How VoIP Saves You Money

We're all trying to save money these days and taking advantage of VoIP technology is one painless way of doing so. Not only is it financially painless but it can be more reliable than analog phone service.

If you look on your phone bill and you'll see taxes, fees and regulatory charges. One of those is undoubtedly the FUSF, or Federal Universal Service Fund. The FUSF is a charge the government levies on phone companies in order to help provide phone service to everyone, even those that cannot afford it. Most phone companies pass that cost along to their customers. Recently the FUSF has taken upon itself to help pay for cell phones for low income people. That's a noble goal since cell phones have saved many lives but why in the world should another tax be levied on you? VoIP saves you money partially because it does not have a FUSF tax. Sure, it's only about $3.50 per month but that's $42 a year!

You don't need special equipment with VoIP, no telephone handsets or headsets. You may need an Analog Telephone Adaptor (ATA) which is usually provided to you by the service you sign up with. Use your standard phone if you like; it converts your analog voice into a digital signal that the internet can understand and transmit along the internet through your broadband modem. ATA is a small, thin box that blends right in with your other hardware. It can be used with cordless phones so that you can use any handset in the house or office. If you have more than one computer just connect a router to the ATA instead of the PC.

If you don't want an extra piece of hardware hanging about you can download software like Skype and then you won't need any type of telephone paraphernalia. You'll simply make and receive calls from your PC in a separate window. You'll still get caller ID, voice mail, contact lists and all the other features you can have on a landline or a cellular phone.

Traditionally, analog phones only allowed 2 people to speak unless you set up conference call with all the charges that come with it. VoIP allows you to set up a conference with a whole team scattered all over the world. It won't tie up your regular phone line, if you have one, so you can surf the web and take care of communications while your teen blissfully monopolizes your regular phone line!

VoIP - Before You Implement

With VoIP becoming the preferred method of transmitting streaming conversations during a phone call, it is essential that a network be properly configured for optimum audio quality. VoIP provides flexibility and versatility as well as lower operating costs, however if the conversation isn't clear or at all understandable then the benefits to VoIP are lost.

Prioritizing voice traffic on a network is critical to quality audio during phone calls. The network will simply try to resend data during times of network congestion when transferring or sending data. Data is usually very forgiving and will pick up where it left off when the bottleneck clears. In the case of a voice conversation, this particular strategy results in garbled conversation because of latency, delay, and jitter. The voice data packets must be reassembled on the receiving end in the order in which they were transmitted. Since data is sent over multiple channels by choosing the best route based on current network traffic, it is essential that the data packets be reassembled on the receiving end in a specific order and that they are not subject to the time delays seen during times of heavy network congestion.

This is accomplished through network prioritization. All voice traffic is given priority on the network through the use of QOS. This configuration acts in much the same way as a police officer directing traffic in a busy city. When the amount of data comes close to or exceeds the available bandwidth on a network, QOS will stop all data traffic while allowing voice traffic to continue on. In this way, you're telling your network data packets that are designated as voice packets are always given priority for routing.

Before introducing VoIP technology into your network, it is essential that a network assessment be completed prior to implementation. Depending on the expected call volume and the current level of utilization on the network, it might be necessary to add additional bandwidth to accommodate voice traffic. Depending on the average number of calls per day, you could see a significant demand on network bandwidth. If the amount of bandwidth is inadequate, voice quality will suffer with garbled audio and poor performance.

Although inadequate bandwidth is a common problem with VoIP audio quality, there are other factors in the network that can affect the quality of a VoIP call. Some networks simply have too many hops or transfers from router to router. Each of these hops can create delay because each router has to read the packet to determine its destination and then route to the next point in the route. Obviously, the greater the number of times that this occurs the greater the delay the packet will experience before reaching its destination. Whenever possible, VoIP traffic should be routed in as direct a line as possible with a minimal number of hops. This can be accomplished by designating a VLAN or virtual local area network exclusively for VoIP traffic. The VLAN can be configured to route voice data in a more direct line than the rest of the data being transmitted on the network.

An additional consideration is the codec used for audio compression. Generally, the standard for voice signal compression is a codec of G.729. This is the most widely used codec to maintain as much audio quality as possible while minimizing the amount of bandwidth needed. Another codec commonly used is G.711. Since this codec has a higher rate of sampling the audio quality is equivalent to that which you would hear for a standard analog signal. However, this particular codec requires a great deal of bandwidth and some networks simply cannot afford the bandwidth required.

Another consideration is connectivity to the phone system itself. Another advantage to VoIP technology is the elimination of the need for dual wiring. Since both the phone and the computer can share the same network connection, only one cable is required at each desk or station. This minimizes infrastructure cost when implementing a VoIP solution. Older digital phones require their own wiring to a telephony backplane. This usually involved "punching down" on an old-style 66 block or 110 block to provide connectivity to the phone system. IP phones can be plugged into the cat 5 patch panel just like a computer. Additionally, having a POE enabled data switch negates the need to have external power supplies to power the phones. POE, or power over Ethernet provides both power and network connectivity to the phones.

VoIP technology has many advantages for large and small businesses alike. However without proper implementation and configuration VoIP technology can suffer from poor audio quality and overall poor performance. This is why it is important to review your current overall infrastructure configuration and capacity prior to implementation. If this important step is left out, your experience with voice over IP technology won't be as successful as you hoped.

Can VoIP Be Reliable?

VoIP has many advantages that make it an appealing solution for many business owners. The idea of transmitting voice traffic over an existing data network at a significant reduction in cost has become too hard to pass up for most business owners. However, few realize the potential limitations or issues that can arise with call quality when using VoIP.

This is not the fault of VoIP itself. In a perfect world voice and data traffic are happily traveling through the network side-by-side without interference or interruption. This can work, however this utopia is not achieved without careful consideration and network analysis. This is because when you start sending voice traffic over the data network it now has to compete with other data to reach its destination. If the network does not have enough bandwidth, or has too many hops, the speed with which the voice packets reach their destination can be slowed down considerably. Each hand-off of the packets takes extra time and slows the overall transmission speed of the packets. With a voice conversation, this delay is unacceptable and results in a garbled conversation that is difficult to understand, often sounding like a cell phone out of range.

Two key components of an optimal performing VoIP system are capacity and redundancy. Obviously, the network should be able to accommodate the current call volume as well as the bandwidth requirements that are used for data transmission. Determining the required bandwidth requires taking a careful look at the current call volume as well as the utilization on the network. If the network is experiencing issues or slowdowns without the addition of VoIP traffic, it stands to reason that these issues would need to be resolved before adding the demand of VoIP. Any additions or enhancements would need to have an estimate of bandwidth needed for voice traffic. Without the proper capacity on the network, the VoIP quality will suffer leading to frustrated employees and customers alike.

Redundancy allows for a backup to be in place in the event that any component or piece of the production system should fail. This could be hardware in the phone system, hardware on the network as well as the trunks that lead to the outside world. Any failure could result in an a major outage that takes down communications. This is why redundancy is such an important aspect of network configuration and setup. Having a single point of failure can be quite costly when you consider loss of revenue due to business grinding to a halt.

Most phone systems have dual hardware to handle the processing of VoIP calls in the event that one of the components should fail. Theoretically, failover should occur without much fanfare. However, there is always the possibility that calls could be dropped during the failover process. If any calls are being processed by the failing hardware, the call would be unable to proceed. Any new calls however, would automatically be processed by the redundant hardware.

This same concept holds true for the data equipment where one router or switch will pick up where the other failed. This essential for maximum system availability for any network and doesn't just apply to VoIP traffic. Any company that does not have backup hardware in place may find themselves in a tough spot should the data hardware go down. Not only does it stop the transmission of data, but halts all VoIP calls as well, effectively leaving the business with no way to communicate.

Some communications outages are due to a loss of connectivity with the carrier. Weather or some other disaster can bring outages to the carrier's network. Fiber cuts are common where construction is occurring and have been known to take down communications for a large number of the carrier's customers. For this type of scenario, having an alternate carrier is a good way to insure that your business always has a connection to the outside world. By using multiple carriers, the business is not at the mercy of only one carrier and you will have options in the event you have to reroute your network traffic. Since VoIP travels over the network, the familiar expression "the network is down" takes on a whole new meaning. Not only will you not be able to communicate via email, but your phones are dead in the water as well. To resolve this issue, you configure your data equipment to begin routing traffic over the backup carrier's communications if the primary is inaccessible. Having this setup insures that you always have communication and that data and voice traffic will continue to route.

System availability is the key to any successful IT solution. VoIP traffic is dependent on network availability as well as capacity. Additionally, configuring the network to properly handle VoIP traffic is essential for a successful outcome. VoIP technology can be a lower cost reliable solution when the proper measures are taken to insure maximum availability.

What Is a VoIP Gateway?

VoIP technology is fast becoming the new standard for communications. Although there are still a number of people using the standard analog signal over copper wiring, there are many people venturing into VoIP with the hopes of cutting costs on their phone bill.

This is because VoIP technology converts the analog signal to data packets that are sent over a broadband network in the same way that data is sent. This bypasses the PSTN which bases its cost on the distance between where the call originated to its destination. Data does not have these restrictions and in most cases the data packets are sent on different routes and reassembled on the receiving end. Since many data packets from many data transmissions can occupy the same trunk at the same time, the cost of using the trunk is spread out over many users. With standard analog phone conversations, the line is limited to one call and is not available to any other calls until the conversation ends, so the person initiating the call is shouldering the entire cost.

Using true VoIP is when the caller initiates a call and the destination number is also using VoIP so that the entire call is transmitted over the data network in the form of data packets. But what happens if you were to call someone still using a standard analog phone? This is where a VoIP gateway comes into play. A VoIP gateway essentially acts as a translator between the old PSTN network and newer VoIP technology.

If a call comes into the gateway and is in the form of voice data packets, the gateway will decompress the conversation and convert the voice signal to digital and then to analog to be sent over the PSTN. If the call comes into the gateway as an analog signal, it is then converted to digital and then compressed using a codec to break up the conversation into packets to be sent over the data network. The gateway in simplest terms is nothing more than a media converter that transitions the call to whichever format is needed to be understood on the receiving end. As more and more people convert to VoIP technology, there will be less need for this conversion process to take place.

The codec used is dependent on the audio quality desired and the amount of bandwidth for voice data transmission available on the network. There are several, however the most common are G.711 and G.729. The G.711 usually provides a better audio quality but also requires a much larger amount of bandwidth to support. The G.729 is probably the most popular because it provides the best compromise in regards to audio quality with minimal bandwidth requirements. The G.711 takes 64 Kbps versus only 8 Kbps for the G.729. Most VoIP gateways support both codec's.

There are also various VoIP protocols supported by most gateways with the most common being the H.323 and SIP protocols. However, H.323 is on the way out as the newer and more flexible SIP takes its place. SIP or session initiation protocol is a text-based protocol that controls the initiation and break down of a VoIP session. It allows for multiple technologies to be present in one session and is making the convergence of different multimedia such as video, voice and instant messaging possible.

In most cases, the gateway is used in one of two ways to allow intercommunication between VoIP and non-VoIP hardware. One way is to use a gateway to convert incoming calls from the PSTN to VoIP calls so that they can be received on a VoIP system. Another way is to connect a standard phone system to the network so that calls can be made via VoIP. In both scenarios, they are generally setup using the protocols and codecs described above.

Gateways can come in either software or hardware form. Hardware based gateways are more popular and can either be a standalone box or could be a card or module added on to existing equipment. Software versions are set up on a server where existing servers can be leveraged to run the gateway software in addition to other services. As a general rule, the software solution is a bit cheaper than its hardware counterparts.

The gateway is only part of the equation and performance is based on many factors such as available network bandwidth, security, protocols and codecs. All of these different aspects of VoIP configuration are going to play a role in the quality and reliability that one could expect. Since most VoIP calls today still interact with the PSTN, a VoIP gateway should be included in any VoIP communication solution. Before deciding on a hardware or software solution, determine if there are already resources available that can be leveraged to keep costs down. This way you can you still take advantage of VoIP while waiting on the rest of the world to catch up.

VoIP Security - How Secure Is It?

With the advent of VoIP technology, the question of VoIP security comes to mind for many who are afraid that their conversations are being listened to by people they don't even know. Although this is certainly a possibility, so far there haven't been any widespread reports of this occurring. It might surprise you to know that your phone conversations on your landline and your cell phone can also be tapped into, just in a different way.

Since VoIP conversations are converted to digital data and then transmitted in the form of data packets, just like traditional data packets they can be intercepted and listened to. This might be a bit disconcerting to hear, but in many cases the security breach comes from the user themselves. If making calls using a computer, there could be malware present that is sending these packets to a server over the internet. Once there, the hackers then listen to the conversation.

Why would someone thousands of miles away want to you hear your conversation? Well, people that write intrusive software or malware are generally only trying to gain one thing, your private information. Although we benefit from new technologies such as VoIP, unfortunately so do would be thieves. It's not out of the realm of possibilities for malware to be written to "listen in" to your conversations and retrieve the data based on recognized key terms such as "pin number". If you called your bank and spoke with a bank employee, your private information could get into the wrong hands.

This sounds like a frightening prospect, but there are ways to combat the theft of your voice conversations. Encryption is key to prevent anyone from having unauthorized access. This way the conversation is encrypted when it's converted to digital data which renders it useless to anyone who might try to gain your information. They can still get the data, but since it's encrypted they would have no way to listen to it. For older style phone systems that use digital phones, for someone to listen to your conversation they would have physically be at the location to "tap into" the line. VoIP conversations can be intercepted from anywhere in the world which is why encryption becomes so important.

It goes without saying that the data network itself should have the protection of firewalls. Firewalls are the gatekeepers of a network and will allow or deny traffic depending on the configuration. The firewall serves to act as the main entry point for data. By relying on a firewall, the need for each individual device to have its own security is negated and administering security on the network becomes more simplified. Unfortunately, the firewall becomes yet another point through which data packets must pass. Since all of the packets enter and leave the network through the firewall, congestion can become heavy and a bottleneck created. Some networks are configured to have two firewalls one for standard data and one for VoIP data. This way the firewall specific to voice traffic can be configured with the appropriate security settings that would apply only to VoIP packets. It is also recommended that a firewall maintain the QoS or routing priority information on the packet header. VoIP traffic must have priority throughout the network to maintain higher voice quality during phone calls.

There are several places within the network where it is appropriate to implement firewall protection for added VoIP security. The first is in front of the IP phone system itself. This protects the phone system from any attacks that might originate from with the local area network and provides additional protection. VoIP traffic from remote workers should also be closely monitored to ensure that viruses, worms or other digital nuisances don't travel to the IP PBX via the VPN connection. Lastly, monitoring should occur on the carrier side to monitor for attacks from the carrier network itself. This is true even if the trunk is not connected to the internet.

There are several ways to configure multiple firewalls so that one is specific to VoIP traffic while the other handles other data. They can be set up in line with the data firewall being the first line of protection and the VoIP firewall immediately behind it or the data firewall can handle data traffic on its own while still processing VoIP traffic after it has been filtered by the VoIP firewall. The last configuration is one of both the data and VoIP firewalls sitting behind a router where the data firewall and the VoIP firewall filter data exclusive to their individual purpose. Regardless of the configuration, having a separate VoIP firewall is another layer of security protecting both your voice conversations an IP phone system.

VoIP is innovative technology whose time has come. With the proper configuration and security settings, VoIP telephony can be just as secure as telephony using circuit switched routing. VoIP security is easily achieved with planning and appropriate security measures.

Getting Rid of the Telephone Number

How many of us remember the phone numbers of even five of our friends? When you make a call on your smartphone, you don't type in the phone number - you click on the person's name instead. The only reason why phone numbers are still important is because we have regular phones where you actually have to dial a number and because it serves its purpose as an identifier when you want to give it to someone else.

But while the telephone number has served us faithfully for decades, it's increasingly at odds with the way identities are managed in today's world. For one, the phone number is a model of inflexibility. Whenever you change your device, you have to struggle to keep your existing phone number. It's not easily accessible from another country and you have to pay huge fees in order to continue operating as usual. Compare this to a modern form of addressing like email. You never have to change your email ID when you change your device and moving to a new country is irrelevant in the context of the Internet.

The two networks - phone and Internet highlight how much the world has changed in the last decade. The Internet is a pan global network which behaves the same no matter where in the world you are - and this is how an addressing system should look like. When you ask someone for their "number," you're actually asking for a handle on which you can reach them. A telephone number is notoriously fickle. Most phone numbers move around and there's no permanency.

VoIP and SIP systems are trying to change this by making a person's SIP address just like an email ID. Depending on which company you have your SIP address with, your identifier will be your username@whateveritsp.com. Such things are easier to remember as well. Telephone numbers with their numeric systems are like IP addresses - meaningless. But an SIP address is like a website - a branding tool that also makes sense and is easy to keep track of.

Right now the biggest hurdle to the migration to SIP addresses is that services have poor or no interconnectivity. The day a VoIP service's customers can communicate with those of any other VoIP service in the world is the day VoIP would have truly arrived. There are numerous problems along with efforts by the telcos to prevent such an inter connected directory, but we're advancing slowly and the time isn't far off when telephone numbers will be a thing of the past.