If you are not comfortable with Asterisk based IP PBX and you love working with Windows, there is a solution for you. 3CX phone system is a SIP based IP PBX for Windows with lots of promising features like voicemail via email, Unified Messaging (MS outlook integration) and call parking.

They have four different versions (Free Edition, Small Business Edition, Professional Edition and Enterprise Edition) to suit your needs, all of them support unlimited extensions (internal and external). The major differences between their free and commercial editions are that commercial editions support 8-32 numbers of Simultaneous Calls (depending on commercial editions) and other features like BLF status, call recording, show callers in queue and remote configuration of extensions. Commercial Editions also support G729 codecs to save bandwidth. 

3CX Phone System includes a compact VOIP Client which can be used in combination with a headset, as a fully functioning SIP software phone, or in combination with an SIP hardware phone. It works with all popular SIP soft and hard phones, SIP trunking providers and SIP VOIP Gateways like Grandstream, Patton (can be automatically configured within 3CX Phone System), Cisco, Linksys and others.

3CX Phone System has a built in VPN solution, which eliminates firewall reconfiguration and it also has an integrated Fax server that works with T.38 protocol .

I have been using their free version from last two years along with Trixbox CE and Pro at home with three SIP software phones (two internal extensions and one external extension) without any problem. 

3CX Phone System is well developed, easy to use and complete IP PBX application which challenges Asterisk based PBX systems.  For details visit their website  

If you are in any way involved with telecom, then you must have heard of SIPConnect. SIPConnect is the culmination of the effort by SIP Forum to leverage interactive IP communication services and solutions.

SIPConnect is the methodology and architecture specifications to deliver direct SIP connectivity with VoIP service providers to ensure consistent quality of service, low cost and enhanced feature set.

Companies like Cbeyond have utilized this methodology to provide consistent and enhanced services to Small to Medium Business’s (SMBs) while reducing costs.

When shopping for companies providing SIPConnect service, always look for SIPConnect Compliant certification mark . This mark can only be used by companies that use solutions and services that comply with the mandatory requirements of SIPConnect specification as published by the SIP Forum.

I’ve talked about potential pros of VoIP as it pertains to direct cost savings to the enterprise. There are some hidden benefits of VoIP as well, these don’t get the same attention or get measured like the other well known features.

Since VoIP works over the same infrastructure as your data traffic, this makes it very easy for VoIP systems to be integrated with your data systems. This is becoming more apparent as software like Microsoft’s Exchange 2007 gets released with direct unified communication hooks.

What this means to you is, you can easily integrate your voice and data to have one point of contact for your users. All email, faxes and voicemails come into the same universal inbox, this inbox gets backed up and is accessible via proprietary client (Outlook), web (OWA) or handheld.

Imagine the cost savings, your field and office personnel all use the same method of getting access to all forms of communication. Learning curves get flattened quickly, user productivity rises and users are happier while their frustration levels drop.

Most VoIP systems will offer such integration out of the box with little configuration. A big player, as I mentioned in an earlier article, is Microsoft. Microsoft’s strategy is to leave your phone system as it is and still enable VoIP and Unified Communications via software; Moto – VOIP as you are. Microsoft has already signed up vendor partners like Nortel, D-Link and a few others

Session Initiation Protocol (SIP) is a signaling protocol that is used by technology products for creating session-oriented connections between two or more endpoints on an IP network. These endpoints could be IP telephones, instant messaging clients, or collaborative multimedia conference applications.

Many businesses are sipping it up…(No pun intended) What I mean is, they are taking their time adopting to the technology and for good reason. Although SIP technology is cheaper (In some cases free), the cost of upgrading infrastrucure to support Quality of Service (QOS) can sometimes outweigh the savings.

While the protocol is now “final” enough to build sophisticated telecom systems using SIP, the Internet Engineering Task Force (IETF) working groups continue to forge new ways on how to apply SIP. One way that has been making it’s way to the top is SIP Connect.

What is SIPconnect?

SIPconnect is a standards-based method of interconnection between IP PBXs and VoIP service provider networks. It specifies the architecture, required protocols and features, and implementation rules necessary for seamless peering between IP PBXs and VoIP service providers.

Let’s take a look at the various applications: 

For end users in the small business or large enterprise, SIPconnect eliminates, or greatly reduces, the need for a costly gateway at the end user’s site.  In addition, new features from the service provider or the IP PBX vendor will be delivered more quickly.

For VARs and Interconnects, the SIPconnect compliant service provider will handle the complexites associated with connecting the SIP Trunk to the legacy TDM world.  This eliminates the need for a lot of integration and troubleshooting work and allows the partners to focus on other, revenue generating activities.

For Equipment and Software Vendors (Application Servers, IP PBX Vendors, SIP Proxies), SIPconnect greatly reduces the time and resources required to verify interoperability in the network.  This single item, interoperability, currently drains many man hours from the services and equipment providers that could be better used creating new and more exciting features.  This in turns reduces the amount of revenue that could be realized from those new features.

The SIP Trunking Service Provider that uses SIPconnect SIP trunks realizes higher revenue streams much more quickly.  Network services are rolled out more rapidly, greatly reducing the time to revenue for new services and features.  In addition, the service provider can greatly reduce the time and staff required to complete interoperability testing.

In the meanwhile…While service providers like Cbeyond continue the slow and steady rollout of SIP connect sit back, relax, and take a SIP!

Let’s face it, if Microsoft builds it, they will buy it. Partnered with Aastra, D-Link, Quanta, and Uniden, Microsoft has developed complete end-to-end VoIP phone systems based on Microsoft Response Point software platform. Released on November 12, 2007 MS has finally begun their journey into the SMB VoIP market a voip PBX appliance that is still very limited in it’s capability at this time. Although late in the game, I am sure they have plans of dominating the market in the near future.

It currently supports  up to 50 users with analog trunks. The future will support SIP trunks with no plans for digital trunks such as T1/PRI made known at this time. It does however boast superior voice recognition technology. Most of the end user menu options are accesible via a voice driven menu at the touch of the trademarked “ResponsePoint” button. Check out this short demo video to see it in action: http://video.msn.com/video.aspx?mkt=en-us&vid=2a5ff762-46e8-40cd-8cee-aad6087111c4.

If not a pbx solution for the business, I think it would make an excellent solution for the home. Check out some images here: http://blog.tmcnet.com/blog/tom-keating/voip/microsoft-response-point-voip-for-business.asp

VoIP requires robust infrastructure that has to be operational 24×7. If your email goes down for 10 minutes it’s not as critical, but if your voice services go down for 10 minutes, it’s a huge deal!

Keeping in mind how critical voice communications is, it is essential to plan the network accordingly. Don’t go cheap on the essential infrastructure components like network switches, routers and even cabling. Most companies tend to use the cheapest contractor for cabling when it can be one of the most critical components of the network that can haunt you for years to come. Make sure who-ever installs your cabling tests and certifies it for the amount of data that will be pushed through it.

Use intelligent switches that support multiple VLANs and QoS. This will come in handy to segregate different types of traffic, like data and voice traffic and prioritize voice traffic. VLANs allow you to have one physical network but logically have it split into multiple, so your voice can travel on VLAN1 while your data travels on VLAN2. You can have other services like video travel on VLAN3.

This not only provides your users a better experience for different voice, data and video services but also will assist in quicker troubleshooting of network issues as they arise and allow for testing of new services before deployment.

Remember, your entire infrastructure has to perform together. You can’t have one switch supporting VLANs while others don’t and you can’t have switches supporting VLANs while your routers don’t. Make the investment in the beginning and enjoy a successful roll-out instead of having to patch things on a routine basis later on.

If you don’t have in-house expertise for some of these technologies, it makes sense to go to a third party. If your planning is right from the beginning, it will be easier to scale your network and provide the quality of service your users expect.

Remember, you’ll have to provide POE (Power over Ethernet) for your VoIP nodes (handsets). You should have power backup on the POE circuit so your phones will stay operational even if there’s a power failure, users will expect this.

Recently, I talked about how some ISPs tend to block SIP traffic. If you have a branch office that does block SIP traffic, then you have the logical option to change the SIP port on your devices. This however, is not always feasible. Next best thing would be, if you have a VPN between the branch office and main office, to create static routes in your routers to route all SIP traffic over the VPN tunnel.

You’ll notice ISPs blocking SIP calls usually do this by blocking SIP port 5060 so if you change the port on your phone system to something else, and configure your end nodes to use the new port, they will work. This traffic will be visible to your ISP though, so the better option will be instead of changing ports to just route SIP traffic over you VPN.

Always check your local laws as I am no lawyer so any ideas you try, you are doing at your own risk.

I can’t believe ISPs are actually thinking they can hold the flood back by simply blocking SIP port, such ISPs are truly grasping at straws to keep their ancient model in the market.

For small business, investment in a phone system (be it VoIP or traditional) is critical and usually large. True, some VoIP systems can reduce this upfront cost, but you may end up sacrificing on feature set if you go with the wrong provider. There is an alternative; if you are just starting out or a branch office is just starting up in a new market, consider hosted PBX service.

There are many companies now providing hosted PBX services. These services are seamless and you still have a good VoIP phone at your desk. Usually these companies will give you a control panel where you can create extensions, handle call routing, messages and on hold music along with a host of other features.

You really don’t sacrifice feature set by going the hosted route. The thing to consider is how much does the service cost over a long period of time. Since hosted services come with a recurring fee, over time this could add up, but for a quick setup in a new market, I believe it’s a life saver.

There are a lot of companies out there providing hosted service, a Google search provides a lot of hits. I have had experience with Aptela and I can’t say their service has been flawless, but it’s been acceptable. Just remember, since hsoted VoIP requires internet access, if your internet goes down, so does your phone service.

We know Cisco is well entrenched in the enterprise, and now it is going full steam towards SMB space.

It will be interesting to see how the Fonality’s and Talkswitch’s of the world fair against Cisco who wants to compete and is showing that it’s ready by introducing very competitively priced products like the  Cisco Unified Communications 500 Series. We know if Cisco is ready to take over the SMB market, it has plenty of resources to do so and has the financial strength to stay the course as competitors are pushed out. Here are some recently released Cisco products for SMB market:

• Cisco Unified Communications 500 Series adds 32- and 48-user systems. The initial release of the Unified Communications 500 Series addressed the unified communications needs of small businesses with eight to16 users. These two new platforms allow small businesses to support up to 48 phones per location.
• Cisco SMARTnet Service for SBCS is Cisco’s award-winning technical support service program, offering multilevel service options and access to an extensive range of technical resources. Sold by a Cisco channel partner, customers get direct, anytime access to Cisco engineers for configuration assistance, problem diagnosis and recommendations through 24-hour telephone or Web-based support. The offering also includes software assurance with application and Operating System software updates and upgrades.
• Cisco Catalyst® Express Ethernet Switches provide flexibility of configuration to meet different networking requirements. The four new switches offer 24 10/100 access ports with Power over Ethernet (PoE); 4 10/100 access ports with PoE and 20 access ports10/100; 24 ports of Gigabit Ethernet; or 24 10/100 access ports.
• Cisco Mobility Express: Cisco has added ease-of-deployment features for the Cisco 521 Wireless Express Access Point and Cisco 526 Wireless Express Mobility Controller. They include an easy-to-set-up guest access portal; automatic policy configuration for voice, data and guest access; simplified security set-up with preset low-, medium-, and high-security options; and integration with Cisco Configuration Assistant.

Since Cisco has already proven itself in the enterprise market and SMBs definitely recognize the brand name, I believe it’s just a matter of time, and if Cisco stays the course, it will take over the VoIP phone system market in this space.

Here’s the news regarding SMB from Cisco.

It is tempting to assume that you can simply implement VoIP on your existing network, as most vendors claim, and everything will just work, this is not the case.

For VoIP to be successful you have to have a certain level of service in place. Unlike data packets, voice packets must make it to their destination, if a packet is lost, it cannot be resent without repercussions.

So before you start planning your VoIP roll-out, do some home work, you might have to upgrade your network switches to prioritize voice traffic, you may have to upgrade your routers or even network wiring.

Here is an interesting article from Network World about VoIP analysis tools that will help you along the way.

Remember, not all switches/routers are the same; There is a difference between a high dollar (HP Procurve, Cisco, etc.) switch and a low cost (Linksys, D-Link, Netgear, etc.) switch. I know both might say they support 10/100 MBPS, both might be managed but if you test both side by side in a lab, you will see the difference.

Always have some sort of network traffic sniffer/analyzer in your toolkit, sooner or later, you will need it. Ethereal is a good opensource network packet analyzer to have in your arsenal. Ethereal will let you see the traffic flowing back and forth on your network so you’re not running blind when problems arise with voice quality and end node connectivity.

Know what codec your VoIP devices are using, all codecs are not the same and voice quality along with bandwidth usage is heavily dependent on codec used.