Thursday, July 27, 2006

Packet8 Business Calling Plans....VoIP At It's Best

Packet8 offers three Internet phone service plans for small and medium sized businesses:

* Virtual Office - our most comprehensive hosted PBX platform, requires a minimum of three unlimited extensions

* Virtual Attendant- the a la carte solution for business, can be implemented with as little as one virtual extension

* Business 2000- a single-line VoIP business phone service

All Packet8 plans help companies dramatically lower their telecommunications costs and are well suited to today's dynamic business environments. Simple set-up, ease of use and a low subscription price make Packet8 business Internet phone services an outstanding value.

Virtual Office

Virtual Office is a hosted Internet PBX solution comprised of powerful business class features including a customizable auto-attendant, 3 digit extension to extension dialing worldwide, ring groups, and conference bridge. Each extension includes its own non blocking line service, 3-way conferencing, multiple call handling, business-class voicemail, direct inward dialing (DID), call forwarding, transfers, conferencing, caller ID, music on hold, and more. Virtual Office requires a minimum of three unlimited extensions which include unlimited local and long distance calling to the United States and Canada for $39.99/month per extension.

This service also includes a conference bridge with up to 20 participants. Add a receptionist console, toll free numbers or virtual numbers at a small additional cost.

Virtual Attendant

Virtual Attendant, our powerful, automated inbound call routing solution, is an a la carte service enabling small businesses to connect wired, mobile, home phones and other Virtual Office extensions under a single primary phone number. Virtual Attendant can be used as either an independent service or in conjunction with an accompanying Virtual Office Service Plan for subsequent unique auto attendants. Callers are greeted promptly by an automated attendant that forwards the incoming caller to the designated extension numbers. Virtual Attendant calling plans start at $14.99/month plus associated extension charges.

Business 2000

The Business 2000 calling plan is a VoIP telephone service that includes a single direct-inbound-dial line with a basic business feature set that includes voicemail, caller ID, call waiting, three-way calling, find me/follow-me, simultaneous ring, call forwarding, 7 digit local dialing and more. Business 2000 calling plans include 2000 outbound/inbound PSTN minutes and are priced at $34.99 per month.

You can get more information and order your internet phone service plan at:

Packet8 Business Calling Plans

Tuesday, July 25, 2006

Insights On Bandwidth Requirements For A Business VoIP System

How Much Bandwidth Is Required To Deploy VoIP For Business Applications?

This is the most commonly asked question on a VoIP network. That's because voice is much more sensitive to traffic congestion, on the network, than data. When implementing VoIP over a LAN, you obviously have much more bandwidth at your disposal. Therefore, you can configure VoIP devices to use a more bandwidth-intensive codec, such as the G.711, which consumes up to 87.2 kbps of NEB (Nominal Ethernet Bandwidth) in one direction. This will ensure better voice quality. In addition to this, you would also need to analyze the existing traffic patterns on your network. If it's already congested with a lot of broadcast traffic or other bandwidth-intensive applications, then the response time would be higher. This in turn will affect the voice quality of VoIP calls. You might face breaks or choppy voice due to this.

One common misconception about VOIP is that it is a bandwidth hog, when, in fact, voice is a very efficient type of traffic. Voice compression standards like G.729 (8:1) and G.723 (10:1) are used to minimize the bandwidth required for voice. G.723, for instance, is the maximum compression rate and requires only 5.3K bps (plus an added 7-8K bps for IP overhead). Even at maximum compression, your VOIP solution will still provide near toll-quality voice.

As a rule of thumb, 14K bps of bandwidth per call is ideal. This includes the compressed voice packet and the IP overhead. To determine total VOIP bandwidth needed per location, take the number of VOIP channels being used and multiply by 14K bps. Then double this number to accommodate for both voice and data traffic.

It should also be noted that bandwidth is used only when someone is speaking. A silence suppression/Voice Activation Detection (VAD) feature is an option that frees unused call bandwidth for data traffic. This is significant, since callers are usually silent for 60 percent of the call.

An easy tool you can use for both planning and system checkups....for this "Lines To IP Bandwidth Calculator" from WestBay: IP Bandwidth Calculator

Another good bandwidth calculator is found at

There are also several good software tools that can help you. Check out, which has some specific tools to help you with VoIP deployment. Plus, they even have other tools to help you calculate the response time and throughput on your network.

While deploying VoIP on LANs, experts recommend that you create a separate VLAN (Virtual LANs) on your network for IP telephony. This will keep the voice and data networks separate, and anything happening on one will not affect the other. For this, you'll also have to ensure that you're using manageable switches on your network, which support VLANs. Having a separate VLAN will also ensure that your VoIP network remains unaffected from security threats that might occur on the data network. A DoS attack for instance, would affect a VoIP network much more adversely than a data network.

The key issue in deploying VoIP over WAN links is that you have limited bandwidth. So you have to start by determining how many simultaneous voice calls you'd want to hold over your WAN links. Then determine how much bandwidth would be consumed by each voice call. For this, you have to take into account the compression technique to use, the payload size of voice packets, and the type of link used for VoIP communication. There are a lot of different codecs that can be used for VoIP communication, supporting bit rates that range from 5.3 Kbps to 64 Kbps. Using the codec data along with the payload size and type of link, it's possible to calculate the amount of bandwidth required per call. You'll also find a lot of bandwidth calculators available on the Net to help you with this job (such as the examples above).

A Critical Factor Is Ensuring Quality of Service

Once you have calculated the bandwidth required for VoIP, you have to ensure you're your voice calls get that much bandwidth. This is when you have to enforce the QoS policies, else, you will get poor quality reception, delays, jitters, missed speech or even dropped calls. Typically toll quality calls require at least 16-20 kbps of bandwidth. QoS is normally controlled at the router level. Therefore, you'll need a router that let's you configure QoS for VoIP packets on your WAN links.
You can also deploy switches that support QoS for VoIP on your LAN. There are also some bandwidth-management solutions available that support QoS policies.

Now For The Details Behind The Scenes

Remember.....the techie details ARE important, but NOT overwhelming.

VoIP creates two types of network traffic – the call control messages used to setup and manage connections between users, and the digitally encoded voice conversations. The call setup and management protocols involve simple messaging between IP phones and an IP PBX. These protocols use very little bandwidth and they do not have stringent latency requirements. A delay of a few seconds in setting up a call is usually acceptable. The real challenge is to satisfy the bandwidth demands of the digitized voice streams between users. Each call consumes a nearly constant amount of bandwidth for the duration of the call. How much bandwidth is needed for each call? That depends primarily on the voice encoding technique used as well as a couple of other variables.

Two voice encoding standards are widely supported by VoIP products. The first is the G.711 standard that uses the same PCM encoding used on the PSTN at a bit rate of 64 kbps. In contrast to the PSTN approach of sending 8-bit PCM voice samples at 125 microseconds Page 2 intervals, G.711 packs multiple samples into each IP packet sent. Packing multiple PCM voice samples into a single IP packet reduces packet header overhead. Each VoIP packet is made up of IP/UDP/RTP headers in addition to the voice sample payload. Because these headers total 40 bytes per packet it is important to minimize the total number of packets sent.

The maximum payload sizes are limited by the encoding latency as payload size is increased. G.711 payloads are usually limited to 160 bytes (20 ms. of voice) or 240 bytes (30 ms. of voice) because larger payloads would increase the encoding latency beyond acceptable limits and cause perceptible delays in conversations. G.729 is another widely supported voice encoding standard. G.729 encodes voice at a bit rate of 8 kbps by compressing as well as digitizing the voice signals. This compression is lossy and can degrade voice quality compared to G.711 encoding. The payloads of G.729 packets are typically 20 or 40 bytes. Although G.711 and G.720 encode voice at bit rates of 64 kbps and 8 kbps respectively, the actual link bandwidth consumed is greater because of the IP/UDP/RTP packet header overhead.

The actual link bandwidth requirements for G.711 and G729 are:

* G.711 with 160 byte payloads 83 kbps
* G.711 with 240 byte payloads 76 kbps
* G.729 with 20 byte payloads 26.4 kbps
* G.729 with 40 byte payloads 17.2 kbps

Link bandwidth requirements can be reduced for all encoding schemes by using a technique called RTP Header Compression (cRTP). cRTP operates hop-by-hop and compresses the 40 byte IP/UDP/RTP headers to 2 or 4 bytes.

Link bandwidth requirements when using cRTP are:

* G.711 with 160 byte payloads 68 kbps
* G.711 with 240 byte payloads 66 kbps
* G.729 with 20 byte payloads 11.2 kbps
* G.729 with 40 byte payloads 9.6 kbps

Another technique, called Voice Activity Detection (VAD) can further reduce link bandwidth requirements by detecting periods of silence in conversations and preventing packets of silence from being sent. VAD works with all encoding standards and can typically reduce the per call traffic volume by about one third, but its statistical nature means that actual link bandwidth requirements are reduced only in situations where a large number of VoIP calls share a link.

VoIP has three specific performance requirements that have to be met in order to provide toll quality voice conversations.

The first is end-to-end latency. Anyone who has ever tried to carry on a conversation over a satellite link knows how excessive latency impacts quality. Long delays make it difficult for callers to determine when the person at the other end has finished talking. This results in very unnatural speech patterns. How much latency is too much? A rule of thumb is that one-way latency should not exceed 150 milliseconds. 150 millisecond delays are noticeable, but when latency exceeds 250 milliseconds it becomes difficult to carry on a conversation. Latency is a non-issue on the PSTN, but delays on IP networks can easily cause latency to exceed 150 milliseconds.

End-to-end latency is the sum of encoding/decoding latency and transmission latency. The level of compression provided by the codec is proportional to the encoding/decoding latency it introduces. For example, G.711 performs no compression and adds negligible latency while G.729 codecs compress voice to 8 kbps but add a one-way delay of about 25 ms. More significant delays can occur when voice packets are transmitted across a network, particularly when low speed WAN links are involved. On T1/E1 and faster links this latency is only a small fraction of the total one-way latency budget of 150 ms., but on low-speed links the situation is very different. A single 1,500 byte packet on a 64 kbps link will push the latency beyond the 150 ms mark and even on a 128 kbps link, nearly two thirds of the total delay budget is consumed by just the transmission delay. This problem is compounded by the fact that compressed voice formats such as G.729 are more likely to be used over low-speed WAN links, and these algorithms contribute their own latency to the total end-to-end delay.

Even when voice packets are not blocked by data packets they are subject to their own serialization delay – the amount of time that is takes to clock the bits onto a serial link. Again, this delay is determined by packet size and link speed. Reductions in packet size result in less serialization delay and therefore, lower end-to-end latency.

Another key performance metric is jitter. Jitter is the amount of variation in latency that is experienced over time. IP phones have some ability to buffer incoming audio streams to compensate for jitter, but excessive jitter can disrupt conversations. Again, the PSTN has virtually no latency and therefore no jitter, but enterprise IP networks are subject to jitter caused by congestion on LANs and WANs and by packet buffering in routers and other network devices.

The third important performance metric is packet loss. Since VoIP is a real-time audio service that uses UDP transport protocols, there is no way to recover lost packets. Packet loss can result in a metallic sound or dropouts in conversations that can be very frustrating to users. The PSTN experiences virtually no loss of digitized voice, but IP networks routinely experience packet loss due primarily to congestion.

The key to meeting all of the VoIP performance requirements is adequate bandwidth, and the simplest solution is to throw bandwidth at the problem. This approach is being used successfully on enterprise LANs that have been upgraded to switched 100 Mbps and gigabit Ethernet. The real challenge is the wide area network. Private WAN facilities such as frame relay and private lines are very expensive and as a result most enterprises still have very limited bandwidth between their headquarters and their remote offices. Half of all WAN links between corporate headquarters and remote offices are 56kbps/64kbps or lower. Most other remote offices operate at speeds of 128kbps to 512kbps and fewer than 10% are T1/E1 or greater.

So What's The Solution?

Now, all of this may seem overwhelming and utterly impossible to implement for your organization. Not so. You and your IT staff need not experience high blood presure and assorted other health ailments while planning, designing, and implementing a VoIP system. Simply take advantage of the free consultation available through Business VoIP Solution and you can rest easy knowing everything will be taken care of.

Friday, July 21, 2006

Challenges And Solutions For Medical Imaging Bandwidth Requirements

A Picture Archiving and Communications System (PACS) is integral to the smooth, timely, and quality delivery of health care in every medical setting today. Not only are they integral but they are crucial to the clinical and business aspects of radiology practice as we know it. However, PACS have long faced challenges in delivering this digital imaging support to such diagnostic modalities as X-ray, ultrasound, computed tomography (CT), magnetic resonance imaging [MRI], Positron Emission Tomography (PET), and Teleradiology.

The main issue has always been the availability of sufficient bandwidth (load and speed) a reasonable support the growing demand for quick easy web-based access by medical providers. As Medical Imagery becomes more and more digitalized....with bandwidth improvements, communication will be faster and easier, and it will be possible to transmit heavier studies in less time and with high quality.

An internal (facility owned) PACS leverages a common infrastructure for all the digital imaging modalities and provides image storage and archiving....with recall as needed....for an entire medical facility or campus. By instituting a web enabled distribution system a facility PACS is able to provide ready image access to the immediate radiology department as well as the full range of clinicians and specialists, especially surgeons and referring physicians. To ensure functionality at the high level required means facing the heavy bandwidth appetite of the modalities supported.

Even an Application Service Provider (ASP) company that hosts applications, manages them and rents access to images from a centrally managed facility is not immune to the bandwidth concern. ASP providers allow an institution to outsource information technology applications infrastructure, management, support and maintenance. As defined by the ASP Industry Consortium, ASP service is designed to “deliver and manage applications and computer services from remote data centres to multiple users via the Internet or a private network.” therein lies they're challenge....a high bandwidth requirement delivered over often a subtantial difference on an on-demand basis.

PACS manufacturers have developed numerous solutions to get around the bandwidth problem. They've compressed images, supported standard network interfaces and protocols such as Ethernet and TCP/IP, and deployed local area networks (LANs) with high bandwidths to link hospitals or referring physicians in a contained environment. But how do they handle bandwidth when institutions are separated by tens or hundreds of miles, especially since images have become larger and more complicated?

Some PACS vendors rely on the communications infrastructure in an area, which varies with the bandwidth that is available from the local telephone company and the price a hospital is willing to pay, said Frederick Wagner, manager of PACS for Toshiba. Other PACS providers offer streaming technology that transports high-quality images in real time over any bandwidth, including telephone lines and enterprise-wide LANs.

Another contributing solution is a technology called Pixels-on-Demand by Real Time Media. This technology speeds processing by capturing images from archives or PACS storage without waiting for preprocessing, immediately streams data from selected regions of interest, and delivers the most visually important features of an image to the viewer first.

The underlying solution to the bandwidth issue goes beyond even system technologies, network interfaces, image compression, and infrastructure protocols. It lies with the provision of the appropriate bandwidth capacity (circuits) a reasonable cost....via leveraging the fiber-optic infrastructure available throughout the United States. Enabling direct fiber-optic connectivity internally, or between hospitals and distant data centers, is the most cost-effective application of bandwidth. Use of Optical Carrier (Sonet Ring) bandwidth (likely OC3 or OC48) or Gigabit Ethernet allows a medical facility to optimize it's Local Area Network (LAN). While ASP organizations can scale their application service provider (ASP) service to small imaging centers as well as large, far-flung health systems.

To find a fiber optic infrastructure provider which can deliver the bandwidth solution for your medical imaging application.....I strongly recommend that you take advantage of the free consultation providing by Medical Imaging Bandwidth Solution.

Wednesday, July 19, 2006

Business Prospects Of Wimax -- An ISP point of View

Unlike most people's expectation of rural deployments for WiMax, you might consider targeting SME's in urban areas. There are several reasons for this:

• There is a growing demand in business for bandwidth capable of carrying symmetrical traffic, for voice, applications and uploading of larger files.

• There is a small but growing need for separated last mile services. Currently, however many wired service providers you have, they all use the incumbents' last mile infrastructure based on its nearest telephone exchange location unless you have paid for an expensive dig from the next nearest exchange. This leads to single points of failure and the potential for business communications to be down for days, as can happen say with a cable duct fire somewhere in the spoke.

Your worst case environment would be a very high-density urban area with lots of interfering buildings, has multiple fibre networks, ADSL and SDSL in every exchange, hundreds of competing suppliers, a restrictive property planning regime with many 'listed' buildings, and no spare spectrum for FWA except the public 5.8GHz band.

To do this, because of the scale of competition from other service providers, your model needs to be disruptive. It has to offer things that businesses need (like QoS, toll-quality VoIP, high-quality video, symmetric bandwidth, higher capacities and network separation etc) at a lower cost.

This means stripping all unnecessary cost out of the model. You'll benefit from a quality RF planning tool that gives you a major advantage over other operators - mapping exactly where you can provide service, how to set up the customer antenna, what bandwidth can be achieved etc, based on your base-stations. You need to know exactly how to tune base-stations to avoid blackspots - without needing an RF team.

Although Wi-Fi and WiMAX often get confused, they are very different from an operators perspective. Wi-Fi is plug and play with no control over the wireless interface. WiMAX is not, it behaves more like a carrier ATM network. Wi-Fi is built into laptops and handsets, whereas FWA WiMAX requires larger standalone receivers (yours should mount on customer rooftops for optimum utilty).

The benefit is that WiMAX is very spectrally efficient, at least 50% more so than 3G networks, so it has much higher data-carrying capabilities in limited spectrum. All Wi-Fi shares the same public spectrum - WiMAX can work across a wide range. Wi-Fi provides service over a range of 100m, your WiMAX needs to provide 10Mbps over a range of 1.3km from a base-station non-line-of sight.

WiMAX can create carrier-class networks, Wi-Fi cannot – not even with mesh networks. However, Wi-Fi with WiMAX backhaul gets some of the benefits of WiMAX as the backhaul such as VPN’s. A lot of WiMAX customer equipment will come with Wi-Fi built in.

Don’t wait for mobile (802.16e) WiMAX – your experience with vendors may be that they're around fourteen months to two years behind on their promised delivery dates, and further delays could occur to key requirements. Don’t expect good enough 802.16e equipment to build a network with until late 2007 at the earliest, and no usable CPE until 2008 – as it’s mobile battery life is crucial and that will take time to get right.

There are big enough markets for FWA now. The most important thing is to grab the scarce resources first – spectrum etc – and make them yours. Except in those undeveloped countries without a mobile operator, mobile WiMAX will be very difficult to establish against incumbent operators with large installed bases because the areas covered are important to customers – which is not a consideration for FWA.

Monday, July 17, 2006

Telarus Launches VARSearch(tm) to Aide Users Looking for Network Equipment and Telephone System Installers

DRAPER, UTAH - June 22, 2006 - Telarus, Inc., parent company of - provider of real-time T1 quotes, today announced the release of version 1.0 of VARSearch(tm), a real-time search for network equipment and telephone system installers and dealers in every local market across the United States. The VARSearch(tm) technology is available for public use at

"Value Added Resellers (VARs) and network integrators are in high demand by our T1 and DS3 clients" commented Adam Edwards, CEO of Telarus, Inc. "Creating a way for our customers to connect with these VARs was the next logical step in our progression as a company. We strive to build networks of experts who all mutually benefit from their relationship with Telarus."

As of today's release, VARSearch(tm) boasts 863 registered VARs, with 275 who are actively working with new telecom and network equipment clients. That number is expect to increase dramatically as the program gains traction and more VARs find out about the free lead referral service.

According to Patrick Oborn, Chief Innovations Officer of Telarus, will become the next 'killer app' in the world of online communications shopping.

" has been in the works ever since one of our dedicated business customers asked us who we recommend to install a new T1 card in their telecom closet. The fact was, we didn't know. In tomorrow's telecommunications market, you'll not only have to know who the best high-speed internet and voice providers are, but also who is best suited to assist with the installation, program the new PBX, set up the clients' VoIP handsets, etc. With VARSearth(tm), Telarus/ now have all of these bases covered."

Oborn continued, "The next version of our VARSearch(tm) software will include customer reviews of VARs, better selection and matching algorithms, and a host of other suggestions we've received from our users." is a free advertising resource for VARs who want to get their name in front of potential buyers and ShopforT1 agents alike. The VARs only pay an advertising fee when they meet with success in the program, and find a new customer.

"By taking the risk away from the VARs, who are admittedly not marketing specialists, we help them keep their fixed costs to a minimum while still delivering potential new customers on which they rely for growth. Other programs charge up-front fees to join, in addition to other costs. We will undoubtedly be the low cost leader in this space" continued Edwards, when asked how is different from other VAR referral services. "We look forward to delivering a great deal of free leads to our VAR partners" concluded Edwards.

About is a provider of real-time T1 price quotes using the patent-pending GeoQuote, delivering a unique opportunity for business shoppers to view pricing and availability in real-time for over 12 high-speed data and dedicated voice providers. In addition, offers complimentary consulting service from trained and licensed telecom experts. These independent agents assist potential T1 customers with the pricing of dedicated services and help them find the best service to meet the needs of the business and their budget. The company's Web site is's GeoQuote compares in real-time the prices of the following T1 broadband and voice providers: ACC Business, New Edge Networks, PowerNet Global, XO Communications, Xspedius, USLEC, Netifice, AT&T, TelePacific, Broadwing, MegaPath, and Nuvox.

About is a wholly owned subsidiary of Telarus, Inc, delivering real-time VAR location information through its breakthrough VARSearch(tm) technology. Visitors are driven to by Telarus's growing network of online marketing specialists as well as by professional telecommunication sale consultants. is a free match-making service that connects people looking for telephone and network equipment installers and dealers with the VARs in their local area who provide these valuable services. Some of the marque brands offered by our VARs are Cisco, NEC, Avaya, Toshiba, Nortel, Vodavi, Extreme Networks, Juniper, Foundry Networks, and Artisoft.

Friday, July 14, 2006

VoIP Technology Shows Significant Promise For Call Center Operations

Before plunging into VoIP head first, it's important that businesses understand just what they are "talking" about when they begin looking at VoIP technology for their call center operations. Understanding these subtleties will ensure proper planning and appropriate decisons. The first key is to realize that "VoIP" is the basic term where cost alone seems to be the driving incentive. But IP Telephony is so much more. Unlike VoIP lite, IP telephony is not simply about cost savings. The benefits of IP Telephony to call center operations include rich applications, enabling mobility, increased productivity, and enhanced business continuity.


VoIP is the basic transport of voice in a packet form on an IP-based data network. It is the transmission of telephony over a data network and offers little in the way of features and functionality. IP Telephony uses VoIP but is a software application suite offering rich feature applications. These often-modular applications lend themselves to cost-effective integration with other applications that share the IP network. Voice and Data Convergence may be defined as the integration of voice and data applications in a common environment. Of particular significance is the integration of communications applications with key business applications. The latter are usually tied to business processes, which are central to an organization’s operations.

IP telephony lends itself to contact centres for the ease of integration with sophisticated multimedia applications as well as computer telephony integration, intelligent call routing and distributed or virtual contact centre applications. The merging of voice and data applications, such as Unified Messaging, is perhaps indicative of where IP telephony as a voice-based application leaves off and convergence starts.

Voice and data convergence may be construed as the coming together of voice and data in a common environment. This simplistic definition belies the significance of convergence. The reality is that converging voice and data enables the integration of voice communications applications (such as teleconferencing and speech access) with key business applications (such as sales force automation and supply chain management). These business applications are predicated on business processes that are the lifeblood of most organizations. By marrying these applications on any network and on any device, the door is opened to deriving new levels of business value.


In recent years the number of companies looking to up grade their call center infrastructure via implementation of VoIP technology has grown dramatically. Merging voice and data on a single network and deploying an IP-based contact center platform allows companies to route calls to home and satellite offices more efficiently. This approach is delivering on the promise of helping companies grow their business, apply productivity enhancing applications, and expand call center operations easily and cost effectively. Scaling for growth to new remote service centers is a smooth transition as each is treated as an add-on node to the existing IP network.

Companies can add remote staff to call center queues when needed and can retain key employees by letting them work from home. The entire process can utilize one application to manage all media for routing and reporting across agent locations. An additional benefit is the ability to deliver business applications over this new network when necessary.

Potential hurdles to implementing pure VoIP include preparing the network with switch and router upgrades, replacing all the desktop phones, and upgrading adjunct systems such as voice mail. These are not insurmountable issues and can either be accomplished all at oce (shotgun) or in a phased in approach. However it is accomplished the business benefits far outweigh any initial challenges.
It's safe to say that the great migration to the IP contact center is well underway. While there are many approaches, vendors and users agree that the decision is not driven by the technology, but rather by business applications that the technology enables. While companies may appear to take very different paths to VoIP, each is able to make the right decision for their current and future business needs from a myriad of solution options.


In general, however, the migration is happening very slowly. Art Schoeller, an analyst at The Yankee Group, says, "The move to IP in the contact center is inevitable but not imminent. The transition from TDM to IP, catalyzed by Cisco, is much like the transition from analog to digital systems, which was catalyzed by Rolm. Like that transition over 20 years ago, this transition will take time. And this one is more complex."

Where this transition seems to have found it's lead is among smaller business entities. Most IP contact center installations have occurred in small to midsize businesses (SMB). Many of these SMBs use home agents and remote offices. SMBs tend to be more willing than larger companies to take risks, many are growing, and they are reaping the benefits of flexibility and agility. Seeing this untapped potential larger businesses are begining to follow suite albeit at a somewhat slower far.

As of now there are fewer large installations in place, and they are generally multisite, often with overseas positions (including outsourcers). The major system vendors such as Avaya, Cisco, and Nortel all say they have pure IP installations of 2,000 seats or more. That's impressive...and it works. It won't be long before the pace and numbers of installations among larger companies grows significantly. They won't long be able to deny the benefits offered and the potential positive impact on process and cost efficiencies.

"The industries making radical changes are the ones who are suffering the most pain from economic and market forces, such as teleservices [outsourcers], airlines, telecom and high-tech companies," says Lawrence Byrd, a convergence strategist at Avaya. "These companies are seeking substantial cost savings from infrastructure consolidation, for example reducing 30 separate [automatic call distributors] to one or two, moving away from the complex and expensive network routing architectures of the 1990s, and intelligently routing the right customer to the right agent, wherever they are."

"These companies understand that they must make more significant investments in network optimization, as well as changes to their business processes and how they manage their people. But they are willing to do so for the payback offered. IP telephony in the contact center is the technology enabler for such transformation," he says.


Today, many of the large call center installations - those exceeding 200 seats - are hybrid solutions, some sites are TDM, some are IP. Companies use IP trunking between sites and IP to some desktops, for example, at new sites or sites where the switch has been upgraded. The traditional PBX can serve as a gateway, converting between TDM and IP.

Businesses with multiple locations are turning autonomous sites into satellite sites, significantly reducing the numbers of servers, applications and licenses required for functions such as routing, reporting, Computer Telephony Integration (CTI), quality monitoring and workforce management.

Another trend is higher adoption rates in Europe/Middle East/Africa and Asia Pacific. North America is generally slower to adopt IP contact center technologies because of more conservative and risk-averse decision-makers, and more large installed systems. However, of Cisco's 1,500 installations worldwide for example, approximately half are in North America.

Yet another trend is for companies to adopt VoIP in the enterprise first and then in the contact center. Gartner analyst Bern Elliot says IP system sales already have overtaken TDM system sales for corporations, but "IP adoption in the call center will lag." Elliot predicts that traditional TDM-based call centers will remain the dominant architecture for new system sales in North American until mid-2006. IP-based call center systems comprise approximately 10% of new system sales today.


Businesses leery of IP contact centers typically express concerns about security, quality, reliability and scalability. Early implementers say they faced challenges, primarily with quality of service, but they used assessment, configuration, testing and monitoring to successfully address those issues. However, the rule of thumb is that if you've done what you need to do for your network for other applications, running phones on IP is not a leap of faith.

Many early implementers say voice is more secure and more reliable over IP than it was in a TDM world, and the enhancements to their networks for voice also have benefited their data applications. For example, many clearly saw the potential benefits for growth, flexibility and disaster recovery.

When a significant disaster occurs and a business must trigger its disaster-recovery plan, it is a relief to easily be able to add seats at other sites and reroute calls quickly, with no effect on service. It is also reassuring when system continuity enables the following of the rigorous security processes applied to all other applications for your voice and call center applications.

Many companies have found that TDM is just too expensive for what they want to do. Often they'll discover that a pure IP solution offers their company lower total cost of ownership than TDM, with additional savings over time by avoiding proprietary hardware. Frequently they'll also see benefits from virtual operations across sites and CTI in hours instead of months. Also seen have been savings on wiring, moves, adds and changes, and networking of remote locations, while buying flexibility for the future including multimedia enhancements.


The breakthrough in adoption of IP in the contact center will occur as more companies share evidence that it is low risk, it works, and there are quantifiable business benefits. Any initial trepidation will soon disappear as companies recognize that VoIP is a technology that's right for them.....and whose time has come for the call center industry.

Tip....for assistance in finding just the right fit in a VoIP solution for your call center operation take advantage of the FREE consulatative services at Business VoIP Solution.

Wednesday, July 12, 2006

Just What Is MPLS, What Does It Do, And Where Can You Get It?

Your boss has put you in charge of finding a replacement for your frame relay. Your service is slow, expensive, and inflexible. Someone in your IT department mentions MPLS, but those initials are foreign to you.

Don't on to learn what MPLS is and what it can do for you:

* Multi-protocol label switching (MPLS) was invented to solve the problem of bridging multiple disparate protocols such as Frame Relay, ATM and Ethernet.

* Over the last few years, MPLS has won the battle within core networks and has become the dominant internetworking technology.

* Carriers have deployed IP/MPLS core routers in order to support existing legacy networks more cost effectively.

* Fundamentally, MPLS employs an encapsulation technique providing internetworking between different technologies, coupled with signaling protocols necessary to discover, configure and manage connectivity.

* In addition to signaling protocols, MPLS uses resiliency protocols, such as Fast Re-route and Bi-directional Fault Detection, to determine failures and in turn switching to standby links.

* The strength of these protocols combined with the popularity of MPLS core networks have made MPLS a logical choice for extending into metro networks.

* Vendors have responded by creating MPLS-enabled multi-service platforms with a mixture of legacy port options.

MPLS combines the multi-service and traffic management capabilities of ATM with the scalability of packet networks to create a best-of-breed service provider network. Key drivers towards MPLS deployment include:

* Cost reduction through data network convergence: MPLS facilitates the convergence of disparate Frame Relay, ATM, Ethernet and IP networks onto a single infrastructure to reduce capital and operational expenses.

* Integration of voice, video and data services: MPLS's traffic management capabilities enable this services "triple play" on a common backbone.

* New high-margin revenue opportunities through MPLS-based service offerings: MPLS's flexibility, high availability and multi-service support enables service providers to offer strict SLAs, increasing revenue and margins.

MPLS can lower operational costs by integrating multiple services onto a common backbone; has integrated capabilities which enable service providers to offer strict SLAs and thereby increase revenue and margins; and facilitates the convergence of disparate Frame Relay, ATM, Ethernet and IP networks onto a single infrastructure to reduce capital and operating expenses.

Next question.....which providers offer this service and how do you find out if it is available in all of your locations?

Easy answer....take advantage of the free consultative services at Their staff specializes in finding just the right bandwidth solutions to fit any business application requirement. They'll do all the research for your company including negotiating the best rate from available vendors in your area.

Monday, July 10, 2006

Vonage Stock Continues To Take A Nose Dive

This may not be strict broadband tech news....but it's still timely and important.

Vonage stock continued it's downward spiral closing at $7.40 a share's lowest level yet. That's quite the drop from their opening IPO price of $17. Hmmmmm.....seems like I said something like this would happen in a warning post here just before Vonage went public.

Given how Vonage is put together, their existing high business costs at the time, their lack of any substantial presence in the enterprise QoS arena or even real small-medium business enhancement markets....their stock performance shouldn't be any real surprise to anyone who had their eyes open (and hopefully their wallets closed). If eyes weren't open before you can bet that investors are wide awake now. It appears that there's no real return-on-investment strategy for this debacle....unless a buy back is in the offing and a return to private equity.

This whole situation is further compounded by yet another patent infringement suit against Vonage. Being in the legal hot seat once AGAIN for allegedly infringing on patents couldn't come at a worst time. The potential costs of damages and royalties worth about $180 million will just further add to th huge losses from the failed IPO. The whole law suit issue will probably contribute to further declines in stock price too.

In contrast....Packet8 is a picture of health compared to Vonage. As well as perfectly positioned in the right markets with their business VoIP solutions driven by their virtual office IP PBX and enterprise packages. Some folks hype the jingle (Vonage)...and some just build a business (Packet8). With the results really pretty predictable.

What does this mean to consumers? Vonage isn't someplace I'd want to have my faith in right now. Their prices are very likely to go up....and they are perfectly ripe for a takeover from any number of directions (can you say Baby Bell?). If that happens expect service and support to erode, features to be streamlined, and consumer costs to take yet another hike. With that in mind I suggest businesses of any size play it smart and head to Packet8 if not already there. Much safer and more stable for the long haul.

Friday, July 07, 2006

Tier One vs Tier Two Internet Service Providers

ISP Tiers

The "tier" terminology has arisen mainly in the Internet Service Provider (ISP) industry press to describe ISP size and connectivity. The lower the tier number, the bigger the ISP.

"Tier One" providers are the national/international backbones. There are between one and two dozen of them, and include the original bad boys Sprint, MCI, UUNet and BBN Planet, plus newer Fiber Optic Networks like Global Crossing, and Qwest. To be a Tier One provider, you need cross-national backbone links of T3 speeds or higher (45Mbps), and peer at one or more of the major peering points (multi-company POPs for sharing Internet traffic). Some Tier Ones have dial-up access arms, but they tend to focus on high-speed, high-reliability leased lines.

A "Tier Two" provider is an ISP who is connected to a Tier One provider. These tend to be regional (multi-city) networks, although this does include some nationals and some locals. Most Tier Two ISPs provide a wide range of services from dial-up to web hosting and dedicated lines.

A Tier Three provider would be one connected to a Tier Two provider, and so on. These ISPs tend to be smaller shops that focus on one service, such as dial-up access or web hosting.

Tier One is Better?

The "tier" terminology is a useful metaphor for describing the ISP industry today. However, like any metaphor, it is a simplification. Many people mistakenly believe that lower-numbered tiers are "better" than higher-numbered tiers. In reality, a provider at any tier can have good connectivity, and different solutions are appropriate for different needs. In general a Tier 1 IS usually better than a Tier 2.....but not always. The key is be smart and do your homework on what you need...and what "they" can provide.

The first misconception is that more bandwidth is always better than less. While that sounds obvious, what is much more important is bandwidth utilization. I'd rather be on a T1 that peaks at 50% use than a T3 that is always clogged at 100%. Providers at every tier go through periods of overutilization, and providers usually have different levels of utilization at different POPs. While UUNet may be a great choice in Chicago, it may be a lousy choice in San Francisco (a fictitious example!).

The second misconception is that bandwidth is the only important aspect of an upstream provider. Connectivity and reliability can be as important, or even more important to some organizations.

Connectivity is how far the network is from the major backbones. A provider that has a T1 to Sprint and a T1 to MCI has better connectivity than a provider that has two T1s to the same company.

Reliability is affected by the type of link you choose and by your provider's redundancy. A leased line (56Kbps or T1) will be more reliable (and more expensive) than a switched service like ISDN. Redundancy is how many ways out your Internet traffic can take. Again, this varies not only from provider to provider, but from city to city. Big Net may have five T1s leaving its home city, but only one leaving yours.

How to Choose

In the end, it generally doesn't matter a gnat's whisker what tier your provider is on. What matters most is cost, service, bandwidth utilization, connectivity and reliability. Focus on those parameters overall rather than simply the phrase "Tier".

If money is no object while 24-hour availability is crucial, a tier one provider will be your best choice. On the other hand, if you need technical assistance, web hosting, and other services besides a simple data pipe, a tier two provider may make sense. And so on.

What tier are you on? It may not matter as much as you think. Than may.

For assistance in determining just the right provider to meet your business requirements.....take advantage of the free consultation provided by

Wednesday, July 05, 2006

Bandwidth Speed Tests And Other Network Testing Tools

Many technical factors affect a person's or business's network performance. From the current workload, processing power and amount of memory on either end of the connection, to the protocols supported by the modem, to bottlenecks at intermediate devices or "hops," performance monitoring involves many variables. No one monitoring tool can fully capture all of these nuances.

So, the following.....although not a complete list by any means...are offered to help you find the right bandwidth testing tool to meet your specific needs. These cover such areas as speed test, line quality tests, packet loss, latency variability, bandwidth calculator, mobile speed, ping time, line monitoring, and much more.

TestMy.Net Accurate upload and download connection speed testing, with the ability to track your connection history. Though bandwidth testing is's primary service they also offer many other internet related tools, such as traceroute, ping, whois, DNS query. Tool box of bandwidth tools covering speed test, line quality tests, packet loss, latency variability, bandwidth calculator, mobile speed, ping time, line monitoring.

Iperf Iperf is a tool to measure maximum TCP bandwidth, allowing the tuning of various parameters and UDP characteristics. Iperf reports bandwidth, delay jitter, datagram loss.

CNET Bandwidth Meter speed test

NetWorx Networx is a simple, yet powerful tool that helps you objectively evaluate your bandwidth situation. You can use it to collect bandwidth usage data and measure the speed of your Internet or any other network connection. NetWorx can help you identify possible sources of network problems, ensure that you do not exceed the bandwidth limits specified by your ISP, or track down suspicious network activity characteristic of Trojan horses and hacker attacks.

Testing and Tools Network analysis software tools for LAN/WAN analysis, network bandwidth and device, Installation, analysis and monitoring of bandwidth path performance, and more.

ANL - CIS web100 based Network Diagnostic Tool (NDT) Located at Argonne National Laboratory, Argonne, Illinois U.S.A.; 1000 Mbps (Gigabit Ethernet) network connection. This java applet was developed to test the reliablity and operational status of your desktop computer and network connection. It does this by sending data between your computer and this remote NDT server. These tests will determine:

* The slowest link in the end-to-end path (Dial-up modem to 10 Gbps Ethernet/OC-192)
* The Ethernet duplex setting (full or half)
* If congestion is limiting end-to-end throughput
* It can also identify 2 serious error conditions:
- Duplex Mismatch
- Excessive packet loss due to faulty cables.]

List of Speed Tests

ESNet List Network Monitoring Tools reported to be in use at 11 ESnet sites in a survey made by the ESnet Network Monitoring Task Force (NMTF) and completed in October 1995. Try these popular traceroute, ip tracer, bandwidth speed test, email tracer, find ip address, cpu speed test, and IP lookup utilities.

Monday, July 03, 2006

Sony Ericsson Z520a (Video Phone)

The new Sony Ericsson Z520 is one hot little phone! Weighing in at just 3.3 oz, this tiny flip phone is loaded with technology including Bluetooth, a high-quality camera, video capture capability and two color displays. And, there are limitless ways to communicate with text messaging, picture and video messaging, AOL Instant Messenger and an email client built in. Along with a huge battery life and renowned Sony Ericsson quality, the Z520 is going to be one hot selling phone!

Here's the breakdown on features:


* Very small, lightweight flip phone
* Bluetooth Wireless Technology
* VGA-quality camera takes print-quality photos
* Capture video clips up to 60 seconds long
* AOL Instant Messenger, plus text messaging and multimedia messaging built-in
* Voice activated dialing
* Twin color displays
* Super-long battery life

What's In The Box With The Phone

* Additional Items Included - Battery, Wall Charger, User Guide

Advanced Features

* Digital Camera - Yes, VGA-quality (640 x 480 Pixel Resolution Max), 4x Digital Zoom
* Streaming Multimedia Support - Yes, 3GPP Support
* MP3 Player - Yes, Plays MP3 or AAC Music Files, Store Music In The On-board Memory
* Bluetooth Wireless Technology - Yes
* Video Capture / Camcorder - Yes, Up To 60 Seconds Length, MPEG4 Format
* Infrared Port - Yes
* Data Capable / Use This Phone As A Modem - Yes, With Software and Cable Sold Separately
* PC Synchronization - Yes, With Software and Cable Sold Separately

Messaging Features

* Mobile Web Browsing - Yes, WAP 2.0 Browser Built-in With One-Touch Access Button
* Multimedia Messaging - Yes, Via MMS
* Text Messaging (SMS) - Yes, Plus Pre-created Text Messaging Templates
* Email Client - Yes, Supports POP3, IMAP4 and SMTP Protocols
* Instant Messenger Built-in - Yes, AOL IM Built-in

Personalization and Fun Features

* Polyphonic Ringtones - Yes, 40 Chords
* Custom Ringtones - Yes
* Pre-loaded Ringtones - Yes
* MP3 Ringtones - Yes, Plus External Light Up Effects
* Ringer Profiles - Yes, Plays MP3 or AAC True Tones Music Files
* Picture Caller ID - Yes, Plus External Light Up Effects
* Multiple Languages - Yes
* Languages Supported - English, French, Spanish
* Games - Pre-loaded New York Nights By GameSoft, Plus Downloadable Java Titles
* Customizable Faceplates - Yes, Front And Back Covers Are Changeable
* Customizable Graphics - Yes
* Customizable Themes - Yes

Core Features

* Color Main Display - 128 x 160 Pixels, Over 65,000 Colors Displayed
* External Display - 101 x 80 Pixels, Over 4,000 Colors Displayed
* Speakerphone - Yes
* Voice-activated Dialing - Yes
* TTY Compatible - Yes
* To-Do List - Yes
* Voice Memo - Yes
* Alarm - Yes
* Calculator - Yes
* Calendar - Yes
* Mini-USB Port - Yes
* Vibrate - Yes, Plus External Light Up Effects
* Phonebook Capacity - 500 Entries
* Multiple Numbers Per Name - Yes, Up to 5 Entries Per Contact

Battery Life

* Battery Type - LiIon
* Talk Time - Up to 540 Minutes
* Standby Time - Up to 400 Hours

Technical Specifications

* Application Platform - Java
* Network Compatibility - GSM 850, GSM 900, GSM 1800, GSM 1900
* Data Download Speed - GPRS Class 10 (Up to 70 Kbps)
* Ringtone Types Supported - MIDI, MP3, AAC
* Predictive Text Entry - Yes, T9
* Built-In Memory - 16MB
* Dimensions - 3.2 in x 1.8 in x 0.9 in
* Weight - 3.3 oz

Compatibility Features

* Device Supports Voice Plans - Yes
* Device Supports Cingular MEdia Services - Yes
* Available For Purchase Without Service Plan - Yes

You can find a Sony Ericsson Z520a (Video Phone)....or search for other types of cell phones, accessories, and calling plans at CELLULAR.