that includes the operating system, which
is the streamlined and customized rPath Linux. rPath is designed for custom
appliances such as AsteriskNOW. It has a different set of commands than true
Linux, so you’ll have to get used to new ways of doing things. It also includes
the excellent AsteriskGUI, which is for administering both Asterisk and rPath
Linux.
Asterisk Business Edition comes with support, training, and custom programming options. Unlike the free versions, it undergoes considerable quality control testing. It also comes with the AsteriskGUI.
The Asterisk Appliance comes on a sleek small box about the size of a 12-port switch, and is designed to support up to 50 users. This is based on an embedded Linux, uClinux, and it also includes the AsteriskGUI. It supports up to 8 analog ports, if you need them, or just plain old pure VoIP.
Which one?
Hopefully this series has given you enough information to decide which, if any
of these, you want to try. There is no such thing as plug-and-play. There is
a learning curve no matter which one you pick—not only in telephony but
in IP networking and name services, and bandwidth management. The upside is
any of these offer you a wealth of features and flexibility that traditional,
way-more-expensive PBX systems don’t even come close to offering.
Resources
EnterpriseVoIPplanet.com
VoIP-info.org
