Making use of fixed free Voice-over-Internet Protocol (VoIP) allows small businesses to expand without adding to their overhead costs, it has been claimed.
Samuel Schneider, marketing director at communications provider Bandwidth.com, said the prohibitive expense and maintenance demands of traditional Internet Protocol-Private Branch Exchange (IP-PBX) technology has led to the creation of a new hosted PBX model which requires monthly fees rather than an up-front investment.
"Phonebooth will absolutely bring an affordability to VoIP that hasn't been seen yet in the market, because it is free of any license fees and it does not confine small businesses to a certain number of seats, so they are free to grow," he added.
Last month, Bandwidth.com launched Phonebooth, an IP-PBX, which is a private telephone network used within an enterprise that can access a limited number of external lines.
The group claims current hosted models confine small businesses to a certain number of seats, whereas they need flexibility to sustain growth.
Friday, October 31, 2008
WFMSG Announces Asterisk Integration
Community workforce management solution provider, The WorkForce Management Software Group, Inc. (WFMSG), has reportedly developed and deployed an interface into theAsterisk engine, an open source telephony platform.
Released under the GNU General Public License (GPL), the Asterisk allows developers and integrators to create advanced communication solutions. This open source telephony engine is available for download free of charge. Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris.
Asterisk has also reportedly added various features to existing call center solutions by adding remote IP agent capabilities, advanced skills-based routing, predictive and bulk dialing, and so on.
"Interfacing into this platform broadens Community's appeal and reach into a wider market," says Peter A. Schmidt, WFMSG principal and chief technology officer.
"It is crucial that our solution interface is not only to the more traditional PBX platforms such as Nortel and Avaya but to more open platforms such asCisco, which we announced earlier this year, and now Asterisk," added Schmidt.
"We feel that presenting an end to end solution to our clients increases our value proposition," said Todd A. Cotharin, principal responsible for WFMSG operations.
"By adding Asterisk to our extensive library of active integration, we are further reducing cost to our customers. Reducing cost of entry and ownership, coupled with intelligent deployment processes that capture ROI, is the strategic difference for Community in the market”, stated Cotharin.
Cotharin also confirmed that the Asterisk interface is just one more facet of the company’s bundled approach that makes it easy for its clients to invest in the Community solution without worrying about unforeseen expenses.
WFMG has been offering contact center performance optimization professional services that are designed to maximize and leverage investments in people and technology.
VoIP Providers Using IP-Based E911 as Cost-Effective, Competitive Differentiator
GREENWOOD VILLAGE, Colo., Oct 30, 2008 -- E911 is commanding even greater attention in the voice-over-IP (VoIP) market as Monday's FCC ruling became more stringent and reports of casualties related to VoIP E911 limitations are publicized. Not only could the liability of ignoring emerging 911 capabilities be disastrous for a VoIP company, sales could also suffer as consumers are beginning to seriously evaluate 911 functionality in their voice buying decisions. To be competitive, VoIP companies are getting in front of FCC compliance and consumer demand for public safety. Historically, 911 technology enhancement was cost-prohibitive for VoIP providers, but rising IP-based E911 companies are allowing them to compete both on technology innovation and price, due to a lower cost infrastructure.
E911 companies are catering to the safety-conscious market. VoIP providers are using companies like VIXXI Solutions, an IP-based, nationwide E911 provider that focuses heavily on product innovation. They have developed efficient geospatial technology that far exceeds current compliancy standards while reducing costs for VoIP providers. In addition to supporting nomadic VoIP users, they provide unlimited caller-customized data, such as pre-existing health conditions, to a public safety answering point (PSAP). Additionally, they offer the ability to simultaneously notify relatives when a 911 call is made, with strong implications for the elderly living alone or parents of traveling teenagers.
These VoIP providers can now improve their time-to-revenue and the safety of their customers with immediate address validation in the 911 master street address guide (MSAG). The new Internet-based technologies additionally allow for real-time PSAP boundary updates ensuring correct call routing, a current FCC ruling. Also for compliancy, VIXXI uniquely provides their VoIP customers with 711 (deaf relay).
"It's important for VoIP companies to understand the risk of not providing the best E911 technology available, both from a liability and sales perspective. Consumers are considering 911 functionality and their family's safety when selecting a VoIP provider," said Stefanie Linnemann, Marketing Analyst for VIXXI. "The good news is that VoIP providers can protect their customers with minimal time and resource investment. Efficient IP technology saves our customers 20% or more off their current E911 costs."
While E911 historically has been a challenge that comes with the territory for VoIP providers, companies like VIXXI are making it easy to provide exceptional E911 functionality at aggressive prices. E911 is no longer a necessary distraction from a VoIP provider's core business; it has evolved into a means for VoIP companies to quickly and cost-effectively differentiate themselves in an area of high priority for voice consumers.
VoIP number portability introduced in South Korea
Telecoms Korea is reporting that number portability for internet telephony has been introduced in South Korea, the service will allow fixed line users to transfer to internet voice packages whilst retaining their existing number. The introduction of number portability for VoIP operators was originally due at the end of August 2008 but, according to TeleGeography’s GlobalComms database, dominant fixed line telco KT Corp put pressure on the Korea Communications Commission to delay the launch.
Thursday, October 30, 2008
Digium Reinvents AsteriskNOW
When Asterisk sponsor Digium, Inc. released AsteriskNOW, early in January 2007, the stated aim was to provide an open-source telephony "appliance." Not appliance as in hardware/software combo, but appliance as in 'all the software you need to get a complete telephony application up and running'—even if you don't know a lot about Linux administration.
AsteriskNOW contained all necessary Linux components "built in" (and nothing that was not needed) and came with AsteriskGUI, a simple graphical admin module.
The lean, mean offering has been very successful by any standard.
Last week, Digium announced the beta release its first major upgrade, AsteriskNOW 1.5, available for immediate free download.
There are two principal changes to the original AsteriskNOW 1.4 release: the replacement of the rPath Linux implementation with the highly popular and very stable CentOS distribution, and the integration of the powerful FreePBX admin/user interface—used in the popular trixbox and PBX in a Flash applications—as an alternative to AsteriskGUI.
The modular, customizable Web-based FreePBX UI, in particular, would seem to be moving AsteriskNOW in a new direction—away from the pared-down, 'only-what-you-need approach' more toward an 'add-components-and-build-complex-applications' stance.
Or, to put it slightly differently, AsteriskNOW 1.4 moved Digium's product line away from the telephony platform identity of Asterisk 1.2 and 1.4, toward a no-frills, easy-to-implement telephony application. The integration of the FreePBX UI would seem to be moving it back in the platform direction.
Indeed, in addition to simplifying the process like creating interactive voice response prompts and other messages, providing system statistics, aiding in setting up trunk lines, and the like, FreePBX excels in adding, removing, and configuring other software modules –SQL or CRM, for instance.
"Since its beginning, FreePBX has been instrumental in the adoption of Asterisk by a segment of the community who may not have otherwise taken the plunge," stated Philippe Lindheimer, leader of the FreePBX project. "We are delighted to extend our relationship with Digium and become part of this new AsteriskNOW distribution."
Echoing Lindheimer, Digium vice president of product management and strategy Bill Miller declared, "AsteriskNOW provides a distinct option for organizations that want the ability to create highly customized applications using Asterisk, along with the convenience of built-in Linux and other open source software. We're committed to Asterisk as an open source platform for telephony development, and AsteriskNOW is a major part of that commitment."Calling the VoIP Way
1. Know Your Options
Before plugging in, it helps to understand how the technology is changing the market. For starters, both traditional and VoIP providers now refer to what they sell as a PBX, meaning a system dedicated to a single business customer.
Old-school PBXs use traditional technology -- known as circuit-switching or TDM -- and are often called proprietary, because they are usually owned or maintained by your phone company. Circuit-switched PBXs remain common with big companies; small-business versions are less robust.
Now there's VoIP. Here, the phone system piggybacks on the computer network's cables and routers; an Ethernet cable from a wall jack connects to the phone at each workstation. Calls among far-flung offices will be routed through the Internet and your IP PBX, meaning you escape telephone charges. The features are mostly functions of software, not hardware. As a result, VoIP lets small businesses get big-time features. And if the sound isn't quite landline quality, it's improving fast.
2. Choosing Among VoIPs
Within the VoIP universe are two basic options, each with its advantages and disadvantages. Use the dollar figures below as general guidelines; prices can vary dramatically, depending on the provider and when you enter this changing market.
Hosted VoIP A hosted VoIP system is housed and managed off-site by an independent company (which may be a traditional telephone company playing in the VoIP arena). Your company connects to the host via the Internet.
Cost This can be an inexpensive arrangement, not least because there are some 700 companies in the U.S. alone competing to sell hosted VoIP systems. And it costs hosts little to add incremental clients to their networks, because they usually route outbound calls over the Internet, which avoids the telephone network's origination and long-distance charges.
Most commonly, the host charges per extension and by the amount of external calling time. Plans can also be tailored for unusual call patterns, such as frequent international dialing or faxing. It's important to analyze your call patterns before selecting a plan. (The list of features -- caller ID, on-hold music, advanced voice-mail capabilities -- doesn't vary much among plans.)
In addition, there are usually one-time fees for installation, licenses, and network equipment; depending on a company's size, the cost can exceed $2,000. Plus, you will need to buy special IP phones, sometimes available only from the host. These start at around $150 per phone. (You can often rent the phones as part of a monthly package, but this is seldom cost-effective in the long run.) Bottom line: After the start-up costs, a company can expect to spend at least $30 to $40 per extension per month for a basic system with unlimited domestic calling.
Pros Hosted VoIP has relatively low up-front costs and fast start-up. Most hosts frequently upgrade their service, which may protect you from technological obsolescence.
Cons After you have bought phones and sometimes a service contract, it's not easy to switch providers to get a better deal. It can be difficult and expensive to obtain specialized applications, such as those for call centers, from a host. Also, a hosted phone system puts you at the mercy of your Internet connection. "If that goes down," says Mike Zygiel, a network consultant in Bridgewater, Massachusetts, "you're down." Finally, the industry hasn't stabilized; companies come and go and offer varying sound quality and reliability.
Note Some providers offer a scaled-down hosted service called vPBX, which allows a very small company with remote branches or employees to receive inbound calls at a single phone number, at which a virtual receptionist passes the calls to the appropriate landline or cell phone. These services generally begin at about $15 a month for a limited amount of inbound minutes. However, you will still have to rely on your carrier for outbound calls.
Private VoIP Your company can buy its own VoIP hardware and software, which manage internal communications such as calls within and among offices. External calls are typically sent over phone lines, but now you have the option of using a technology called SIP, or session initiation protocol, to route them over the Web, with cost advantages similar to those of hosted services.
Cost A proprietary VoIP PBX, designed by a network consultant, runs about $500 to $800 per extension. Add a maintenance contract, and you will pay $10 to $16 per extension per month (amortized over five years), plus phone service charges.
Pros Besides being much more customizable, private VoIP will probably be cheaper than a host over the long run.
Cons There are high up-front or financing costs. You will need to maintain the system, which requires in-house expertise or a service contract (typically 10 percent of the initial investment, says Jennifer Huxley of Telec, a VoIP consultancy in Springfield, Missouri). If your calls are routed over phone lines, you will pay extra. And private VoIP is generally unsuitable for very small companies. "It only starts making sense to buy a system when you've got 10 employees," says Tom Wales, an independent consultant in Maine and Connecticut.
Note Some consultants will install a network at your facilities but maintain ownership of it. They will manage it on your behalf, charging a monthly fee -- a sort of hybrid of hosted and private. This typically costs from $45 to $55 per outbound line per month, says Jeff Ott, CEO of TotalCarrierSolutions.com, a telecom reseller in Missouri City, Texas.
3. Get a Line to the Outside
Whichever system you choose, you will need a hardwired connection to the outside, either to the phone company or to an Internet provider. Phone companies often bundle phone and data services. Any VoIP platform that sends calls over the Web, even to a host, will need substantial bandwidth. A T1 line, says Ott, can carry up to 15 VoIP conversations at a time and costs $300 to $600 a month, plus up to $90 for what's called quality of service (see "A Telecom Glossary" on the following page).
It's a good idea to have separate Internet services for voice and data, to serve as backup should one go down. "I always recommend redundancy, because now, if you lose your Internet access, you lose your voice and Internet," says Ott. "Most businesses can afford to lose one -- but not both." If your proprietary system relies on SIP, consultants recommend that you build in telephone network redundancy as well.
Hacking VoIP--New from No Starch: New Book Shows How Easy it Is to Attack VoIP
San Francisco, CA, October 29, 2008-Voice over Internet Protocol (VoIP) is an increasingly widespread new technology that allows users to escape the tyranny of big telecom and make phone calls over the Internet. But while VoIP may be cheap and convenient, it's notoriously lacking in security. With little effort, attackers can eavesdrop on conversations, disrupt phone calls, inject content into existing conversations, change caller IDs, and access sensitive information-all without the awareness of the VoIP users making the phone calls.
Voice over Internet Protocol ( VoIP ) is an increasingly widespread new technology that allows users to escape the tyranny of big telecom and make phone calls over the Internet. But while VoIP may be cheap and convenient, it's notoriously lacking in security. With little effort, attackers can eavesdrop on conversations, disrupt phone calls, inject content into existing conversations, change caller IDs, and access sensitive information—all without the awareness of the VoIP users making the phone calls.
Hacking VoIP approaches VoIP security from two angles, explaining VoIP's many security holes to both hackers and administrators. The book raises awareness of the importance of VoIP security, describes potential attacks, explains VoIP's biggest weaknesses, and offers solutions for protecting against potential exposure and attacks. Readers learn how to defend against VoIP attacks as they explore issues with VoIP security and the boundaries of VoIP protocols.
"VoIP is fun, but it's remarkably easy to attack," said No Starch Press founder Bill Pollock. "People think that when they pick up the telephone they're on a secure line, but not when that call is being made over VoIP. Hacking VoIP demonstrates just how easy it is to attack VoIP, and how best to plug those security holes."
Hacking VoIP explains every aspect of VoIP security, discusses popular security assessment tools, and explores the inherent vulnerabilities of common hardware and software packages. Readers learn how to:
Identify and defend against VoIP security attacks such as eavesdropping, audio injection, caller ID spoofing, and VoIP phishing
Audit VoIP network security and assess the security of enterprise-level VoIP networks such as Cisco, Avaya, and Asterisk and home implementations like Yahoo! and Vonage
Use VoIP protocols like H.323, SIP, RTP, and IAX
Locate potential vulnerabilities in any VoIP network
Use both existing and newly released VoIP security tools
Whether setting up and defending VoIP networks against attacks or just having sick fun testing the limits of VoIP security, Hacking VoIP is every user's go-to source for VoIP security and defense.
For more information, to schedule an interview with the book's author, or for a review copy of Hacking VoIP, please contact Travis Peterson at No Starch Press ( nostarchpr@oreilly.com, +1.415.863.9900, x300 ), or visit www.nostarch.com.
Voice over Internet Protocol ( VoIP ) is an increasingly widespread new technology that allows users to escape the tyranny of big telecom and make phone calls over the Internet. But while VoIP may be cheap and convenient, it's notoriously lacking in security. With little effort, attackers can eavesdrop on conversations, disrupt phone calls, inject content into existing conversations, change caller IDs, and access sensitive information—all without the awareness of the VoIP users making the phone calls.
Hacking VoIP approaches VoIP security from two angles, explaining VoIP's many security holes to both hackers and administrators. The book raises awareness of the importance of VoIP security, describes potential attacks, explains VoIP's biggest weaknesses, and offers solutions for protecting against potential exposure and attacks. Readers learn how to defend against VoIP attacks as they explore issues with VoIP security and the boundaries of VoIP protocols.
"VoIP is fun, but it's remarkably easy to attack," said No Starch Press founder Bill Pollock. "People think that when they pick up the telephone they're on a secure line, but not when that call is being made over VoIP. Hacking VoIP demonstrates just how easy it is to attack VoIP, and how best to plug those security holes."
Hacking VoIP explains every aspect of VoIP security, discusses popular security assessment tools, and explores the inherent vulnerabilities of common hardware and software packages. Readers learn how to:
Identify and defend against VoIP security attacks such as eavesdropping, audio injection, caller ID spoofing, and VoIP phishing
Audit VoIP network security and assess the security of enterprise-level VoIP networks such as Cisco, Avaya, and Asterisk and home implementations like Yahoo! and Vonage
Use VoIP protocols like H.323, SIP, RTP, and IAX
Locate potential vulnerabilities in any VoIP network
Use both existing and newly released VoIP security tools
Whether setting up and defending VoIP networks against attacks or just having sick fun testing the limits of VoIP security, Hacking VoIP is every user's go-to source for VoIP security and defense.
For more information, to schedule an interview with the book's author, or for a review copy of Hacking VoIP, please contact Travis Peterson at No Starch Press ( nostarchpr@oreilly.com, +1.415.863.9900, x300 ), or visit www.nostarch.com.
Wednesday, October 29, 2008
Sangoma and 3CX Ink Interoperability Alliance
WASHINGTON, Oct 28, 2008 (ASCRIBE NEWS via COMTEX) -- Sangoma Technologies, a supplier of software centric media and signal processing hardware, announced it has partnered with 3CX, a SIP-based PBX provider.
The technology partnership will ensure interoperability of 3CX Phone System Version 7.0 for Windows with Sangoma NetBorder Express gateway cards. Through this alliance, 3CX will be able to provide businesses with the option of running their PBX application on a server that has an internally or remotely deployed gateway.
"3CX is completely based on the SIP standard, so providing support for Sangoma's NetBorder Express Gateway was the natural next step in our product's development," said Nick Galea, CEO at 3CX.
"We are pleased to have 3CX as a technology partner, as demand for NetBorder Express grows," said Frederic Dickey, Director of Product Management for the NetBorder product line.
"The offer of 3CX Phone System with Sangoma Gateway Cards delivers a software-based IP PBX that is simple to install, configure, and manage, with advanced VoIP features such as low call costs and integration with business applications," said Galea. "This helps our mutual customers increase productivity, reduce their telecommunication costs, and boost their profit margins."
Ditech Networks Enhances Voice Processing Platform for VoIP, 3G, and Web 2.0 Services to Offer Greater Flexibility for Network Operators, Enterprises
Ditech Networks, Inc., a leading provider of voice quality solutions to the world's communications industry, today announced that its industry-leading Packet Voice Processor (PVP) is now available in a smaller size that gives network operators and enterprises greater flexibility to support VoIP, 3G and Web 2.0 services. PVP now can be configured and deployed to support 1,000-4,000 sessions at the network's edge, and up to 16,000 sessions in the core of the network. This range and configurability make Ditech's PVP product the industry's most flexible voice quality platform for IP services.
Although the demand for VoIP, 3G and Web 2.0 services continues to increase significantly, the volume of IP voice traffic can vary widely at different points in the network. Network operators and enterprises need the ability to support a wide range of call volumes and the flexibility to increase capacity as demand for IP voice services grows. Ditech's PVP product can support early stage markets and services, in addition to supporting the high-capacity requirements of IP voice services in the core of the network.
"The new low-density PVP chassis has all of the same functionality and performance of the higher-capacity product, but implemented in a way that offers network operators and enterprises more deployment options," said Karl Brown, vice president of marketing at Ditech Networks. "Ditech's new form factor and session capacity ensures that PVP can meet the different network requirements of supporting the growth of IP voice services."
Ditech's PVP product now is available in two platform options: 13RU, 14-slot chassis for up to 16,000 sessions; and 5RU, 6-slot chassis for 1,000-4,000 sessions. Cards in the 6-slot chassis can be redeployed in the 14-slot chassis as the demand for network capacity increases. Both platforms are available with all three of Ditech's industry-leading applications: Voice Quality Assurance (VQA), Experience Intelligence (EXi), and Dynamic Transcoding. VQA non-intrusively and automatically detects and eliminates problems that degrade voice quality. The application mitigates network-induced impairments, impairments from the caller's environment, and impairments from callers' devices, in real time, on both ends of the call. EXi non-intrusively measures speech quality in real time, on every call. Information from EXi about signal, noise, echo and frame/packet loss can be input into the ITU G.107 E Model to compute MOS or R-Factor scores much more accurately than solutions that rely on packet loss and delay statistics alone. In addition, EXi statistics can be used directly in pre-configured reports, or integrated with existing network test and measurement systems. Dynamic Transcoding supports a wide range of compressed and uncompressed codecs, allowing service providers to directly connect diverse codecs without compromising scalability, network performance or call quality. The following codecs are available for both platform options: G.711 (I 1/4-law, A-law, with full support for Appendix II and intelligent packet restoration,) G.723.1, G.723.1a, G.726, G.729, G.729a, G.729b, G.729ab and iLBC.
Diagnostic tool updated for VoIP phones
A new version of a network diagnostic tool released last week includes extra support for VoIP phones.
BBX 4.5, from Integrated Broadband Services (IBBS), also has other newly-added features, including the ability to examine subscribers' in-home wiring remotely.
The company's vice president of products, Chris Anderson, said that studies have shown that people are beginning to use VoIP phones in the home in a bid to save money.
He added: "Our research identified the need for robust and integrated provisioning and diagnostic software for VoIP service.
"This release enables our customers to significantly improve their delivery of VoIP services to end subscribers."
Research released by Dell'Oro Group last month suggested that the IP telephony carrier market shrank by ten per cent sequentially to $861 (£470) in the second quarter of this year.
This was mainly due to service providers investing in projects rather than infrastructure, the company suggested, with the North American market exhibiting the biggest signs of this.
BBX 4.5, from Integrated Broadband Services (IBBS), also has other newly-added features, including the ability to examine subscribers' in-home wiring remotely.
The company's vice president of products, Chris Anderson, said that studies have shown that people are beginning to use VoIP phones in the home in a bid to save money.
He added: "Our research identified the need for robust and integrated provisioning and diagnostic software for VoIP service.
"This release enables our customers to significantly improve their delivery of VoIP services to end subscribers."
Research released by Dell'Oro Group last month suggested that the IP telephony carrier market shrank by ten per cent sequentially to $861 (£470) in the second quarter of this year.
This was mainly due to service providers investing in projects rather than infrastructure, the company suggested, with the North American market exhibiting the biggest signs of this.
Tuesday, October 28, 2008
Asterisk VoIP Hosted PBX – A Cisco Training Course
Rapid and trustful communication systems have become a norm in the modern world. For this reason, people are increasingly dependent on communication tools like the internet and the telephone which are the two major branches of the communication industry.
The telephone allows you to talk directly to anyone in any part of the world be it your friends, family or your clients. It is not only quick it is quite dependable too. This is such an important tool that now it has become one of the basic necessities in every home and business. Businesses depend quite a lot on this mode of communication and their life would get stranded without connectivity through a telephone.
The internet has also capture the world by storm through its ability to provide super fast and consistent means of communication. The amazing factor to the internet is all the applications that it offers for communication can be accomplished with very low cost. People can keep connected with other people through emails and chat.
The modern times has seen another advance in the field of communication by providing the latest communication tool, the Voice over Internet Protocol (VoIP) or the Internet Voice. This is very similar to talking over the regular telephone but it is much more advanced. This has become so popular in the modern times that people are seriously contemplating eliminating the use of the regular telephone and switching over entirely to the VoIP system.
VoIP is far superior to the conventional telephone in that it provides many more features and advantages which the latter cannot even dream about. VoIP is through the internet and the voice signals are converted into digital format and then transmitted to the other person. The digital format of the voice signals ensures that there is no loss of voice quality and the communication is quicker and crystal clear as compared to a regular telephone conversation.
A great attraction that VoIP provides is the cost factor. Using this system, a person can make long distance calls to any other place in the world at a very low price or even for free. When compared to calls made long distance over the regular telephone, the communication is much faster, clearer and much cheaper when using the VoIP service. So, you will end up with lesser bills over the month in comparison to using the regular telephone line.
VoIP provides other advantages too. Let us suppose you require a PBX in your organization. If you purchase one to be used with your regular telephone line, it will be quite expensive. But, if you purchase one and use it over the VoIP system, it will be much cheaper and highly consistent. The functioning of the PBX system is similar in both the modalities but if the PBX is used with the VoIP system, it will be through the internet rather than through the telephone line.
VoIP is easily obtainable in today’s times. You can find many service providers marketing the services on the internet who will provide business VoIP. But, if the purpose is to install VoIP system at home or if your intention is to make VoIP a career path, you will have to learn about setting up the VoIP system.
Installing a VoIP system is taught by Cisco Training Courses which has several such courses, one of them the Asterisk VoIP Hosted PBX.
If you opt for this course, you will be provided with all the training materials you will require to understand the process of setting up a VoIP system in any kind of setup. VoIP is beneficial for residential and business purposes. But, Asterisk VoIP Hosted PBX is more suitable for big organizations in meeting their communication needs.
Cisco provides all relevant information about the various VoIP PBX systems that are available and their utility. Using the Asterisk VoIP Hosted PBX, one can integrate all the phone lines in the organization into a single channel for communication that uses the internet to route the calls.
IP PBX is a powerful tool for communication within any office setup. Using this piece of hardware, the entire organization’s communication needs can be met and easily regulated.
If you are intending to take up VoIP as a career path, you need to contemplate about Cisco VoIP Training Courses. The advantage this course provides is the hands-on experience using the Asterisk VoIP Hosted PBX. VoIP is definitely a good career path what with the amount of popularity it is gaining by the day and the pace at which it is eliminating the need for regular telephones.
Learn the secrets behind voip qos as well as implementing the cisco voip for large corporations when you visit http://www.businessvoipcentral.com, for more free tips and strategies in VOIP broadband management.
FaxBack's NET SatisFAXtion VoIP Fax Wins the 2008 Internet Telephony Excellence Award
FaxBack is a provider of high-density VoIP-fax solutions. The company’s NET SatisFAXtion VoIP fax server helps organizations complete their business communication strategy beyond voice and data, by easily adding VoIP-fax capability to IP environments.
Through NET SatisFAXtion, people can easily send, receive and manage faxes from their desktop through email and business applications likeMicrosoft Word. By integrating with major document management systems, the solution facilitates HIPAA/SOX compliance. NET SatisFAXtion is also easy to install, via traditional phone lines or VoIP networks.
According to company sources, NET SatisFAXtion can capture inbound faxes and route them electronically to a remote user’s inbox, or to a remote printer using a variety of options. This capability is ideally suited for distributed companies with multiple locations.
Because NET SatisFAXtion also integrates fax machines directly to the VoIP fax server, it enables complete archiving and tracking of all fax traffic. The solution can also scale to an enterprise solution to support any size businesses as they grow.
The editorial staff of INTERNET TELEPHONY magazine recognizes the innovation and performance breakthroughs that the NET SatisFAXtion fax server brings to operators, SMBs and large enterprises, according toGreg Galitzine, editorial director of INTERNET TELEPHONY.
He noted that VoIP fax remains one of the final frontiers in the IP-communications industry, and appreciated FaxBack efforts to address the final challenges needed to reliably deliver IP-faxes on a large scale.
“We’re pleased to include FaxBack in our list of winners,” Galitzine added.
Mike Oliszewski, co-founder and CTO for FaxBack, thanked TMC ) for this prestigious award, as it helps to validate their leadership in the VoIP-fax industry. The company is also working extensively with carriers, through their recently announced certification program, to help them enhance the interoperability and reliability of their VoIP-fax solutions.
Oliszewski emphasized that this combination of product innovation and commitment to their partners makes FaxBack the leader in the VoIP-fax industry. They are pleased to see INTERNET TELEPHONY magazine recognize their leadership.
US Mid-Sized Business VoIP Integrated Access Services Expenditures by Size of Business, 2007-2012 is Out Now
DUBLIN, Ireland, Oct 28, 2008 -- Research and Markets ( http://www.researchandmarkets.com/research/8104d6/us_midsized_busin) has announced the addition of the "US Mid-Sized Business VoIP Integrated Access Services Expenditures by Size of Business, 2007-2012" report to their offering.
This Excel-based Data-rich Deliverable (DRD) that is part of the Business IP subscription includes market intelligence on business expenditures on VoIP Integrated Access Services (IAS) by size of business. Compass Intelligence defines Business IP as services used to carry and/or connect to the Internet. This category include Full and Fractional T1 and T3 connections, broadband services, dial-up Internet access, OC-X services and both TDM and VoIP-based Integrated Access. Compass Intelligence defines VoIP integrated access services as all integrated access services over IP connections that provide both voice and data access over a single connection. This would include VoCable, VoDSL and Voice over Packet services. Size of business includes Mid-Sized Business (100-999 employees). The Expert Guide for this deliverable is Kneko Burney. Forecasts are from 2007 through 2012 and include annual growth rate, as well as percentage of total market.
Sources: Our segment and market forecasts, which include business expenditures, market demographics, and usage and adoption statistics, are built using multiple sources, including our proprietary research. These sources include, but are not limited to:
-- Secondary research
-- Government data and statistics (e.g. department of commerce, federal communication commission, bureau of labour statistics and us census bureau
-- Primary research
-- Vendor-based research
-- In-depth interviews with key decision-makers (where relevant)
We select data sources to provide greatest degree of perspective on each market or segment, in addition to the highest level of data accuracy, stability, and consistency over time.
Monday, October 27, 2008
Xlink Cellular Bluetooth Gateway - Dump The Landline For Good!
Many folks these days are dumping their land based phone lines in favor of cellular, VOIP, and SKYPE services. Younger generations especially, appreciate the lower service costs, portability, and fun factor associated with these technologies.
One company, Xlink, is capitalizing on the switch to cellular.
Theyhave recently launched the Xlink Cellular Bluetooth Gateway. This handy little device pairs with your mobile phone via Bluetooth and essentially pipes your cellular service through the standard phone wiring in your home or office.
With the Xlink Bluetooth Gateway you can:
- Connect up to 3 Bluetooth enabled cell phones to your home phones
- Answer your cell phone calls from any phone in the house
- Fully utilize your cell phone weekend and evenings minutes
- Use in your home or at the office
The Xlink Blutooth Gateway comes in two flavors. One is for use without landline service, and one for use with or without landline service for those who aren't ready to cut the cord just yet. Given that land based phone service can cost $40-$100+ per month this device can possibly pay for itself very quickly! Go ahead, simplify your life, reduce your bills, and cut the cord for good!
It's about time
As we sift through the Troubleshooter inbox and its endless catalogue of technological snafus, often it's the minor troubles that catch our eye.
Thus it was with Frances Bottrell and her cordless phone. As Frances explained, she had recently "gone naked" - that is, switched her internet service provider (ISP) to iiNet and elected to make and receive all her telephone calls over the internet using VoIP (Voice over Internet Protocol). About the same time she bought a Uniden cordless phone, which was when her dramas started.
"Every time I receive an incoming call the phone changes time," she says. "Constantly it resets itself for two hours earlier. If you miss two or three calls a day there is no way of knowing when they called."
Bottrell's first port of call was Uniden but ironically there is no contact number on its website. Bottrell emailed the support address but to no avail.
"I rang iiNet, who assured me it was nothing to do with the connection but the support person said he had sort of, kind of, heard of that happening and to contact Uniden," Bottrell said.
At this point she contacted us.
We managed to find a phone number for Uniden in the White Pages and spoke to marketing manager Mark Willis.
He chased down the issue with his support staff. They had heard of the problem before, it turns out. But the fault was with the ISP, they said.
"Basically, what is happening is that the software which controls VoIP has an incorrect day/time stamp within it," said the support person. "Every time the customer receives a call, the day and time stamp on the phone software is changed."
By now feeling she was trapped in some sort of techno Groundhog Day, Bottrell called iiNet again to be told the problem was definitely not at their end.
Righto, then. Pausing only to grab the deerstalker hat and pipe, we called Andy McIntyre, general manager sales and marketing at iiNet. He in turn put the issue to his technical staff. Their theory was that the problem lay with the router Bottrell was using.
"They believe the time zone ... is not set correctly and is quite possibly the culprit, as it is sending incorrect time information to the handset when the SIP [Session Initiation Protocol] call comes in," McIntyre said.
One of the iiNet boffins gave Bottrell a call and talked her through the process of resetting the time zone for her modem and, as if by magic, all was suddenly well.
To say Bottrell was pleased would be something of an understatement.
We love a happy ending.
TelcoBridges' Technology Powers Aegis Mobility's DriveAssist™ Mobile Call Management System
Montreal, Canada, October 27, 2008 -- TelcoBridges™ today announced that the new DriveAssist mobile call management system recently announced by Aegis Mobility and Nationwide Insurance, is supported by TelcoBridges' award-winning hardware platform. DriveAssist is a network integrated software application that detects when a mobile phone user is driving, and it provides a set of user-defined options for managing incoming and outgoing calls, text messages, and internet access, much like a personal assistant.
"Cutting-edge solutions like DriveAssist illustrate a new era in mobile value-added services, and we're thrilled to help Aegis Mobility and Nationwide Insurance bring this technology to market," said Gaetan Campeau, President and CEO of TelcoBridges. "Solutions providers and operators looking to deploy services to a large user base are turning to TelcoBridges in increasing numbers - because we can provide the carrier-grade reliability and scalability needed to ensure that their innovative applications work flawlessly, anywhere and all the time."
DriveAssist utilizes all the sensors in the phone, including GPS, WiFi, baseband cellular, accelerometers, and proprietary technology created by Aegis Mobility, to detect motion in a manner that distinguishes between walking, and driving. While in motion, DriveAssist manages mobile phone use according to user-defined policies, set up through a web-based policy manager, which will not restrict E911 connections, or other designated emergency/priority calls - such as those coming from a business/fleet manager, or from a parent. More information on how DriveAssist works can be found on: http://www.aegismobility.com/.
"DriveAssist demands carrier grade equipment that will scale to tens-of-millions of subscribers, while providing 'five-nines' reliability - so we're running it on TelcoBridges' hardware - to ensure the capacity and the coast-to-coast availability we need to support our user base," said David Hattey, CEO of Aegis Mobility. "Their hardware and software have exceeded our expectations and TelcoBridges has been a tremendous partner to Aegis."
TelcoBridges connects DriveAssist to the mobile carrier's switching center via high-availability SS7 links. And TelcoBridges is also providing the IVR capabilities needed to store content and playback voice prompts on thousands of channels simultaneously. These capabilities form the core of the DriveAssist "personal assistant" functionality, which manages incoming calls and text messages while users are driving.
A recent National Highway Transportation Safety Association (NHTSA) study reported that more than 80-percent of motor vehicle crashes and near-crashes result from driver inattention. Another survey conducted by "Harris Poll" indicated that 90-percent of Americans believe that cell phone texting while driving is just as dangerous as drinking and driving, and more than 75-percent of respondents support legislation limiting cell phone use while driving. Today, 17 States currently ban cell phone use while driving, and every remaining state is considering similar measures. DriveAssist addresses the issues that can arise when drivers get distracted by their phones while out on the road.
Large Wi-Fi Contract Granted To Pixavi by StatoilHydro to Cover Their Facilities With Explosion Proof Wireless Lan
Stavanger, Norway October 27, 2008 -- In what is considered one of the largest contracts of its kind, Pixavi (previously VisiWear) has a while ago been awarded the exclusive right to supply it's range of explosion proof and Intrinsically safe Wi-Fi Wireless LAN, wireless network and antenna products to StatoilHydro, one of the largest offshore oil and gas companies in the world.
The contract involves wireless coverage on all StatoilHydro's Norway installations, both onshore and offshore.
"Clearly it involves a large quantity of wireless infrastructure products and at the same time It underlines our position as one of the leading vendors of industrialized and EX certified wireless network products," stated Managing Director, Christian Rokseth.
"We are very satisfied with this contract. Through the decision of building wireless networks throughout their assets, StatoilHydro has clearly seen the benefits of building a solid wireless infrastructure base in order to cut cabling cost, increase reliability and lower the maintenance cost of their installations. Further, they will be able to enjoy and profit on new and innovative applications like Pixavi's own intrinsically safe 802.11n wireless video communication technology, Location tracking products, Wireless CCTV products and also third party wireless VoIP, PDAs, Rugged laptops and wireless Sensor Technology in order to increase efficiency and safety on these installations."
"Wireless networks are increasingly being adopted in mission critical systems.", mr. Rokseth continues. "With basis on redundant frequency operations and reliable hardware and software technologies, Wi-Fi wireless networks are gradually replacing cabling in applications that previously were unthinkable because of lack of reliability and redundancy. The new 802.11n standard also offers close to the same network speed as cabling and thereby, wireless networking has definitely become a cost effective alternative to cabling within industrial communication applications."
Pixavi Sales Manager, Thomas Zaubi, stated: "Combining our Xpoint Explosion proof and Intrinsically safe access points with the Xbeam EX certified high gain antennas, including omnidirectional, sectorial and directional antennas and our ultra low loss coax cable series opts for a very powerful solution. It simply enables our customers to create a cost efficient, secure, solid and reliable wireless coverage in hazardous area zone 1 and zone 2."
The contract involves wireless coverage on all StatoilHydro's Norway installations, both onshore and offshore.
"Clearly it involves a large quantity of wireless infrastructure products and at the same time It underlines our position as one of the leading vendors of industrialized and EX certified wireless network products," stated Managing Director, Christian Rokseth.
"We are very satisfied with this contract. Through the decision of building wireless networks throughout their assets, StatoilHydro has clearly seen the benefits of building a solid wireless infrastructure base in order to cut cabling cost, increase reliability and lower the maintenance cost of their installations. Further, they will be able to enjoy and profit on new and innovative applications like Pixavi's own intrinsically safe 802.11n wireless video communication technology, Location tracking products, Wireless CCTV products and also third party wireless VoIP, PDAs, Rugged laptops and wireless Sensor Technology in order to increase efficiency and safety on these installations."
"Wireless networks are increasingly being adopted in mission critical systems.", mr. Rokseth continues. "With basis on redundant frequency operations and reliable hardware and software technologies, Wi-Fi wireless networks are gradually replacing cabling in applications that previously were unthinkable because of lack of reliability and redundancy. The new 802.11n standard also offers close to the same network speed as cabling and thereby, wireless networking has definitely become a cost effective alternative to cabling within industrial communication applications."
Pixavi Sales Manager, Thomas Zaubi, stated: "Combining our Xpoint Explosion proof and Intrinsically safe access points with the Xbeam EX certified high gain antennas, including omnidirectional, sectorial and directional antennas and our ultra low loss coax cable series opts for a very powerful solution. It simply enables our customers to create a cost efficient, secure, solid and reliable wireless coverage in hazardous area zone 1 and zone 2."
Wednesday, October 22, 2008
Free software offers cheaper long distance calls
This could help significantly narrow the digital divide "at the social level between rich and poor and geographical levels, between city and village," says Giovanni Maruzzelli, an Italian expert in the field currently touring India.
The Italian techie has held meetings at IIT-Madras, at Auroville, and at Mumbai, and now is scheduled to speak in Goa and Hyderabad.
Maruzzelli is the man behind the celliax.org project, that works with Internet telephony, computers, sound cards and mobile phones -- bringing all together in amazing ways.
Celliax uses second-hand, recycled and cheap cellphones as interfaces between VoIP and the GSM networks.
VoIP stands for Voice over Internet Protocol, and is optimized for the transmission of voice through the Internet or other packet-switched networks. GSM is Global System for Mobile communications, originally from Groupe SpéciaMobile.
"I'm still at the beginning of my trip. But in each place I've been very refreshed by, and glad to see, the people that come to the presentations of Asterisk-celliax-skypiax," Maruzzelli, 44, told IANS.
He said: "Voice communication, when it is managed by advanced technologies like Asterisk (the Open Source PBX and telephony platform) and VoIP, allows a large public to tap the same benefits of information access and interactivity that the Internet allows to the technical advanced part of the population."
In India, he said, "I want to get acquainted with the technical communities that relate (as users, developers, entrepreneurs, administrators, teachers, etc) to free and open source software. I'm making presentations about the free software that I'm now contributing to.
"I see that there is a precise awareness, also among people who have no technical knowledge, about how strategic the new voice communication technologies -- and mobile communication -- could be for India.
"It's much easier, on many occasions, for people to interact using a phone than using a computer. It is important to move toward an approach that combines low cost, low power, recycling, and sustainability."
Maruzzelli pointed out that India has "wide differences between countryside and the big cities. In such a context, organizations, communities, companies and public administration have to evaluate and use each tools that allows them to interconnect with and between people."
Maruzzelli's website www.celliax.org is the gathering point for the development of celliax, skypiax and directoriax technologies, that allow for a cheap interconnection between fixed lines, Skype, GSM, and VoIP.
The software he has worked on is used to connect the Asterisk PBX (www.asterisk.org) or private branch exchange to the GSM and Skype networks for making and receiving voice calls and SMSes.
Voice menus, the phone interrogation of databases, speech synthesis and recognition, automatic attendants - these are technologies ready right now to be implemented, he said.
There is a fast growing market for any technology that can save money in telecommunication, he added.
VoIP, Asterisk, FreeSwitch, and the other open source technologies allow for bigger savings, and for extreme flexibility. Both at the level of big telco and at the small office or tiny community level, Maruzzelli added.
Maruzzelli was founder of the first mass consumer Internet service provider and portal in Italy, partner in an incubator and venture capital private fund and an Internet and telecom investment expert for the World Bank-IFC in Serbia.
"So I know very well that if you start from technologies that have a high degree of usefulness and a great potential for penetration, you can build a viable and successful business," he added. "All the pieces are there, and I see a very bright future in India for all the open source technologies related to VoIP."
"The Indian elite technologists are the best in the world; but this is not news. With such a big population, India will have to grow a much bigger number of medium and advanced techies, who can bring about innovations in all parts of the country," he added.
How does an IP-PBX work?
If you're thinking about jumping into the world of VoIP, one option is an on-site IP-PBX. With products ranging from freeware open source to turn-key appliances, you'll find many alternatives for businesses of any size.
The term "IP-PBX" is commonly used, but it is slightly misleading because the roles the old PBX systems and new VoIP-enabled ones are not a one-to-one match.
In standard analog telephony, each phone is connected using pairs of wires to a central switch, called a PBX, which then connects to the local telephone company and long distance providers using more wires. Every call goes through the PBX and every telephony feature (such as automated attendant, call forwarding or voice mail) is a function of the PBX.
In the world of VoIP telephony, the function of the PBX is more distributed. For example, once a call is set up, the actual voice traffic can go directly from one IP telephone to another IP telephone without passing through the PBX.
Most IP-PBX systems are designed to work with IP-based telephones connected to an existing corporate LAN. While it's not strictly necessary, many network managers place the IP phones on a dedicated-and firewalled-VoIP LAN.
The IP-PBX itself is the only piece of the VoIP network which will need to talk to the Internet, either to deliver calls to a VoIP telephony provider or just to have access to necessary software updates. These Internet-based telephony service providers can accept calls from you using standardized protocols, such as SIP, and deliver those calls to local numbers, long distance, or international phone networks. The connection works both ways as well: an Internet-based telephony provider can take calls coming from the traditional phone network to your phone numbers and deliver them to you using VoIP protocols directly over the Internet.
Smaller companies with only a few analog phone lines can use analog gateways to make a connection to a regular public switched telephone network. If you're using digital trunk services, such as PRI ISDN or traditional T1 circuits, you'll need to have a PRI or T1 gateway. Some hardware IP-PBXes have analog, PRI and T1 gateways "on board," but it is just as common to have these gateways as separate, LAN-connected devices.
Smaller businesses may want to use hosted IP PBX services as an alternative to having their IP-PBX on site. Since the connection between IP telephones and the IP PBX is over, well, IP, there's no reason they all have to be co-located in the way a traditional PBX requires. Hosted IP PBX services offer the same advantage of many other outsourced products, including a more comprehensive feature set than is found in low-cost IP-PBXes and lower capital and operational costs. When a hosted IP PBX provider also sells local and long distance service, they may be able to cut those costs by aggregating business customers to gain a larger volume discount.
However, hosted IP PBX services have drawbacks that make them inappropriate for many businesses. The most common deployment model requires all calls, even those between people on the same LAN, to go through the IP PBX. This puts significant stress on Internet connections, requiring lower latency and greater attention to QoS controls. Small business Internet connections over asymmetric broadband connections such cable modems or DSL are rarely capable of providing acceptable service for more than a few users.
SIP Director honored for delivering exceptional VoIP/IP communications solutions
(Media-Newswire.com) - Radware, ( NASDAQ: RDWR ), the leading provider of integrated application delivery solutions for business-smart networking, announced today that Technology Marketing Corporation ( TMC ) has named SIP Director as a recipient of a 2008 INTERNET TELEPHONY Excellence Award presented by INTERNET TELEPHONY magazine ( www.itmag.com ).
Radware’s SIP Director is a first-to-market, fully SIP-aware application delivery controller ( ADC ), based on the company’s award winning AppDirector, an intelligent ADC solution. Designed with vendors, network equipment providers, system integrators, telecom equipment manufactures and carriers in mind, SIP Director was developed to address SIP-specific carrier-grade and security application delivery requirements, and answers the need for collaboration applications in a converged Web and VoIP application delivery environment.
“This INTERNET TELEPHONY Excellence Award by Technology Marketing Corporation recognizes Radware’s strong commitment to offering carriers the solutions they need to meet today’s fast-paced, high-volume infrastructure environment,” said David Aviv, Vice President, Advanced Services, Radware. “As SIP adoption increases, and the standard becomes more mission critical, carriers and operators will require solutions that can guarantee SIP service delivery. SIP Director offers the high availability, resilience, scalability and interoperability carriers need to meet their growing network requirements.”
“Advancing IP communications and providing real solutions in the marketplace has earned Radware recognition from the editors of INTERNET TELEPHONY and an INTERNET TELEPHONY Excellence Award. SIP Director has excelled in delivering value and distinguishing itself as a leading solution in the marketplace,” said Greg Galitzine, Editorial Director of INTERNET TELEPHONY.
“We are proud to present Radware with a 2008 INTERNET TELEPHONY Excellence Award. SIP Director exemplifies the spirit of the Excellence Award - making an outstanding contribution to IP communications,” indicated Rich Tehrani, Editor-in-Chief of INTERNET TELEPHONY.
Radware’s SIP Director is a first-to-market, fully SIP-aware application delivery controller ( ADC ), based on the company’s award winning AppDirector, an intelligent ADC solution. Designed with vendors, network equipment providers, system integrators, telecom equipment manufactures and carriers in mind, SIP Director was developed to address SIP-specific carrier-grade and security application delivery requirements, and answers the need for collaboration applications in a converged Web and VoIP application delivery environment.
“This INTERNET TELEPHONY Excellence Award by Technology Marketing Corporation recognizes Radware’s strong commitment to offering carriers the solutions they need to meet today’s fast-paced, high-volume infrastructure environment,” said David Aviv, Vice President, Advanced Services, Radware. “As SIP adoption increases, and the standard becomes more mission critical, carriers and operators will require solutions that can guarantee SIP service delivery. SIP Director offers the high availability, resilience, scalability and interoperability carriers need to meet their growing network requirements.”
“Advancing IP communications and providing real solutions in the marketplace has earned Radware recognition from the editors of INTERNET TELEPHONY and an INTERNET TELEPHONY Excellence Award. SIP Director has excelled in delivering value and distinguishing itself as a leading solution in the marketplace,” said Greg Galitzine, Editorial Director of INTERNET TELEPHONY.
“We are proud to present Radware with a 2008 INTERNET TELEPHONY Excellence Award. SIP Director exemplifies the spirit of the Excellence Award - making an outstanding contribution to IP communications,” indicated Rich Tehrani, Editor-in-Chief of INTERNET TELEPHONY.
Monday, October 20, 2008
Global IP Solutions AB: Successful Mobile VoIP on iPhones and Smartphones Whitepaper Available from Global IP Solutions
Global IP Solutions, the leading provider of IP multimedia processing solutions, announced today the availability of a whitepaper that highlights the challenges and opportunities critical for high quality when implementing mobile VoIP, called “Implementing VoIP for iPhones and Smartphones,” written by Roar Hagen, CTO at Global IP Solutions.
The paper addresses how developers can overcome problems inherent in mobile IP communications, such as packet loss and delay, jitter, acoustic echo and OS tuning. It also explains how applications can be VoIP-enabled to take advantage of the unique capabilities of today’s smart phones, like access to the internet. To download the paper for free, visit www.gipscorp.com.
Apple Inc.'s iPhone 3G was the top-selling smartphone in the U.S. from June through August, capturing 24 percent of the market, according to market research firm NPD Group, a market research firm for the wireless industry. The total smartphone market is growing rapidly, with revenues reaching nearly $1.7 billion already this year, according to NPD Group.
“Smartphones and especially the iPhone present the most promising opportunity for growth of the mobile IP market, and an excellent platform for developing applications that create real-time VoIP. However, mobile application developers, wireless service providers and handset manufacturers need to ensure the end user experience is first-class to ensure success in the market. GIPS audio processing expertise is recognized worldwide and ensures high-quality VoIP, even under adverse Wi-Fi conditions,” said Roar Hagen, CTO, Global IP Solutions.
Hagen continued, “These full featured devices have introduced ground breaking design and multimedia capabilities to the mobile world and their adoption rate has been staggering, even during this down economy. Application developers, wireless service providers and manufacturers can bring expert high quality voice and video applications to mobile devices in record time with GIPS’ MediaEngines. This will allow both the enterprise and consumer mobile community to benefit from new IP communication capabilities on the road, such as VoIP calling, social networking and video conferencing as well as integrated multi-player chat.”
GIPS is recognized for its media processing expertise for IP communications with the invention of the IETF standard iLBC. With more than 800 million downloads globally on various platforms, GIPS continues to innovate by enabling VoIP on mobile devices, allowing accelerated time-to-market with cutting edge Mobile VoIP applications.
XO Communications Marks 15,000 Business VoIP Customers
XO Communications announced that its industry-leading business voice over IP services bundle, XO IP Flex, has been deployed by more than 15,000 businesses and supports more than 475,000 customer employees.
-- As a converged IP services solution, XO IP Flex offers businesses all the benefits of IP for voice, Internet access and web hosting with enhanced features, functionality and value for one flat monthly price without having to replace their existing phone systems.
-- Launched nationwide in April 2005, XO IP Flex has become one of the most widely deployed VoIP services bundles for small and medium-sized businesses and distributed enterprises across a wide range of industries. XO Communications has continually enhanced XO IP Flex with new features including higher-bandwidth options, integration with XO's MPLS IP-VPN service, adding Unified Communications capabilities with XO Anywhere, and offering the industry's first simplified, bandwidth-based pricing for converged IP services.
-- XO IP Flex's unique pricing simplifies how businesses can buy and scale IP services to support their communications needs. Unlike other approaches to IP pricing that still are based on traditional TDM services pricing models, XO's bandwidth-based pricing acknowledges that voice is simply another IP application and offers rates based on the size of the port, not on the number of voice lines. Customers simply select an IP port speed from 1.5 to 45 Mbps, a calling plan and any additional features. Because voice is just another application on the IP port, customers pay nothing for incremental lines or voice channels provisioned within the port speed they have with their XO IP Flex service.
-- The combination of XO IP Flex and XO Anywhere enables customers to take advantage of the capabilities of VoIP and Unified Communications without having to make capital investments in new equipment, devices or software.
TELES VoIP and Mobile Gateways receive Siemens "HiPath ready" certificate
TELES, the Berlin based access gateway vendor, has been awarded Siemens Enterprise Communications "HiPath Ready" certificate following a rigorous interoperability test. TELES participated in the HiPath Ready program in order to meet the telecommunications market's requirements for speed and interoperability.
As part of the �HiPath Ready� certification program, Siemens checks the compatibility of partner products and their ability to integrate with selected HiPath systems. On behalf of the Technology Partner, Siemens performs extensive integration and function tests.
The TELES Access Gateways product family enables PBX connection to VoIP networks, TDM, and mobile networks (GSM/UMTS). Following the successful completion of the interoperability test, existing HiPath 4000 and OpenScape Voice SIP-based softswitches now have an alternative for extending their services to all kinds of networks.
When asked about the benefits of certification, Thomas Haydn, TELES Product Marketing Director, said: "We are proud to receive the HiPath Ready award in recognition of the compatibility, reliability, and flexibility of TELES products and their ability to fit in to the Siemens HiPath and OpenScape Voice systems."
The TELES VoIPBOX provides HiPath and OpenScape Voice customers with a VoIP gateway with several TDM interfaces, transcoding, SIP to H.323 protocol conversion, remote survivability, and emergency break-out support. In addition, the TELES iGATE mobile gateway enables VoIP/TDM calls to be terminated in GSM/CDMA and UMTS networks, allows email to SMS and SMS to email conversion, as well as enabling least cost routing to different mobile networks.
About TELES
TELES provides worldwide wireline and wireless telephony service providers with complete Class 4 and Class 5 soft switch solutions, VoIP gateways, and mobile gateways for GSM, CDMA, and UMTS networks. Celebrating, this year, our 25th anniversary, we are proud of our successful track record having deployed more than 200 live carrier and service provider networks with millions of voice ports.
A Deutsche Boerse Prime Standard listed company with headquarters in Berlin, Germany, TELES is a global NGN player with offices in Western Europe, Asia Pacific, and Latin America. We maintain a worldwide network of distribution channels, system integrators, and technology partners.
As part of the �HiPath Ready� certification program, Siemens checks the compatibility of partner products and their ability to integrate with selected HiPath systems. On behalf of the Technology Partner, Siemens performs extensive integration and function tests.
The TELES Access Gateways product family enables PBX connection to VoIP networks, TDM, and mobile networks (GSM/UMTS). Following the successful completion of the interoperability test, existing HiPath 4000 and OpenScape Voice SIP-based softswitches now have an alternative for extending their services to all kinds of networks.
Find Solutions for Enterprises, SMBs & Service Providers at the INTERNET TELEPHONY Conference and EXPO West, September 16-18, 2008. Los Angeles, California.When asked about the benefits of certification, Thomas Haydn, TELES Product Marketing Director, said: "We are proud to receive the HiPath Ready award in recognition of the compatibility, reliability, and flexibility of TELES products and their ability to fit in to the Siemens HiPath and OpenScape Voice systems."
The TELES VoIPBOX provides HiPath and OpenScape Voice customers with a VoIP gateway with several TDM interfaces, transcoding, SIP to H.323 protocol conversion, remote survivability, and emergency break-out support. In addition, the TELES iGATE mobile gateway enables VoIP/TDM calls to be terminated in GSM/CDMA and UMTS networks, allows email to SMS and SMS to email conversion, as well as enabling least cost routing to different mobile networks.
About TELES
TELES provides worldwide wireline and wireless telephony service providers with complete Class 4 and Class 5 soft switch solutions, VoIP gateways, and mobile gateways for GSM, CDMA, and UMTS networks. Celebrating, this year, our 25th anniversary, we are proud of our successful track record having deployed more than 200 live carrier and service provider networks with millions of voice ports.
A Deutsche Boerse Prime Standard listed company with headquarters in Berlin, Germany, TELES is a global NGN player with offices in Western Europe, Asia Pacific, and Latin America. We maintain a worldwide network of distribution channels, system integrators, and technology partners.
Digium® Makes it Easy for Companies to Benefit from Genuine Asterisk® with FreePBX User Interface for AsteriskNOW™
HUNTSVILLE, Ala. Digium®, creators and primary corporate sponsors of the popular Asterisk® open source telephony platform, announced today the release of a new version of the award-winning AsteriskNOW™ software appliance. Designed to significantly simplify the process of installing, operating and managing an Asterisk-based telephony system, the new release includes the popular FreePBX web-based Asterisk management interface and an array of other open source components. AsteriskNOW 1.5 allows organizations of all sizes, as well as systems integrators and other solutions providers, to benefit from the power of genuine Asterisk while enjoying the ease of use of a software appliance.
Asterisk is the world’s leading open source telephony engine and toolkit. It offers flexibility that is unheard of in the world of proprietary communications in order to empower developers and integrators to create advanced communication solutions. Licensed under the GNU General Public License (GPL), Asterisk is the most commonly used open source voice over IP (VoIP) software.
“Asterisk has always been a powerful platform for building telephony applications,” said Mark Spencer, CTO of Digium and the creator of the Asterisk project. “The first version of AsteriskNOW made it really easy to get started. Now with the second release we’re responding to our community and adding a wealth of additional features and functions.”
AsteriskNOW 1.5, like its predecessor, installs in 15 to 30 minutes and requires no in-depth knowledge of telephony or Linux to get started. By including the FreePBX administrative interface, Digium has made AsteriskNOW easier to configure and maintain. FreePBX gives administrators the ability to graphically manage phones, VoIP services and add-on applications. The release also includes a change in platform with a move to the freely available CentOS Linux distribution, which is based on an extremely popular and stable enterprise Linux distribution. The addition of FreePBX and the move to CentOS give open source telephony users a familiar, stable, community-driven platform for application development.
“Since its beginning, FreePBX has been instrumental in the adoption of Asterisk by a segment of the community who may not have otherwise taken the plunge,” stated Philippe Lindheimer, leader of the FreePBX project. “We are delighted to extend our relationship with Digium and become part of this new AsteriskNOW distribution. We look forward to broadening our already excellent relationship with the Asterisk community as we expand the market for open telephony solutions.”
The new Digium platform is targeted at technically savvy users looking for an open source telephony development platform that is both easy to use and highly configurable. It is offered as an alternative to Digium’s Switchvox IP PBX.
“AsteriskNOW provides a distinct option for organizations that want the ability to create highly customized applications using Asterisk, along with the convenience of built-in Linux and other open source software,” said Bill Miller, Digium’s vice president of product management and strategy. “We’re committed to Asterisk as an open source platform for telephony development, and AsteriskNOW is a major part of that commitment. Our goal is to promote the global adoption of open source Asterisk into those channels comfortable with a completely open solution, and to offer Switchvox to those who need a world-class SMB IP PBX and support.”
Friday, October 17, 2008
Alternate Access Now an Approved Reseller of Trixbox VoIP Solution
Alternate Access, a provider of converged communications solutions, reportedly has become an approved reseller of Fonality’s trixbox VoIP solution (pictured below), which provides small and medium-sized businesses a flexible, open source IP telephony solution.
Officials from both companies say the trixbox system offers the benefits of a premise-based solution and hosted IP phone system, storing critical phone system data securely at Fonality’s data centers. That reliability is designed to ensure that businesses will be up and running within one hour of rebooting the system or receiving new equipment if a customer’s phone system experiences an outage or is damaged.
According to Kelly Lumpkin, Alternate Access’s chief executive officer and director of business development, the company is continuously reviewing new products to offer customers a range of quality communications solutions.
“Fonality’s trixbox is ready for primetime,” Lumpkin said. “It is easy to use and exploits both the open Linux operating system and open sourceAsterisk communications engine – making the trixbox affordable and practical for customers familiar with point and click GUI interfaces.”
Now, officials with both companies say, technicians can provide real-time updates and receive system alerts about critical system components with support contracts.
That’s expected to help them fix problems before those problems make a significant impact. If a system is damaged, a replacement appliance is cross-shipped to remedy the issue. Advanced features and call privacy round out the premise-based benefits. Software upgrades are also completed automatically and during non-business hours.
The trixbox appliance uses telephony interface boards or gateways to connect to legacy PSTN trunks that provide many of the features of a premise-based system. The company says that the technology provides high audio quality and reliability.
Business VoIP: new money saver?
More than ever before, businesses are looking for new ways to cut down costs. Many are considering using voice over Internet protocol (VoIP) for at least part of their telephone needs.
The number of residential and small office/home office VoIP subscribers worldwide is expected to double by the end of the year to more than 47 million, according to Infonetics Research. Right now, only 17 percent of small and medium-sized businesses have converted to VoIP.
The primary advantage of a VoIP system is that you can make and receive phone calls using your existing high-speed Internet connection and save on long distance charges. The industry reports an average savings of 30 percent. Considering commercial phone rates, this is a considerable saving, especially to a small business owner.
VoIP has other benefits. If your business has several locations, you can establish a private VoIP network that will allow your locations to stay in almost constant contact for no cost. Also, you may be able to avoid paying for both a broadband connection and a traditional phone line. VoIP is extremely portable. Calls can be forwarded anywhere there is access to a computer – at home, on the road, or to a laptop at the beach. Wireless “hot spots” in locations such as airports, parks and cafes allow you to connect to the Internet, enabling you to use your VoIP service wirelessly.
There should be no change in quality, compared to conventional phone service. But because calls are transmitted over the Internet, there may be occasional disruptions during times of heavy Internet usage. A good gauge of this is the frequency of your congestion issues now. If congestion is relatively common, you may want to upgrade your connection or simply drop VoIP.
Before converting to VoIP, you should know that the whole picture includes a few other drawbacks. Some VoIP services don’t work during power outages and the service provider may not offer backup power. Also, not all VoIP services connect directly to emergency services through 9-1-1. You may need to assign different priorities to different types of Internet traffic. VoIP traffic, for example, should be given higher priority than e-mail traffic because voice is more time-sensitive. Finally, VoIP providers may or may not offer directory assistance or white pages listings. But for most businesses that have converted to VoIP, the view has certainly been worth the climb.
It isn’t necessary to use VoIP for your entire phone system. Some businesses have blended systems in which traditional phone service handles the bulk of the calls, while VoIP saves on long distance.
While it is possible to use a dial-up connection for VoIP calling, this connection speed is too slow for commercial use. VoIP requires a broadband (high-speed Internet) connection via cable modem, DSL or a LAN. You also need a computer, a phone adaptor or a specialized VoIP phone. Dedicated VoIP phones plug directly into your high-speed Internet connection. They run several hundred dollars now, but they retain all of the features of current phones – extension dialing, an auto attendant to answer the phone and route calls to VoIP extensions, voice mail boxes, audio conferencing -- and the prices are dropping as VoIP becomes more popular.
If you use a traditional phone with a VoIP adaptor, you will be able to dial as usual, and the service provider may also provide a dial tone. Businesses can implement VoIP between computers using nothing more than downloadable free software like Skype of Asterisk, but additional capabilities are available only through VoIP service providers.
Your computer has to be turned on when making a call only if your service provider requires it, but your broadband Internet connection must be active at all times. And yes, you can use your computer while making a call.
As your VoIP system is being installed, employees should be trained on how to use it. Some of the basic courses offered by system vendors are vital tools for non-technology workers who must master the vocabulary and a basic understanding of IP telephony and networking. As the buzzwords and jargon are clarified, employees will attain key knowledge that cannot be acquired simply by reading trade publications.
The major VoIP service providers – Verizon and Vonage in this area – offer a bewildering array of service plans. Some of these charge for long distance calls outside your calling area; others permit you to call anywhere at a flat rate for a fixed number of minutes.
Business VoIP: Cost, Service, and Features
By Michelle Robart, TMCnet Editor
The way a business handles its incoming calls says a lot about it. When a call comes in, is the caller greeted by an automated attendant with voice prompts? Is there a professional voicemail system? Can someone be reached through the main number at any time?
It is these types of features that give a business its professional edge.
Trends show an increase in service providers, and an increase in the services that are being made available has followed. In addition, VoIP is beginning to include calls to and from mobile phones, which is expected to significantly increase VoIP usage, especially as the smartphone market takes off.
So, with all the business VoIP services available today, how does a business know which service provider will best meet its needs?
Having an affordable and reliable phone service provider is crucial. Today, the majority of businesses have the luxury of having options when it comes to selecting a phone service provider. When shopping for a new service provider, all businesses must first have an understanding of what they are looking for.
Quality customer service is always important and with some help from the Internet, business owners can now do a better job of researching which service provider will best meet their needs. By switching from a “traditional” phone provider to a VoIP company, businesses should expect to receive quality phone and customer service.
Features are always important and when shopping around, make sure to find a provider that offers the features you are looking for. But at the same time it is important that these features actually work and that there is a “real” customer service department to support your business if something ever goes wrong or if you’re simply looking for help.
With Nextiva, all the essential business VoIP features, such as Automated Attendant, Voicemail, flexible call forwarding and find me/follow me, are included in all its VoIP packages.
Read on to discover some of the inherent benefits that Nextiva business VoIP solutions can provide for small businesses.
TMCnet: What are some of the main benefits from switching to a voice over Internet protocol (VoIP) system?
Nextiva: Business VoIP phone service has become quite popular in recent years and its future is bright. Thousands of businesses switch to VoIP each day for a variety of reasons. One of the most popular benefits of switching to VoIP is the cost-savings. A business can save up to 80 percent on its monthly phone bill while receiving many communication features that can strengthen any business operation.
TMCnet: How do companies save on telecom costs by switching to VoIP?
Nextiva: Saving money is just one of the benefits of switching to VoIP. Telecom costs should no longer be of great concern to any business owner, with VoIP an average small business can save up to 80 percent each month by using VoIP. Along with saving money on your monthly phone bill, businesses now have the opportunity to make calls to anywhere in the world without having to worry about an expensive bill. The Nextiva Connect360 comes with unlimited calling to anywhere in the United States and Canada for free.
TMCnet: What exactly is Nextiva Connect and what are its main features?
Nextiva: The Nextiva Connect is a Virtual PBX phone service that can turn the phone system of an average small business into a full-powered system with features that were once only available to large corporations. With the Nextiva Connect, businesses can have a toll-free number, an auto-attendant to answer and route all calls, call forwarding and much more. No business should ever miss a call and the Nextiva Connect is the perfect solution for startups and small business wanting to go to the next-level.
TMCnet: Nextiva prides itself in offering Fortune 500 phone features for fraction of the cost. How do Nextiva products cut monthly phone bills by up to 80 percent?
Nextiva: The Nextiva Connect 360 has been one of the most popular selling business VoIP solutions. For a flat rate of $29.95 per month businesses can truly receive Fortune 500 phone service for a flat monthly rate of only $29.95. The Nextiva Connect 360 includes unlimited calling to anywhere in the U.S. and Canada along with professional features like an auto-attendant, hunt groups, find me/follow me, voicemail to e-mail, and more.
With Nextiva as your service provider, every business owner can feel confident that they won’t be hit with unexpected fees and that the quality of service will always be the best on the market.
TMCnet: In what ways are Nextiva’s VoIP solutions more affordable than other VoIP services?
Nextiva: Nextiva's mission is simple and singular: to create a communications company that provides products and services that enable small businesses to succeed in a global economy. We try to differentiate ourselves from the “traditional” phone company by maintaining a customer first mentality. As a Nextivs user you should expect to be treated as if you are our only customer.
On top of providing quality service, we are always working to provide our users with the best price possible. Your phone bill should be consistent, clear and fair – at Nextiva we will do not “nickel and dime” our users and hope to build strong and long lasting relationships with all our customers.
UK crime-fighting concern over VoIP calls, social networks
The huge growth in VoIP traffic is "jeopardising" the capability of UK law enforcement to investigate all types of crime, senior officials are telling The Times of London.
As more and more calls get routed over the web, police are losing the ability to track call data. There's a big push in the UK to increase the government's capability to access data held by internet services, including social networking sites and game networks.
Presently, security and intelligence agencies can demand to see phone and email traffic from communications service providers. New services ranging from gaming to video sites and technologies such as wireless and broadband are causing a serious coping problem for the police, MI5, customs and other government agencies.
Across the UK there were 519,260 requests for communications data tracking - either CDR, text message or email, but the requests were not for the content. Police and security services say it is becoming difficult to locate communication data details, such as when a call was made and to what number, because there are now so many communications service providers in operation. Multiple user names aren't helping, either.
Police chiefs are quietly lobbing for more powers to collect phone and other communications data, but privacy groups view further extensions of state power as "Orwellian."
Thursday, October 16, 2008
Skype Meets Asterisk
What looks like Skype and works like an open-source PBX? "Skype on Asterisk," of course.
A recently announced collaboration between Skype and open-source telephony solutions provider Digium, "Skype on Asterisk" aims to pair the cost savings of Skype with the versatility and functionality of the Asterisk platform. A Digium-built software connector bridges the two.
With a target market of small and midsized businesses, the partners are touting the effort as a potential big win, with 338 million worldwide Skype users and the tens of millions of Asterisk users. About a third of Skype users are business customers, so there may indeed be a significant market here. (Pricing has yet to be determined.)
Presently available in beta, Skype for Asterisk can be used in conjunction with Asterisk 1.4.x, 1.6.x, and Asterisk Business Edition C.x—as well as the Asterisk bundles, trixbox and AsteriskNOW. Potential users can apply to participate here.
The Asterisk will deliver a range of features to Skype users, including automatic call distribution (ACD), along with the ability to ring incoming calls to individual Skype names, log incoming and outbound calls, handle voicemail, provide voice response (auto attendant) functionality, and deliver least-cost routing—typically, one supposes, through the Skype network.
Among its other functions, Skype on Asterisk has the ability to receive calls through a number of devices. An incoming call will simultaneously ring the user's desk phone and the Skype client. Prefer the audio quality on the desk phone? Don't want to turn off your music to answer the Skype line? Pick up the desk phone. The call is treated just the same. Or set the system to only ring one phone or the other, to make things even simpler.
Skype brings to the table flexibility, as for instance click-to-call functionality from Web sites, along with its promise of lower cost, especially free Skype-to-Skype calling. Outgoing Skype-to-Skype calls are recognized as such by the system and handled as free calls; likewise incoming calls from Skype users. Calls to and from non-Skype users are handled as ordinary calls.
This means, among other things, that a small business could create free connections between all its satellite offices, so long as all participants are Skype users.
By this mechanism alone, "we are going to save you money," said Danny Windham, CEO of Digium.
Certainly cost savings are a big part of the pitch here, but the partners are just as sanguine about the promise of expanded functionality that would come with an open source PBX, as noted above: Click to call, ACD and so on.
In the big picture, many in the SMB world will be tempted by the prospect of an open-source PBX if only by virtue of the added control inherent in such a solution.
Skype is to some extent a free-for-all these days, with individual users putting it to use at their own discretion, much as in the early days of cell phones in a business environment. Tying Skype to a PBX offers the possibility of bringing disjointed Skype use under the unified control of the corporate IT shop.
The ability to connect Skype to an open source PBX could have far-reaching implications, Skype VP Stefan Öberg told Enterprise VoIPplanet.com. He noted that Asterisk already forms the basis of many commercial products presently sold on the market. Thus, any product built upon Asterisk 1.4 or Asterisk 1.6 (Switchvox, Free PBX, etc.) should be able to utilize this new capability.
"Third-party vendors of products based upon the proper releases of Asterisk would still need to purchase the connector from Digium, but they could then bundle it with their application," he said.
At first blush the Digium relationship sounds something like the existing tie between Skype and VoSKY, announced last spring. In that hardware-based arrangement, VoSKY Exchange products add what are essentially Skype trunk lines to any existing PBX. Along with the Skype software running on a separate server, this brings Skype-based calling directly to end-users' desk phones.
Thanks to the Digium connector, Skype for Asterisk goes a step further, integrating Skype directly into the heart of the PBX and thus overcoming limitations such as the number of Skype operations that can take place at the same time.
The system has been designed to attract a broad SMB base. Windham estimates it would cost the typical small business user only a few thousand dollars to implement an Asterisk solution, based on the number of users and needed functionality. Add to this a few hundred dollars to set up the Skype component, and a fairly inexpensive yet flexible system emerges.Resellers frustrated with IP telephony vendor lead generation programsVendor lead generation activities may not be nearly as successful for their chan
Vendor lead generation activities may not be nearly as successful for their channel partners as expected. A new benchmark study from Nemertes Research on IP telephony found that no channel resellers are getting quality leads on a regular basis from the IP telephony vendors they work with.
The IP telephony study is part of a larger group of studies about unified communications and collaboration, specifically as they relate to vendors, product and the channel. One of the more interesting discoveries was the disparity between what vendors say about lead generation programs and what their channel partners said. According to the Nemertes study, channel partners dealing in IP telephony aren't all that impressed with lead generation activities.
"It was very frustrating for a lot of MSPs that I spoke with that they weren't getting the leads, and when they were getting them, they weren't qualified leads," said Karen Trost, research analyst at Nemertes Research.
When asked how often they get leads from the vendors they work with, 77 per cent of resellers said "rarely" and 23 per cent said "never." No resellers said they were getting leads from the vendors often.
Although she wouldn't name names, Trost said some very large IP telephony vendors have a reputation of not sending off leads except to their very largest resellers. One of the perceived reasons for a lack of leads is that resellers aren't top-tier partners, but many also believe that once they work with a vendor for awhile, the relationship gets a little too comfortable and the vendor starts to think it's not necessary to send new leads out.
For the largest accounts, vendors are also taking leads for themselves and trying to sell their solutions directly to the customer, Trost said.
Although 61.5 per cent of channel resellers only sell one IP telephony line, there is still a large percentage of resellers that sell products from multiple vendors in the space. Many of those resellers that sell more than one vendors' products believe that vendors don't pass on new leads because they're afraid the resellers will take the leads and sell competing products instead, Trost said.
There is good news, though. While hardware margins have shrunk, resellers are now finding good margins on the services element of IP telephony.
"They can make a lot more profit margin on services. Actually, the good news is with the economy right now, managed servcies is really increasing," Trost said.
Compared to the 2007 benchmark study, there has been a large increase in interest in managed services, she said. There is some interest in hosted services, but most customers are interested in managed services instead. VARs that are true MSPs are going to do well, she added.
"Virtually everyone is doing something with VoIP. Less than one per cent have no plans whatsoever to deploy VoIP," Trost said about businesses. All sizes of businesses have plans to deploy VoIP in one form or another.
Additionally, most customers prefer to buy from local regional VARs, with 54.3 per cent of businesses surveyed saying that's where they prefer to buy. Many (24.5 per cent) prefer to buy direct from vendors, while 7.4 per cent will go to a large system integrator and a further 7.4 per cent will buy from a carrier. Additionally, 6.4 per cent will buy from multiple sources.
The number of people buying direct is somewhat inconsistent with what vendors say, though, Trost said. With 24.5 per cent of businesses buying direct, it's a big difference between the five to 10 per cent of direct business vendors claim to have.
The IP telephony study is part of an eight-volume benchmark study called "Unifed Communications and Collaboration." Other parts of the study focus on organizational strategies, unified communications applications, Web 2.0 and collaboration, mobility and video over IP.
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