Something about VoIP

Is It the Network? Solving VoIP Problems on a Wireless LAN.

2007-04-17T21:30:00.000+04:00

Introduction
The requirements for a network administrator have always been diverse. Get the users connected, make sure there’s enough bandwidth to run applications, keep the network secure, and so on.

When problems come up on a traditional network, the solutions have often been fairly straightforward. If users aren’t connected, you connect them. If there’s not enough bandwidth, you buy new equipment. If network resources are vulnerable, you introduce appliances and applications that provide enhanced security. It’s a pretty tight loop of problems and solutions.

In today’s networking environment, things have changed. You can save money by putting telephony on the network. That means VoIP. You can increase productivity by allowing wireless access to the LAN. That means Wi-Fi. Things are great until you have to combine the two. That often leads to problems.

Managing a network that includes VoIP and Wi-Fi means more than just learning the black magic that makes each technology work. It also means learning how each technology causes the other to get a little blacker and a little more magical.

The goal of this paper is to combine a deep knowledge of 802.11 protocols, quality of service (QoS) mechanisms and VoIP handsets with the feedback aggregated from training hundreds of IT professionals each year to identify potential problems and solutions when rolling out VoIP on a Wi-Fi network.

Basic VoIP Requirements
To understand how VoIP communications are affected by wireless LANs, one must first understand how VoIP works. Detailing every VoIP protocol and the network requirements of each is outside the scope of this paper, but some basic points must be explained.

Many VoIP handset manufacturers use the G.711 codec because G.711 provides superior quality at the price of relatively high bandwidth requirements. I say, “relatively high bandwidth,” because even the G.711 codec only requires a 64 kilobits per second (kbps) data stream in each direction (uplink to the access point [AP] and downlink from the AP). When encapsulation overhead from RTP, UDP, IP and Wi-Fi headers is accounted for, the bandwidth requirement is pushed to 92 kbps. Since that fails to include control traffic, it’s reasonable to estimate
that 100 kbps in each direction – 200 kbps of total bandwidth – is necessary for each G.711 call.

In addition to understanding basic bandwidth requirements, one must also understand three important problems that may affect call quality: delay, jitter, and packet loss. Delay is the amount of time it takes the sound from your voice to reach the ear of the other person. Maximum acceptable delay limits for VoIP are considered to be 150-200 milliseconds (ms), depending on call quality requirements. Jitter is the variation in delay between packets. The jitter buffer holds packets so that they are received at consistent intervals. Significant jitter
may cause the jitter buffer to increase to the point that delay reaches unacceptable levels. Packet loss occurs when the maximum delay specified in the jitter buffer is exceeded. Packet loss above 5% is considered unacceptable when using the G.711 codec.

Ethernet to Wi-Fi
Once the fundamental principles of VoIP are understood, it becomes important to understand how wireless LANs differ from wired LANs. Perhaps the most important concept to understand is that going from an Ethernet (802.3) network to a Wi-Fi (802.11) network only affects the lowest two layers of the network. From layer three up, wired and wireless LANs are exactly identical. In the VoIP world, this means that IP, UDP and RTP are used in exactly the same manner on a Wi-Fi network as they would be on an Ethernet network. To solve wireless VoIP problems, we must look at how the intrinsic nature of an 802.11 physical layer and data
link layer will affect delay, jitter, and packet loss for VoIP communications. I’ve identified eleven (11) 802.11-specific topics that should be understood in order to resolve problems on a wireless VoIP deployment.

WLAN Capacity
The worst-kept secret in networking is that the throughput of Wi-Fi networks never comes close to hitting the advertised 54 (Mbps) rate. A far better-kept secret is why this is happening and exactly how it affects applications that run on the network.

Whenever a packet of 802.11 data is sent, a series of Interframe Spaces (quiet periods) and Acknowledgments accompany that data. In addition, Wi-Fi networks have a random backoff sequence that allows a wireless AP and the stations that connect to it to share a wireless channel. Since a detailed discussion of these topics is somewhat outside the scope of this paper, let’s summarize by saying over half of that 54 Mbps is going to be lost on your average Wi-Fi network. In fact, numerous independent throughput tests have shown that somewhere
in the neighborhood of 20 Mbps is expected from most APs under good conditions.

It is also important to consider data rate change when determining WLAN capacity.We all know that wireless LANs can send data at 54 Mbps, but that doesn’t mean they will send data at 54 Mbps. Even 802.11g handsets can send data at speeds as low as 1 Mbps, depending on the signal strength the phone receives from the AP. If a wireless VoIP user walks away from the AP during a call, or if someone turns on a microwave oven nearby, his data rate will likely plummet. Since the wireless channel is shared, handsets that transfer data at lower speeds will cause other stations to stay quiet longer, thereby slowing the entire network down to far less than 20 Mbps of aggregate bandwidth.

Even if you have a network where all stations do transfer data at a 54 Mbps data rate, calculating VoIP capacity based on 20 Mbps aggregate bandwidth can be tricky. If you were to divide by the ~200 kpbs of total bandwidth that is required for each call, you might be overjoyed to find that each AP can handle 100 calls at a time. Unfortunately, excessive delay and jitter would cause the network to be rendered unusable if an AP had 100 VoIP handsets calling at once. A typical Wi-Fi phone will require that an uplink packet and a downlink packet be sent approximately every 30 milliseconds. With 100 Wi-Fi phones sharing the network, too much time would be spent waiting for the other phones to take their voice packets over the wireless channel.

Once you begin analyzing delay limits rather than raw bandwidth limits, quantitatively determining a maximum handset capacity per AP becomes impractical. Since the random backoff sequence mentioned earlier is, well, random; we never know with any certainly exactly how much delay the network will experience between wireless frame transmissions. Sterile lab environments handling up to 50 calls per AP have been documented by Wi-Fi phone manufacturers, but those numbers fail to predict what a production environment will see.
A sound starting point is about 20 calls with 802.11g handsets or about 8 calls with 802.11b handsets. Those numbers are on the low side if the RF environment is clean, but for most areas where wireless VoIP is deployed, sticking with a conservative estimate is best.

Collisions and Retransmissions
When you start looking at Wi-Fi collisions and retransmissions, it’s easy to become alarmed no matter what WLAN capacity your network was designed for. Here are a couple of fun facts that might scare IT professionals:

5% is the maximum amount of acceptable packet loss, but Wi-Fi networks routinely see over 10% collisions.
Anything over a 200-millisecond delay can compromise call quality; after a collision, Wi-Fi retransmissions are often sent at a lower speed.

The basic facts about collisions and retransmissions may initially seem like a cause for alarm, but upon closer inspection the situation looks less ominous. Wireless collisions occur at layer two of the network. Packet loss limits for VoIP are calculated at higher layers. An RTP-based VoIP application views layer two only as a dumb data link used for transporting pieces of a conversation. In other words, VoIP doesn’t care what happens at layer two, as long as the data moves from one side to the other at a high enough speed. If a few Wi-Fi frames fail to reach their destination, those frames will be retransmitted immediately. Since 802.11b and 802.11g VoIP frames typically take far less than one millisecond to traverse the wireless channel, even 10% collisions usually means that wireless VoIP conversations will work consistently.

A similar principle applies to wireless retransmission. After a Wi-Fi collision, the Wi-Fi frame must be retransmitted. If the frame must be retransmitted once, it often is sent at the same speed as the initial frame. If the frame is retransmitted multiple times, lower speeds may be used to give the frame a better chance of reaching a distant AP or station.

Even though this drop in speed may appear to be a problem, the fact that layer two is transmitting frames on the microsecond level rather than the millisecond level means that some lower speed frames can be handled. To put it in mathematical terms, a 230-byte VoIP frame sent at 54 Mbps will take 0.034 milliseconds to go from your phone to the AP. Even if retransmissions cause your phone to send that frame at 24 Mbps, it will still only take 0.077 milliseconds to reach the AP. Both of those numbers are so far below the 200 millisecond delay threshold that the VoIP application will never even know that there was a problem.

Even though problems due to collisions and retransmissions are often exaggerated, that’s not to say that they should be ignored entirely. Collision statistics can be tracked using an 802.11 protocol analyzer like Wildpackets Omnipeek or AirMagnet Laptop Analyzer. By observing the percentage of retry (retransmitted) frames on the network, you can see how often collisions are occurring. Gauging the appearance of lowerspeed frames due to retransmissions can also be done using a protocol analyzer. Statistics indicating the number of frames sent at each Wi-Fi data rate are available in both of the aforementioned products.

If excessively high Retry percentages are seen or if large amounts of data send at low speeds is observed, there may be a problem with the network. When such problems do arise, the first place to look is the configuration of the APs, controllers, and servers that comprise the WLAN infrastructure.

Configuring the WLAN Infrastructure
The proper configuration of WLAN controllers, APs, and servers is important for any wireless network, but when VoIP is deployed, there are a few extra configuration settings that should be closely monitored. Every Wi-Fi network will have parameters for configuring the Service Set Identifier (SSID), authentication method, encryption cipher, and channel number. The principles for configuring those settings are the same whether wireless VoIP is being used or not; therefore, they are outside the scope of this paper.

It may seem counterintuitive to think that the way you configure the AP could affect the battery life of wireless VoIP phones, but there are two configuration settings that do just that. The Beacon Interval and Delivery Traffic Indication Message (DTIM) Period are two settings that will affect the power consumption of VoIP handsets if 802.11 Power Management is supported.

When 802.11 Power Management is enabled, stations will allow the wireless radio to sleep between DTIM Beacon frames. While the station sleeps, the AP buffers any data that needs to be sent to the station.

802.11 Power Management does little to help battery life when a VoIP handset is actively on a call (hence the movement towards Wi-Fi Multimedia (WMM) Power Save, which is discussed below), but significantly more power will be conserved if the phone is allowed to sleep longer when not on a call. When the Beacon Interval and/or DTIM Period settings are increased on an access point, DTIM Beacon frames are sent more infrequently. Therefore, stations that enable 802.11 Power Management will wake up more infrequently as well.

Optimal configuration settings for the Beacon Interval and DTIM Period will vary depending on the number of stations and type of traffic on a wireless LAN, but a good starting point when supporting VoIP is a Beacon Interval of 100 kilo-microseconds and a DTIM Period of 5 (Beacon frames). It should also be noted that sending DTIM Beacon frames too infrequently does have a price. Too much latency could be introduced because handsets may stay in a doze state for too long.

Optimizing battery life by properly configuring the WLAN infrastructure is certainly important but, in many cases, the most important configuration setting change when adding VoIP to a wireless LAN is the power output of the AP radios. Typically access points will allow output power settings ranging from as low as 1 milliwatt (mW) to as high as 100 mW. Many IT professionals will use a variety of power output settings when configuring a wireless LAN in order to accommodate predetermined locations for cable drops or areas with different RF environments.

Choosing various output power settings on APs may appear to be a grand idea on the surface, but when VoIP is used on a wireless LAN, call quality will likely suffer. Since VoIP phones generally use variable power levels as a way to enhance battery life, a handset that roams from a high-powered AP to a low-powered AP could see excessive packet loss while the phone adjusts its power level. In fact, many wireless VoIP deployments see excessive numbers of dropped calls because packet loss becomes too great.

Setting power output levels the same on every AP is a necessity because it helps alleviate packet loss when roaming between access points. Still, it takes more than proper AP configuration to manage 802.11 roaming when adding VoIP to a Wi-Fi network.

802.11 Roaming
Even if configuration settings for power output are set identically on every AP, 802.11 roaming may still cause problems for VoIP handsets.

All Wi-Fi devices are designed to go through a series of steps in order to establish a connection with a new AP whenever the current AP reaches an unacceptably low service level. This process is called 802.11 roaming. Roaming occurs no matter what type of security is used over the Wi-Fi network. Whether Wired Equivalent Privacy (WEP), Wi-Fi Protected Access (WPA), WPA2, or (gasp!) no security is used, there will be a delay in network communication as handsets establish a connection to their new access point.

The roaming process is easy to identify, but its effects are hard to quantify. As we discuss in the Wireless Networking II course, the 802.11 frames transmitted during roaming are well known, making statistics-gathering a snap. Any wireless protocol analyzer, such as Wildpackets Omnipeek, will allow a network troubleshooter to quickly identify how often handsets are roaming. The problem is that roaming times vary, depending on a variety of factors. Network traffic, RF interference, and device processing are three of the many factors that can cause roam times to increase to unacceptable levels.

When handling the effect that handset roaming has on wireless VoIP, it’s best to be familiar with both the network and the handset. Make sure that APs are at a good distance to handle roaming. It’s well known that great distances between APs can lead to dead spots, but it’s also important to remember that closely placed APs may cause some handsets to roam too often.

From the handset perspective, find out what roaming mechanisms are built into your handsets. Since 802.11 roaming is initiated by the station and not the AP, handset manufacturers have a significant degree of control over how their phones behave. Low roaming thresholds are generally preferable in wireless VoIP environments because that causes 802.11 roaming to occur less frequently. Some vendors have even created dynamic roaming thresholds. These dynamic thresholds increase when the handset is idle so that the handset associates to the best possible AP. Once a call begins, the roaming threshold lowers so that the call is less likely to be compromised by 802.11 roaming.

WEP Only
Dealing with network performance problems due to 802.11 roaming may be troublesome, but dealing with the vulnerabilities in wireless LAN security can be downright dangerous. In the past, most VoIP handset vendors supported only Wired Equivalent Privacy (WEP) security on VoIP handsets. With WEP encryption easily crackable using freely available software utilities, a number of organizations avoided deploying wireless VoIP because a cracked WEP key could open the network up to sophisticated attacks.

If your VoIP handsets support WEP as their only security method, there are steps that can be taken to deploy wireless VoIP while keeping the rest of the network secure. Most wireless APs support multiple SSIDs, which can be tagged to unique VLANs. Segmenting WEP-only phones onto their own VLAN may cause a minor headache when reconfiguring the network, but at least valuable network resources will be protected if the WEP key is cracked by an attacker.

Over the past year, more and more VoIP handset vendors have started supporting improved security methods like Wi-Fi Protected Access (WPA) and WPA2. WPA is a profile of the 802.11i amendment that fixes the security flaws in WEP. Since WPA uses Temporal Key Integrity Protocol (TKIP) encryption exclusively, older devices may be upgraded with a software patch provided by the vendor. WPA2 requires full compliance with 802.11i, which means that AES-based encryption must be supported. The requirement to support AES-based encryption means that old hardware may have to be replaced.

Support for WPA2 is improving, but it still has a way to go before it becomes pervasive. A number of new VoIP handsets available on the market still only support WEP or WPA. The good news is that any newly released device that applies for Wi-Fi certification must support WPA2. The even better news for wireless VoIP performance is that WPA2 devices will support fast, secure roaming (FSR).

Fast, Secure Roaming
Fast, secure roaming is needed on wireless LANs because handsets on networks that require 802.1X/Extensible Authentication Protocol (EAP) authentication may run into additional difficulties when roaming. 802.1X portbased access control requires a supplicant (VoIP handset) to authenticate to a server before it may pass data through an authenticator (AP). The time it takes for the handset to communicate with the server and wait for its authentication to be processed could be enough for call quality to be compromised or even a call to be dropped.

802.1X/EAP authentication may be used on either a WPA or WPA2 wireless LAN, but network administrators that support wireless VoIP should look for WPA2 support. Even though WPA networks support strong TKIP encryption to go with 802.1X/EAP authentication, the process of authenticating to a server every time 802.11 roaming occurs could pose problems for wireless VoIP users. When WPA2 is used, the network may support 802.11i FSR to alleviate some of these problems.

802.11i FSR involves on of two ways of limiting the number of times a handset must authenticate to a server. The most widely available method is called Pairwise Master Key (PMK) caching. When a station completes its initial association to an AP, a PMK is created as part of that process. The PMK will then be used to create encryption keys for either TKIP or Counter Mode CBC-MAC Protocol (CCMP) encryption. By caching the PMK on the station and the AP, both parties can skip 802.1X/EAP authentication (fast) while maintaining a PMK that ensures that the station is allowed on the network (secure).

The problem with PMK caching is that the initial association to each AP on the wireless LAN still requires a full 802.1X/EAP authentication so that a PMK can be created. This means that PMK caching only enables 802.11i FSR after the initial connection to each AP. Some WLAN controller vendors like Symbol, Cisco and Aruba solve this problem by using proactive key caching (PKC). When PKC is used, the first AP that a station authenticates to forwards the PMK back to the controller so that all other APs managed by that controller can use the PMK for 802.11i FSR. Unfortunately, if your wireless LAN features APs that are not managed by WLAN controllers,
PKC is unavailable to you.

Wireless LANs built without WLAN controllers were not completely forsaken by the designers of 802.11i FSR. The second way of limiting server-based authentications is preauthentication. Preauthentication is an optional part of the 802.11i amendment, so there are a number of stations and APs that lack support for it. Those stations and APs that do support preauthentication use it as an addition to PMK caching. If a VoIP handset sees a nearby access point while scanning the wireless channels, it may perform an 802.1X/EAP authentication by sending frames through its current AP and to the new AP through the wired LAN. Once the handset authenticates to the new AP, it then stores the PMK that is created during 802.1X/EAP authentication (PMK caching) and uses it whenever 802.11 roaming needs to occur.

Network administrators should consider the limitations of preauthentication before counting on it as a solution to wireless VoIP problems. A relatively limited number of station and AP vendors support preauthentication, which makes it impractical in most heterogeneous networks. Even if it is supported, preauthentication requires that APs communicate with each other over the wired LAN. That means that all APs must be from the same vendor, and even if that is the case, some vendors do not support wired communication between their APs.

Virtual Cells
Due to the difficulties of 802.11 roaming and the limitations of 802.11i FSR, some network designers would rather eliminate roaming altogether. One company has taken steps in this direction by creating virtual cells.

A virtual cell is a wireless LAN where all APs have the same Basic Service Set Identifier (BSSID). The BSSID is the MAC address of the AP. By having all APs configured with the same BSSID, it appears to VoIP handsets that there is only one AP covering the entire enterprise. Since there appears to be just one AP, the VoIP handset never has to roam.

Virtual cell technology has been criticized as being inappropriate for high-volume data networks because configuring all APs with the same BSSID also requires all APs to be configured with the same channel. A single channel means that all data traffic is crammed into one shared space, with little room to scale out in high-traffic environments.

The argument about whether virtual cell technology is appropriate for your wireless LAN will not be solved in this paper. If you do find that 802.11 roaming is the cause of your users’ problems or that 802.11i FSR isn’t fast enough, the virtual cell technology that is used by Meru may be a good choice.

QoS Mechanisms
Even a well-designed wireless LAN may fall victim to poor voice quality when converged with an active data network. For that reason, the IEEE 802.11 working group created the 802.11e amendment for QoS.

The 802.11e amendment is still being adopted by many VoIP handset vendors, but most wireless LAN infrastructure vendors already support part or all of 802.11e. You can look for products that support 802.11e by seeking the Wi-Fi Multimedia (WMM) family of certifications.

When the IEEE 802.11 working group created the 802.11e amendment for QoS, they included a number of specifications that would enhance the performance of wireless LANs. Prioritized traffic streams, improved battery life, minimized protocol overhead, and direct links between wireless stations were all part of 802.11e.

Prioritized traffic is the most important part of 802.11e when deploying VoIP over a Wi-Fi network. No matter what types of devices are on the wireless network, it is important to give time-sensitive applications like VoIP priority so that the end-user experience is enhanced. WMM-certified, stations and access points break down traffic streams into one of four access categories (voice, video, best effort, or background). Prioritized access to the wireless channel is then given to applications designated as voice or video.

By deploying a wireless LAN that supports WMM, converged voice and data networks are enhanced. Instead of worrying that a spike in Wi-Fi usage will cripple VoIP calls, network administrators can take solace in the fact that voice packets will be put ahead of best-effort data when contending for access to the wireless channel. In addition, many access point vendors now support WMM with Admission Control. WMM with admission control limits the number of stations in the voice access category, which allows the access point to ensure that associated VoIP handset have an adequate channel allocation to make high-quality calls.

It should be noted that deploying a WMM network goes beyond just ensuring that access points and stations are WMM-certified. For a station to enter the voice access category, the application must support WMM. As an example, making a call with a standard SIP-based handset does not ensure WMM support. Even though voice traffic is being sent over the wireless LAN, the SIP-based handset software must have been developed with support for WMM included.

Battery Life
The Wi-Fi Alliance certifies products that include the 802.11e specifications for improved battery life as WMM Power Save. While WMM focuses on improving the network, WMM Power Save was created with the goal of improving the performance of devices that access the network.

All wireless network interfaces require processing resources in order to transmit wirelessly, but smaller devices like VoIP handsets see a disproportionate amount of battery life drained by the Wi-Fi radio. Larger devices, like laptops and tablet PCs, are able to house more powerful batteries. Since the larger displays, peripheral interfaces, and more powerful processors of those devices use far more power than their handset counterparts, the addition of a wireless radio will typically reduce battery life by only a small percentage. A small VoIP handset will have a similar wireless radio, but the radio drains a larger percentage of batter life because the display, peripheral interfaces, and processors require far less power.

The IEEE 802.11 standard does have power management protocols that are designed to enhance battery life, but these protocols have been found to be inadequate. When legacy 802.11 power management is used, the wireless station must send a polling request (called a Power Save Poll) to the access point for every data frame that is buffered while the handset’s wireless radio sleeps. By sending so many Power Save Poll frames, the station loses much of the battery life that was saved by allowing the wireless radio to doze.

WMM Power Save offers enhanced battery life by changing the way stations request data buffered at the access point. Instead of sending a polling request for each individual frame, the station will send a single request in the form of a data frame. This data frame may contain voice traffic or it may be a null data frame.

Even though WMM Power Save has yet to be widely adopted by access point vendors and VoIP handset vendors, many VoIP handsets do enable a power management method that is an improvement to legacy 802.11 power management. This alternative to 802.11 power management is similar to WMM Power Save in that it eschews Power Save Poll frames. A similar amount of battery life is saved because VoIP handsets use a single data frame to request all buffered data from the AP.

Since there is no certification for devices that support the power management method that is similar to WMM Power Save, the only way to verify which type of power management your VoIP handsets use is by performing a packet capture. When the more efficient power management method is used, null data frames with alternating power management subfields will appear in the packet capture. This method of detection does require a relatively deep understanding of 802.11 protocols and wireless protocol analyzers.

Even if you are inexperienced with wireless packet capture applications, you can still identify devices that use this enhanced method of power management. Instead of examining detailed packet decodes to seek out the power management subfield, applications like Wildpackets Omnipeek will display statistics on null data frames. If a large number of null data frames are captured, it can be reasonably assumed that the more efficientpower management method is in effect.

Handset Quality
The use of efficient power management protocols is just one part of VoIP handset quality. Processing quality, call management features, and form factor all contribute to a user’s satisfaction with a wireless VoIP deployment.

Unfortunately, the state of VoIP handset quality could probably best be described as developmental. Today’s VoIP handsets are greatly improved because of 802.11g, adaptive roaming thresholds, improved security, and next-generation power management protocols. Still, when you hold a Wi-Fi handset in one hand and compare it to your cell phone in the other hand, the VoIP device usually comes up short. The fact is that cell phones were designed from the ground up as voice communication devices, while most wireless LAN radios were designed with laptop or handheld computer use in mind. Also, economies of scale have driven cell phone manufacturers to innovate with cell phones at a level where Wi-Fi handsets have been unable to compete.

The good news is that wireless VoIP phones that close the gap in handset quality appear to be on the way. Companies like Research in Motion, Ascom, and Motorola have announced handsets that will have more robust features. Many of these devices will even support cell phone connections as well as wireless VoIP. These dual-mode handsets are seen as an important driver in fixed-mobile convergence (FMC), which is an effort to allow cell phones to leverage existing wired network infrastructures using VoIP.

Conclusion
It should be clear that IT professionals must understand a variety of topics in order to successfully manage wireless VoIP deployments. One must not only recognize the inherent limitations of the wireless network, but also understand developing protocols for security, roaming, QoS, and power management.

Even with all of these obstacles, wireless VoIP is such an attractive technology that IT professionals would serve themselves well to understand it. Employees love the freedom of wireless phones and businesses love the cost savings of VoIP. In all likelihood, VoIP over Wi-Fi networks is here to stay, and we’re going to have to support it.

Learn More
Learn more about how you can improve productivity, enhance efficiency, and sharpen your competitive edge.
Check out the following Global Knowledge courses:

Wireless Networking I: Integration and Troubleshooting
Wireless Networking II: Security and Analysis
Voice Over IP Foundations

About the Author
Benjamin Miller is a wireless networking instructor for Global Knowledge. Mr. Miller is also the Course Director for the Global Knowledge wireless curriculum. He is a Certified Wireless Networking Expert (CWNE) and Certified Wireless Network Trainer (CWNT).

VoIP without hype. What business need to know. Part 2.

2007-03-31T22:35:00.000+04:00

Upgrading for VoIP: Four Approaches
For most SMBs, the migration to VoIP will involve more than choosing a VSP to replace their current LD provider. It will involve upgrading on two fronts: the LAN – that is, the networking equipment on your premises – and the WAN, including your connection to the Internet and the equipment and capabilities of your service provider. There are four distinct approaches you can take to go VoIP, depending on your quality requirements and budget.

Approach #1: Best and Most Expensive
For the best voice quality you should have two entirely separate LANs at your place of business and two entirely separate WANs (connections to the Internet). Dedicate your first WAN+LAN to data and your second WAN+LAN to voice – thus ensuring no physical possibility that your data packets can “stomp” your voice packets. Then, when planning your voice WAN, try to find budget room for a T1 connection instead of a lower quality DSL/cable connection.

Now, when planning for your two LANs, you will need to purchase two separate routers, each with their own physical wiring, which will terminate as two separate Ethernet jacks at each employee location.This means that your employees will plug their computer into one jack and their IP phone into another. Expect to pay $80-$100 per employee location for each extra “drop”, but much less if you have your wiring company run both drops at the same time.
Finally, when you have built your dedicated WAN+LAN combos, you must carefully choose a VSP!

Approach #2: Second Best, Less Expensive
If you don’t want to pay for two separate LANs, you can still get pretty darn good quality if you do two things. First, you must still get two separate broadband providers for your WANs, one for voice and one for data – again, a T1 if you can afford it. Second, you should upgrade to a QoS Ethernet switch on your LAN. A QoS-capable switch (with IEEE 802.1P support, for example) will ensure that voice packets and data packets are prioritized properly on your LAN – thus when large files are moving across your LAN, your switch will make them pause momentarily to let voice traffic go first. Your users won’t even notice the pause, but your voice quality will be significantly improved.

Approach #3: Cheaper but Harder to Pull Off
A more economical approach, but often the hardest to do right, is to add QoS capabilities to both your LAN and WAN, therefore allowing you to get away with only having one of each. To do this properly, you’ll first need to upgrade your LAN with a QoScapable switch/router. You’ll also need to ensure that your broadband provider has QoS capabilities and that your VSP uses the same type of QoS as you use on your LAN. Yes, QoS comes in many flavors. In fact, the easiest way to pull all of this off is to get a T1 that provides you both Internet access and VoIP-based LD on the same circuit, all with QoS, and all from the same broadband provider!

Approach #4: I Want Really Cheap!
OK, OK, we get it. You love your wallet and you didn’t pay for this white paper. So, at the absolute minimum, you must upgrade your broadband connection to ensure sufficient bandwidth for voice traffic from your premises to the Internet. A typical VoIP call will use about 64KB of upstream and downstream, and you should factor in at least 90KB for your first call because of what is called “IP overhead.” So, do your math assuming peak concurrent line usage. If your peak usage will be 5 concurrent calls, then you add 90KB + (4 x 64KB) = 346KB. Be sure to remember that this is 346KB up *and* down. Many broadband providers will give you much more downstream than they will upstream, so be sure to ask! Finally, if you are going to go cheap, and cannot afford a T1, try to at least get “business-grade” DSL (see sidebar, “What It Will Cost”).

In case we haven’t provided you too much information already, there are two final things to note about making your leap to VoIP. First, ensure that your VSP has a local PSTN media gateway in your area; this will shorten the path your VoIP calls have to take over the Internet before they are converted to travel across the PSTN. Second, try to use the same broadband provider at your main office and all remote offices; VoIP performance is usually better when calls travel over a single provider’s backbone vs. having to “hop” across multiple backbones.

Hybrid IP-PBX: The First Step to VoIP
What Will it Cost?
Whatever approach you decide on for migrating to VoIP, a hybrid IP-PBX is an excellent first step. Hybrids operate in three modes – PSTN,VoIP, and what’s called PSTN-fallback – a mode which ensures that you’ll always have phone service, even during Internet outages. With a hybrid IP-PBX, you can also connect and use analog phones (including cordless sets), IP phones, or a combination. So you can convert select employees to IP telephony according to their needs and the capacity of your Internet connections.

A hybrid IP-PBX enables you to start saving money right away, even if you choose to use the PSTN connections to the outside world. With a hybrid IP-PBX at your business, you get free VoIP calls between offices and with all your telecommuters, but you can selectively choose to pay more for calls across the PSTN where the quality matters most. Think of it this way, your employees get free VoIP calling between themselves, but your customers are guaranteed perfect POTS quality when they call you, or you call them.

But hybrid IP-PBX systems aren’t just about VoIP and cheap LD, they also enable SMBs to improve their “communications image” – presenting callers with a professional (and time-saving) auto-attendant, allowing employees to work from home or the road, 4-digit dialing from anywhere in the world, blending Outlook into your phone system and more.

Hybrid IP-PBX capabilities can include:

unlimited extensions and voicemail
multiple auto-attendants (IVR)
unlimited call queues (ACD)
telecommuters – even for call queues
uploading of professional voice prompts
scheduling – time and date settings
music on hold with upsell messages
parking, paging and call forwarding
integration with Outlook and CRM software
customizable caller ID
click-to-call from your website
extensive reporting for productivity analysis

With the advanced functions and features offered by a hybrid IP-PBX system, your SMB can gain the efficiencies of IP telephony and cost savings of VoIP today without having to sacrifice the quality we have all come to expect from the POTS!

About the Author
Chris Lyman is the CEO and founder of Fonality. Chris combines his extensive telephony and VoIP knowledge with his data-centric roots as founder and CEO of Virtualis, a top ten web hosting company, which he sold in 2000. He often presents at industry forums and conferences on topics such as IP telephony, data architecture and open source business models.
About Fonality
Fonality is the leading provider of affordable hybrid IP-PBX systems for SMBs. Fonality’s PBXtra product line comes VoIP-ready but also supports calling via the PSTN and PSTN-fallback. PBXtra Standard and Call Center editions provide enterprise-class features at a fraction of the cost of traditional industry offerings. Deployed to tens of thousands of business users in the U.S. and other countries, Fonality’s products can be purchased direct or through a network of more than 1,300 resellers. For more information, visit www.fonality.com.

VoIP without hype. What business need to know. Part 1.

VoIP without hype. What business need to know. Part 1.

2007-03-31T22:16:00.000+04:00

VoIP: On the Rise Despite Shortcomings
“With voice over IP, you can have long distance for less – even for free!” That is the Siren’s song being played by VoIP, the Internet’s newest promise. Millions of consumers with broadband connections are responding. And there’s no end in sight; it’s expected that 32 million Internet phone lines will be in use by 2009 (Source: Gartner).

Akin to the cellular phenomenon, consumers are rushing to Voice over Internet Protocol (VoIP) despite the fact that audio quality and reliability are not yet up to traditional landline telephony standards (see “I want my V-O-I-P” section for quality analysis). Downtime and quality aside, the value proposition of VoIP has clearly resonated with consumers. During 2005, U.S. subscriptions to VoIP calling plans, which cost as little as $20 to $25 per month for unlimited domestic long distance (LD), more than tripled, from 1.3 million to 4.5 million (Source: TeleGeography).

Seductive as the savings message may be to consumers, the lure of VoIP is even stronger for businesses – whose monthly LD bills are often in the hundreds or even thousands of dollars. Many of these businesses have exacerbated toll charges because they either pay for calls between branch offices or have call centers, which typically incur heavy line usage. For such firms, slashing LD expenses could literally drop thousands of dollars a month straight to the bottom line. By the same token, the more dependent a business is on telephone communications, the less it can afford to compromise on the quality of its phone connections. Poor audio quality can undermine productivity and customer satisfaction; dropped calls cost sales and money; a total telecom service outage might well be disastrous.

I Want My V-O-I-P
It’s important to approach VoIP with your eyes wide open – to understand the technology involved, the investment required, and an affordable migration path to VoIP that will deliver value to your business, both short- and long-term, without
compromising the quality of your communications. But first, let’s talk quality. A study by Internet performance monitor Keynote Systems found that consumer VoIP service reliability improved from 96.9 percent to 99.1 percent between June
2005 and January 2006. 99.1 may sound good, but actually it is significantly lower than the 99.999 percent reliability rating that people have grown used to with the plain old telephone system (POTS), also referred to as the public-switched telephone network (PSTN). To put these figures into perspective, 96.9 percent uptime actually
equates to over 22 hours of downtime per month and 99.1 percent uptime still equates to more than 6.5 hours of downtime a month. Compare those numbers to 99.999 percent uptime, which equates to just 26 seconds of downtime per month, and you can see the impact of a few decimal points!

Packets 101: What Makes VoIP Vulnerable
As most people understand, VoIP is telephone calling over the Internet. So, it may appear that switching from the POTS to VoIP is a fairly straightforward proposition for the average small or medium size business (SMB). In a typical SMB, all employees have PCs, which are connected together via a local area network (LAN). This LAN is then connected to a router/firewall which talks to the Internet via what is called a wide-area network, or WAN. So, if you put a VoIP-capable phone on each desk and plug it into your LAN, the call goes out your WAN and voilà – IP telephony, right? Wrong.

Many people don’t realize that the Internet was *not* originally built for telephone calling. Neither were most LANs – yes, this means you! Even if you have a great LAN, it is unlikely that your WAN, or the WAN of your Internet Service Provider (ISP) is ready to go. In fact, the Internet Protocol itself was not really designed for real-time communication of any sort – especially the streaming nature inherent in audio, video, or online gaming.

Let’s get geeky. IP-based networks divide information – for example, an email message or Web page – into thousands of small chunks of data called “packets”. Each packet has a “header” which contains both the sender’s and the receiver’s Internet address; this enables the packets, which are transmitted separately, to be reassembled at the receiving end. Think of packets like ants, each of which knows how to get back to their anthill, but doesn’t necessarily need to get back at the same time or by the same route. This ability for a packet to “choose its own route and time” is why data transmission over the Internet is so efficient.

VoIP vs. IP Telephony:A moment of distinction
Recently, VoIP has become a catch-all buzzword. Yet, it is important to distinguish between VoIP, which is a digital transport vehicle for phone calls, and IP telephony, which is a digital phone system based on Internet standards. This is important, because businesses stand to benefit from both VoIP and IP telephony – in substantially different ways.

VoIP is a method of digitizing your voice so that it can be transmitted across the Internet to save money on toll charges. Whereas IP telephony is a way of digitizing your phone system so that it can leverage the Internet, your computer, and your other business software applications (CRM, CTI, Outlook) to increase productivity within your business. VoIP is actually a subset of IP telephony – look at it this way: VoIP is an “arrow in the quiver” of IP telephony.

Any IP telephony system will use VoIP as a way of transmitting voice, in some manner or another (SIP, Skinny, MGCP, H.323). But, an IP telephony system, such as an IP-PBX, goes far beyond cheap phone calls; it enhances business productivity by providing additional features that weren’t available or affordable with legacy phone systems.

See, when data packets carrying email or web pages arrive slowly, or out of order, it is usually not a big deal. So what if it takes you an extra second to download your email? Space out for a minute, watch the paperclip, you know the drill. But this method of “out of order” packet delivery spells disaster for real-time protocols that need packets to arrive at the right time, in the right order, all the time. Welcome to the stringent demands of transmitting real-time audio.

OK, let’s adjust our pocket protectors and really talk about this; during a VoIP call, speech is captured as analog information by a phone, then converted into digital information, compressed, and divided into packets for transmission. This whole process is relatively fast, easy and reliable. The potential problems happen at the receiving end, when the packets must be reassembled in the correct order, absolutely error-free, and reconverted from their digital form into a seamless audio stream – all in real time.

If there are any substantial glitches in transmission, a VoIP call “breaks up” like the reception from a distant radio station. As engineers say, the audio stream stutters. As the people on the call will tell you, it sounds like gibberish. Houston...we.. av...a..ro..bl...em. In the worst case, which is not all that rare, the entire call is dropped – that is, cut off – because the transmission becomes so overwhelmed by problems that the connection simply fails. Dropped calls can mean lost revenue and/or dissatisfied customers!

Is Your Network Ready for VoIP? Probably Not
Here’s a fact: the LANs and Internet connections (WANs) used by most SMBs are simply not ready to handle VoIP. The basic firewalls commonly used for security and virus protection often cause VoIP calls to break up. The low cost routers from the local computer store often don’t have the horsepower to drive quality VoIP calls. LANs can also become congested, especially when users are transferring large files across the internal network, such as when sending or receiving emails with large files attached, downloading documents, doing file backups or copying media files.

Of course, your VoIP network includes not only your LAN but also your WAN. Your WAN begins with your broadband modem and ends with your broadband Internet provider. Most people don’t understand that just having a broadband connection is not enough. You actually need a *high quality* connection to deliver the call quality you need to run a real business. Many SMBs connect to the Internet via DSL or cable, most often with inexpensive modems. While such connections work fine for web browsing and email, they are not designed to handle VoIP transmissions, much less the combination of voice and data.

Into the Weeds: VoIP explained
Let’s talk protocol! This will help us understand why the real-time demands of VoIP put such a strain on the Internet and your LAN. As mentioned previously, the Internet was originally designed so that packets could arrive “out of order” and be reassembled by the client. The protocol most commonly used to transmit these packets is called TCP/IP.

Yet, TCP/IP is rarely used for VoIP packets because this protocol was not designed for real-time communications. Instead, VoIP often uses the UDP protocol. UDP tends to carry low overhead, making it a good choice for voice calls. But, the low overhead of UDP also makes it sensitive to network conditions. Therefore,VoIP-over-UDP can sound poor when it encounters any of following conditions:
    Latency is the time it takes for a data packet to make a round trip between the sending and receiving location. When the average latency of a VoIP connection is above 200ms, call quality suffers. The best VoIP connections have latency under 80ms. Email and web access can gracefully handle latency of 400ms.
   Jitter is the result of variance in latency between subsequent packets. For example, if you ping a network and get results such as 90ms, 92ms, 89ms, the network is jitter-free or, to be exact, it has jitter of 3ms (variance between 92ms and 89ms). But if your pings looked like: 50ms, 70ms, 190ms, then your network has jitter of 120ms. Jitter that exceeds 100ms degrades the quality of a VoIP call. Jitter under 50ms is gracefully handled by most IP phone systems.
   Packet loss occurs most commonly when an Internet network is congested. Under such conditions, packets are often simply discarded. TCP/IP automatically retransmits these lost packets, but VoIP-over-UDP will not. Packet loss will create “stuttering”, or in extreme cases, “silent gaps” in your phone call.

Similarly, many of the WANs now in use by ISPs were designed and built before the advent of IP telephony. They weren’t originally designed to meet the demanding requirements of error-free, reliable VoIP transmission. In fact, most of them actually run on a business model designed for oversubscription, which results in frequent latency and jitter (see sidebar: “Into the Weeds: VoIP explained”). Even the ISPs that aren’t oversubscribed have not yet deployed true quality-of-service (QoS) technology to ensure that voice packets get priority over data packets across their networks.

As if ensuring you have a great LAN and WAN was not enough, you also have to choose a high quality VoIP provider (VSP or ITSP). [Is your acronym meter at full capacity yet?] Much like LD providers, which deliver long distance services over the PSTN, VSPs provide you with VoIP calling over the Internet. Like ISPs, not all VSPs are created equal in terms of network strength, proximity to the PSTN backbone and, of course, good old fashion customer service. The VSP industry is a new one, so be sure to choose carefully. Remember a great LAN and WAN don’t mean anything if you have a weak VSP!

VoIP without hype. What business need to know. Part 2.

Guide To Free Voip Calls

2007-03-30T12:40:00.000+04:00

I am sure you have heard the adage, “There’s no such thing as a free lunch”. Well, obviously the person famous for that saying hadn’t yet heard of VoIP. VoIP is Voice over Internet Protocol- or in simple terms, the ability to transmit voice over the Internet. VoIP has been gaining in strength and popularity as a telecommunications of choice. Many small businesses are opting to go VoIP. It is a viable source for businesses to make long distance calls virtually free, or at a significantly lower cost when compared to other telephone service providers.

With a High Speed Internet connection, you too can experience the power of VoIP. In fact, you may have already tried this technology without even realizing it. There are many services and programs that have been available for free, on the Internet, for the past ten years that have embraced VoIP.

Dialpad was known for allowing people to connect a microphone and speakers to their computers and then make long distance calls virtually world wide for free. Well, Dialpad isn’t free any more, but if you do purchase their monthly service, they do provide an option for unlimited long distance calling. This service is still comparable to other telephone providers.

Currently most of the free VoIP have begun to charge a monthly fee for their services. There is still one program left that is absolutely free. That is Ad Calls. This program displays various advertisements on their display while you can use the service to make free calls. There are limitations to this program however. You can only use the service for ten minutes at a time, and you are unable to call toll free numbers.

The major VoIP programs that embrace free calling left on the Internet are bundled with your Instant Messaging programs. Programs such as Yahoo Messenger have recently been redesigned with the ability to make free voice phone calls using VoIP. You won’t be able to call a landline telephone however. You will need to use the program to call another member who is using the same program. For example, if you have the newly VoIP enabled Yahoo Messenger downloaded to your computer, yet your buddy (who lives across the world) is using the old Yahoo Messenger, you can send them a link to download and upgrade their current version. Once they have installed the new Yahoo Messenger, you will see a “call” button with a phone icon at the top of your Yahoo Messenger chat box. By clicking call, you will be using VoIP technology to send a phone call to your buddy’s computer. The receiving computer will actually ring, like a standard telephone, and your buddy will have the option of accepting or declining the phone call. All that is needed is a high speed Internet connection, this provides the best audio quality and speed), a microphone, and speakers. You and your buddy are free to talk as long as you like. The other advantage to Instant Messengers that allow free VoIP calls is the ability to call more than one person. You can call people on your buddy list, and invite them for a conference call. This has unlimited potential and can be used by small businesses for free training calls, and by people everywhere who just want to stay in touch with friends and family.

Other Internet Messaging services that have embraced VoIP calling include MSN messenger, AOL Instant Messenger, and Google has begun their new service, Google Talk.

VoIP is a preferred method of telecommunications due to the fact that you can integrate audio, video, and web based applications with one program. The Instant Messaging programs are perfect examples of how audio and web based communications are working together to increase communication. Also included with many Instant Messaging programs is the ability to hook up a digital camera or web cam and incorporate video with your audio and web based application.

VoIP Telephones are also beginning to incorporate video with their VoIP services. This will be a very popular trend in the future. As there are over 900 million Instant Messaging subscribers, people are already experiencing the power and freedom of converged communications.

Cellular phones are realizing that people want to combine video with their audio, and many videophones are being released. Some phones have gone the next level and have combined audio, video, and graphics by allowing you to store pictures in your telephone, and use it as a camera.

As the technology of VoIP continues to advance, we can rest assured that more “Beta” programs will be released and as that occurs, keep your eye out for more free VoIP programs.

5 Steps To Voip

2007-03-30T12:35:00.000+04:00

So you’re ready to enjoy unlimited long distance calling and lower monthly phone bills. Well, before you make the switch, here are five steps that will help you get the VoIP plan that is best for you.

1. Make the Decision

Yes, it is new technology, and sometimes we are scared of the unknown. VoIP technology is on the rise and many high profile computer companies are already switching to VoIP and in return are becoming service providers. AOL, Sprint, and Verizon are offering VoIP as well as Internet Messaging services such as Yahoo, MSN, and AOL IM. First you must come to the decision that VoIP is best for you, your family, or your business needs. Research VoIP on the web and its providers. Talk to people that you know have already made the switch. Once you have made your decision to switch to VoIP, you are ready for the next step.

2. Choose the Plan that is Right for You

A good idea is to make a written list of your needs, wants, and features in a phone. This will help you compare different plans and choose the service provider that will best fit your needs. If you are looking for residential or business service, make sure that you note that in your list. Residential plans usually come in two forms- basic (which is a certain amount of minutes) and unlimited (which will give you unlimited local and long distance talk time). Business plans can include amenities such as multiple phones, greater minute talk time, greater unlimited calling plans, business features bundled with your plan, faxing and voicemail to your computer capabilities, and extras such as yellow page listings. Check with the company to find the plan that meets your requirements.

3. Setting up the System

Some service providers will send you the equipment that you need to begin using VoIP. You will first need to make sure that your computer system is compatible with the VoIP service. You will also need to have high speed Internet already installed with your computer. If you are using VoIP for a business or for a call center, it is best to leave the installation to a professional. Installing VoIP for a residence or home office is typically simple enough for most residents to do themselves.

4. Choosing your Phone Number

One of the best features that VoIP has to offer is the freedom to select your own telephone number. If you just want to transfer your old number to your new VoIP system, check with your service provider before making your final decision. Many VoIP providers will allow you to transfer your phone number.

If you want to select a different area code, this option has many advantages. First, for friends and family members who don’t have unlimited long distance calling, selecting the same area code as them can save them money. They will appreciate being able to call you as if it were a local call.

This strategy is also very beneficial for those whose businesses are targeting specific areas. Say there is a California company, yet they want to increase business in Tampa, Florida. By selecting the Tampa, Florida area code, they can increase sales and productivity by having a local number for customers to call. This also reduces the need for a toll free number.

5. Install your Features and Accessories

VoIP comes with many features included in your service plan. For residential customers these include Voicemail, Caller ID, Call Forwarding, Call Block, and Do Not Disturb. Always check with your service provider for a detailed listing of the features included with your plan. For Small businesses, amenities such as multi phone users, free fax lines, and integrating your VoIP and web features are all available. VoIP also contains many of the same features bundled with basic plans as well. When you are choosing a provider, compare their business plans and select the one that best fits your business needs.

As VoIP technology continues to advance, a new array of accessories appears on the market. Today you can choose from VoIP phones, headsets, and even VoIP videophones that work as a web camera incorporating video with your audio.

By: Michael Plante

10 reasons to switch to an IP PBX

2007-03-28T10:47:00.000+04:00

The benefits of replacing your old PBX with an IP PBX

What is an IP PBX?

An IP PBX is a complete telephony system that provides telephone calls over IP data networks. All conversations are sent as data packets over the network.

The technology includes advanced communication features but also provides a significant dose of worry-free scalability and robustness that all enterprises seek. The IP PBX is also able to connect to traditional PSTN lines via an optional gateway - so upgrading day-to-day business communication to this most advanced voice and data network is a breeze!

Enterprises don’t need to disrupt their current external communication infrastructure and operations. With IP PBX deployed, an enterprise can even keep its regular telephone numbers. This way, the IP PBX switches local calls over the data network inside the enterprise and allows all users to share the same external phone lines.

How it works

An IP PBX system consists of one or more SIP phones, an IP PBX server and optionally a VOIP Gateway to connect to existing PSTN lines. The IP PBX server functions in a similar manner to a proxy server: SIP clients, being either soft phones or hardware-based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VOIP gateway or a VOIP service provider.

The top 10 Benefits

Benefit #1: Much easier to install & configure than a proprietary phone system:
An IP PBX runs as software on a computer and can leverage the advanced processing power of the computer and user interface as well as Windows’ features. Anyone proficient in networking and computers can install and maintain an IP PBX. By contrast a proprietary phone system often requires an installer trained on that particular proprietary system!

Benefit #2: Easier to manage because of web/GUI based configuration interface:
An IP PBX can be managed via a web-based configuration interface or a GUI, allowing you to easily maintain and fine tune your phone system. Proprietary phone systems have difficult-to-use interfaces which are often designed to be used only by the phone technicians.

Benefit #3: Significant cost savings using VOIP providers.
With an IP PBX you can easily use a VOIP service provider for long distance and international calls. The monthly savings are significant. If you have branch offices, you can easily connect phone systems between branches and make free phone calls.

Benefit #4: Eliminate phone wiring!
An IP PBX allows you to connect hardware phones directly to a standard computer network port (which it can share with the adjacent computer). Software phones can be installed directly onto the PC. You can now eliminate the phone wiring and make adding or moving of extensions much easier. In new offices you can completely eliminate the extra ports to be used by the phone!

Benefit #5: Eliminate vendor lock in!
IP PBXs are based on the open SIP standard. You can now mix and match any SIP hardware or software phone with any SIP-based IP PBX, PSTN Gateway or VOIP provider. In contrast, a proprietary phone system often requires proprietary phones to use advanced features, and proprietary extension modules to add features.

Benefit #6: Scalable
Proprietary systems are easy to outgrow: Adding more phone lines or extensions often requires expensive hardware modules. In some cases you need an entirely new phone system. Not so with an IP PBX: a standard computer can easily handle a large number of phone lines and extensions – just add more phones to your network to expand!

Benefit #7: Better customer service & productivity:
With an IP PBX you can deliver better customer service and better productivity: Since the telephone system is now computer-based you can integrate phone functions with business applications. For example: Bring up the customer record of the caller automatically when you receive his/her call, dramatically improving customer service and cutting cost by reducing time spent on each caller. Outbound calls can be placed directly from Outlook, removing the need for the user to type in the phone number.

Benefit #8: Twice the phone system features for half the price!
Since an IP PBX is software-based, itis easier for developers to add and improve feature sets. Most VOIP phone systems come with a rich feature set, including auto attendant, voice mail, ring groups, advanced reporting and more. These options are often very expensive in proprietary systems.

Benefit #9 Allow hot desking & roaming
Hot desking – the process of being able to easily move offices/desks based on the task at hand, has become very popular. Unfortunately traditional PBXs require extensions to be re-patched to the new location. With an IP PBX the user simply takes his phone to his new desk – No patching required!

Users can roam too – if an employee has to work from home, he/she can simply fire up their SIP software phone and are able to answer calls to their extension, just as they would in the office. Calls can be diverted anywhere in the world because of the SIP protocol characteristics!

Benefit #10 Better phone usability: SIP phones are easier to use
Employees often struggle using advanced phone features: Setting up a conference, transferring a call – On an old PBX it all requires instruction.

Not so with an IP PBX – all features are easily performed from a user friendly Windows GUI. In addition, users get a better overview of the status of other extensions and of inbound lines and call queues via the IP PBX Windows client. Proprietary systems often require expensive ‘system’ phones to get an idea what is going on on your phone system. Even then, status information is cryptic at best.

Conclusion

Investing in a software-based IP PBX makes a lot of sense, not only for new companies buying a phone system, but also for companies who already have a PBX. An IP PBX delivers such significant savings in management, maintenance, and ongoing call costs, that upgrading to an IP PBX, should be the obvious choice for any company.

By Nick Galea