I know, saving on long distance is just a negligibly small portion of the total saving to be enjoyed by switching to enterprise VoIP but how can you really save on long distance?

So all of the systems that we have deployed, what was the easiest so far? It appears that the best experience so far has been with Allworx. We are not yet partnered with Allworx, but it is on a priority list.

Out of all the systems, Allworx setup was the easiest and quickest to deploy. No issues with phones finding and connecting to system at all. Nice work Allworx!

There are many options available if you are in the market for a VOiP based phone system, everyone wants to sell you a new system and do a rip and replace as far as your trusted traditional phone system goes.

When you want to IP enable your current phone system, there are not many options. Replacing the entire system maybe cost prohibitive and your phone system may be old enough to where you cannot add IP adapters to it.

This is when Microsoft’s Unified Communications strategy might be able to aid in rescue. Instead of having to rip and replace your existing phone system, would you not use software to simply IP enable your voice infrastructure? Microsoft Office Communications Server 2007 is designed with that in mind.

You can integrate your current phone system, with whatever type of Telco connectivity it has to OCS. You can then have your users connect to OCS and unlike regular VOiP systems, you get enhanced IM functionality with presence information embedding in office products (Outlook, Excel, Word), you get VOiP connectivity from handhelds, you get video conferencing capability and Microsoft has tweaked the audio codecs so they perform really well in low bandwidth situations.

To read up more about Microsoft’s Unified Communications strategy, you may want to visit http://www.microsoft.com/uc if you are interested in finding out more about OCS 2007, you can visit http://www.microsoft.com/ocs or reply to this article with questions.

You probably haven’t tried to configure 3CX with Cbeyond’s SIPConnect service, as it’s not publicly supported yet. I’m testing a 3CX connection to Cbeyond’s SIPConnect  as a trial and here are my findings so far;

First of all, it was the simplest easiest SIP Conversion I’ve ever done. We were done in about an hour from start to finish. Keep in mind my 3CX system was already running, extensions were already setup, the only thing I didn’t have was a connection to the PSTN.

In 3CX you should see Cbeyond as a VOIP provider, if you don’t see this option, then first thing you should do is upgrade your 3CX to the latest version. Once you do that, you can use the wizard to add Cbeyond VOIP.

You’ll notice that the wizard auto fills in the SIP proxy addresses, these may be different for your area. Since this option is dimmed out, you need to leave the default addresses, click Next, then click Back, now you’ll notice they are not dimmed anymore! You can now enter the addresses you got from Cbeyond.

Follow the wizard providing your primary phone number as the authentication ID and username along with your Cbeyond provided password. Once configured, go back into the SIP settings for the connection you just created and make sure you have g.711 Ulaw codec selected.

At this point you should be able to register with Cbeyond. Once registered, if you are not able to make outbound calls but your call rules are setup correctly, go into VOIP provder settings and go into Gateway/Provider Outbound Parameters. Now change the “from: user part” to “LineNumber” external number of line. You should now be good to make outbound calls. Reasoning for this is, by default 3CX sends you extension to Cbeyond as the number trying to make the call, and as Cbeyond does not know or care about your local extensions, it denies access. Changing it to the external number lets the call go through.

I will be testing the system over the next few days and will report back. On a side note, sound quality is impeccable over SIPConnect.

You probably have an idea by now how I don’t like Skype, well today I’ll go into some of the details as to why I don’t like it.

Skype is a peer to peer application and under some circumstances, the Skype client uses the specified bandwidth of between 24 and 128KbPS. If you have a high speed internet connection and your firewall is not very restrictive, your PC with Skype might become a “supernode”. In this mode, Skype starts to relay signaling for other Skype users that are behind NAT (network address translation).

You can test this behavior. Run a packet capture on your network before installing Skype, then capture packets after installing Skype. You’ll notice that if you are directly on a publicly routable IP address, your machine starts using a lot (if not all) of your available bandwidth. You’ll notice traffic going out to all sorts of addresses around the globe.

The bandwidth usage can be so considerable that in certain circumstances it has brought down Linksys routers (typically used as residential gateways).

Skype uses both TCP and UDP ports for this traffic and in tests can use up all bandwidth in a 100MbPS connection. Now think about multiple users running Skype (not even using it) at the same time.

So until Skype starts to ship with some sort of a management tool to enterprise to limit this “supernode” mode, I will not recommend anyone use Skype, besides it’s not like it’s free. Why would you want to use Skype when you can use other VoIP services for free!

Lets face it, most small businesses won’t be implementing an elaborate voice system comprising of various components like Cisco Unified Communications Manager, Cisco Unified Communications Manager Express, and all the related hardware and additional software.

To ease the implementation for small business and lower the cost while providing more value, Cisco has come out with the UC500 appliance which does not require a Windows Server (it’s stand alone) and can be bundled with a POE Switch, WiFi Access Point and Voice in a nice, tightly integrated package.

It is inevitable, companies will be looking to integrate UC500 with their Exchange 2007 infrastructure. The good news is, UC500 can work with Exchange 2007 UM with some configuration. You have to make sure the UC500 can see the Exchange 2007 server, you have to allow h323 to h323, h323 to sip, sip to h323 and sip to sip. You have to setup dial-peer and define the IP of the Exchange 2007 server. Remember the codec is g711ulaw, and you should be good to go.

The actual configuration can require a little effort but is easliy doable, remember to apply Exchange 2007 SP1 before even trying!

If you’ve worked with VoIP, then you already know the tremendous savings you can enjoy. There are numerous companies, campuses and organizations that have enjoyed tremendous cost savings, boost in productivity and even entered new markets because of this technology.

A huge cost saving and productivity boost can be gained by integrating your CRM system with your phone system. This is especially true, if not a requirement for any call center. Now you can automate processes that you had to perform manually and eliminate procedures that were prone to human error. For example, a customer calls in to a call center, the customer caller ID is logged and matched up with CRM system that automatically pulls up account information. The call center agent types in the details of the call without having to hunt for customer specific information as it’s already on the screen.

Think about the up-sell potential if you can see past order history immediately as you pickup a call from a customer, you can also convert disgruntled customers to loyal ones by taking care of issues they’ve called about in the past. A simple thing like wishing someone a happy birthday can go a long way, only if your phone system is integrated with your CRM system.

Almost all VoIP phone system allow simple integration with CRM systems. These phone systems make calls to the CRM system via simple URL requests, which means integration is easy and you can connect with almost any modern  CRM system. Saleforce.com even allows integration with Skype!

What is the difference between Comcast and any other VoIP phone service? Granted Comcast VoIP is certified in Harris county to carry fire alarm calls, beyond that it really doesn’t matter whether you are using Comcast’s phone service or if you sign up with a third party like Lingo or Vonage.

Out of all of these providers, I’ve found Lingo to provide the best service and best rates, especially for international calls. Keep in mind, your phone service is only going to be as good as your internet service. If you experience latency issues or huge lag times, as many Comcast customers have experienced, then VoIP may not be such a good idea.

Here’s a quick test, if you have a windows machine click Start, Run and type in CMD then press enter. At the prompt, type in “ping www.google.com -n 30″ without the quotes.

Note the column that starts with “time=”. If these times vary too much and are really high like 190ms or higher then VoIP may not be a good idea. If you get any “request timed out”, then again VoIP may not be a good idea.

I switched from Roadrunner (before it became Time Warner and then Comcast) to a Cbeyond BV1 package because of up time issues and response times. The BV1 package comes with a service level agreement so it’s not just a best effort service but a guaranteed service.

Thanks Naveed for bringing ooVoo to my attention. ooVoo is a great service for video conferencing and voice calls.

ooVoo is in beta right now and offers free voice calls to landlines in USA and Canada. So, for all you road warriors, a nice utility to have for video conferencing and voice calls. You can have up to six people in a video conferencing session and the quality is excellent over most broadband connections.

Contacts can be imported from most applications like Outlook, Gamil, MSN Hotmail, Yahoo mail, etc.

You can also record video conferencing sessions, a great idea for training and record keeping. Try it yourself at http://www.oovoo.com.

An interesting feature about video conferencing with ooVoo is the ability to have video effects, so you can grow a virtual mustache or glasses!

Network load sharing and load balancing in most cases are both good things to have on a data network, but how does this effect your voice newtork?

Before rolling out VoIP on your highly redundant data network, you need to make sure your load sharing does not allow voice packets to take different routes to target and cause packets to arrive out of order. These out of order packets are discarded by VoIP systems and this sequence of events can cause high level of jitter and constant call quality issues. So if you have constant call quality issues, capture some voice packets to see if some device is trying to be too “intelligent” and trying to load balance/share to accomodate high bandwidth usage.