AnchorPoint Business Analytics
Telecom managers who audit bills, make purchasing decisions, manage VoIP, PBXs, and wireless usage, as well as negotiate vendor contracts have a difficult task, especially when they are managing Fortune 1000 companies that can have multiple locations with multiple PBXs that can reach in the dozens or even hundreds. Equally, business unit executives who are accountable for the costs consumed by their departments and CFOs who need data and trend analysis for budgeting, forecasting and cost-saving initiatives; as well as CIOs who need to contain IT budgets and align their costs with business objectives and need real time access to monitor budgets and costs also have a difficult task.
AnchorPoint Business Analytics aims to simplify these tasks with their set of business intelligence tools designed to help companies mitigate risk and increase the efficiency and agility of their communications management initiatives. When implemented with AnchorPointï¿½s suite of Telecom Financial Management (TFM) applications, Business Analytics provides real-time communications data to end-users through its dashboard, benchmarking, and ad-hoc reporting tools. Integration with the AnchorPoint TFM application suite (including the Invoice Management, Asset Management, and Usage Management modules) is made possible through the AnchorPoint Communications Intelligence Engine. Data from each module is collected in a centralized database and processed by the AnchorPoint Communications Intelligence engine. Information is then delivered to the user via the Business Analytics reporting tools to allow companies to re-allocate or eliminate costs, improve operating efficiencies, and increase ROI from their telecom/IT investments. With Business Analytics, users can view information from a single screen in multiple methods ï¿½ charts, reports, meters, dials, etc. ï¿½ that represent data from the AnchorPoint Communications Intelligence engine in a ï¿½business readyï¿½ format.
To develop Business Analytics, AnchorPoint partnered with the well known business intelligence technology market leader, Business Objects. This partnership, unique in the industry, allows AnchorPoint customers to benefit from Business Objectsï¿½ leading reporting and analytic tools.
AnchorPointï¿½s Business Analytics and TFM solutions help companies meet five key business objectives: minimize communications expenses; maximize ROI on communications assets; improve operational efficiencies; enable internal staff to focus on core competencies; and provide a baseline of information to make decisions on new technology purchases (such as VoIP).
AnchorPoint targets companies who have annual telecom expenses starting at $2M and up. Currently, 20 percent of AnchorPointï¿½s customer base is in the Fortune 500 and 80 percent is in the Fortune 2000.
TP-260 SIP Gateway
Every once in a while, TMC Labs hears about a product or reads a product description and we say to ourselves ï¿½Huh? How the heck does that work?ï¿½ Indeed, it is a rare occasion that we are stumped when it comes to technology, but AudioCodes had us stumped when we read the description of their TP-260 SIP Gateway.
The description explained that the TP-260/SIP is a complete ï¿½plug and playï¿½ Media Gateway on a PCI board that uses the PCI bus for power supply only and that no software drivers are needed. No drivers? Whoever heard of a PCI board with no operating system drivers? It just didnï¿½t fit the norm for how any hardware board works and this certainly intrigued us. Upon further investigation we learned that the TP-260 can be installed in any PCI server, no matter what the operating system. Further, it doesnï¿½t use the PCï¿½s CPU, so it doesnï¿½t affect applications on the PC at all.
We werenï¿½t the only ones stumped by AudioCodes product. AudioCodes told us, ï¿½We found out that many VoIP SIP gateway users do not like to put an external box with a different vendorï¿½s name in their solution. They prefer to have the gateway as an embedded solution inside their box. This solution is so different from what the market is used to ï¿½ when you talk about a SIP to PRI VoIP gateway, or transcoding entity, we have discovered that users donï¿½t believe there is such a product. Once theyï¿½ve heard that such a product exists, they donï¿½t want to use an external VoIP gateway any more.ï¿½
It can be used for several VoIP applications. For example, it can be used as a PRI to VoIP gateway with up to eight E1/T1s per single slot board. Secondly, it can be used as a transcoding entity, from almost any LBR coder to other LBR coder, controlled by SIP. The list of codecs supported include G.711, G.726, G.727, G.723.1, G.729A/B, MS-GSM, GSM-FR, GSM-EFR, AMR, EVRC. It can also be used to develop an IP PBX, IP IVR, and next generation switches.
According to AudioCodes, ï¿½The TP-260/SIP board is the first (and as far as we know ï¿½ the only) stand-alone Media Gateway on a PCI board. All other gateways are either an external box, which is less suitable for the market we are addressing, or itï¿½s a board controlled by API running on the PC host (with an external SIP stack, and that means lots of development by the end user).ï¿½
For configuration, the user can configure the board in three ways (via the IP, from a remote PC, or from the local one), SNMP, and Web (the board has an internal Web server that the PC can connect to with a Web browser). In addition to having a Web server onboard, it also is a fully functional media server with voice prompts that can be stored on the boardï¿½s memory (up to 10MB of compressed voice). The voice prompts can also be stored on a remote server, like an HTTP server. The board can connect to this HTTP server, take a prompt and play it to the IP or the PSTN side. (If required, the board transcodes the prompt.)
The TP-260 supports SNMP, G.168-compliant echo cancellation, T.38 Real-Time Fax over IP, as well as a wide selection of in-band and out-of-band tone detection and generation. The TP-260 has a wide selection of TDM interfaces for easy integration with other third-party CTI boards. The TP-260 complies with standard network control protocols including MGCP, MEGACO (H.248) as well as AudioCodesï¿½ proprietary TPNCP. In summary, users ï¿½ especially developers and product designers ï¿½ can take AudioCodesï¿½ innovative board, install it into their PC box, and with no extra development needed users can sell their product as one complete solution in a single box.
SnowShore IP Media Server
Brooktroutï¿½s SnowShore IP Media Server is an open, carrier-class IP media server designed to deliver advanced media processing for SIP applications, which include basic IP messaging and prepaid services to conferencing and video messaging.
According to Brooktrout, the SnowShore IP Media Server is the first ï¿½pure SIPï¿½ media server, which was written by Eric Burger, SnowShoreï¿½s CTO. Its patented Web content integration engine enables the SnowShore IP media server to provide high-performance RTP to HTTP conversion ï¿½ giving it a unique capability to support high scale VoiceXML-based messaging services in the carrier environment. Brooktrout SnowShore IP media server is the first media server deployed to power 3G wireless video messaging with its support of the H.263 video compression codec.
Brooktrout pioneered the use of SIP and VoiceXML for control of its SnowShore IP Media Server. Brooktroutï¿½s suite of media server products are open, carrier-class IP media servers designed to deliver advanced media processing for communications applications, from basic messaging and prepaid services to conferencing and video mail. With standard network and programming interfaces such as SIP, VoiceXML, and MSCML, Brooktroutï¿½s solutions provide interoperability with a broad range of third-party Application Servers in next-generation wireline, wireless, and broadband networks. As proof of its open architecture and ability to interoperate, Brooktrout demonstrates an IBM eServer BladeCenter concurrently running BayPacketsï¿½ Prepaid application, Ubiquityï¿½s Ringback Tone application, and the SnowShore IP Media Server.
Another unique feature of this solution is their Web content integration (RTP-HTTP conversion) that enables the high-performance scaling of VoiceXML messaging services. Their software architecture runs on IBM BladeCenter and Advanced Telecom Computing Architecture (AdvancedTCA) platforms ï¿½ giving carriers an open, robust, and scalable SIP-based media server. In fact it delivers up to 500 sessions per blade and up to 7,000 sessions in a 7U chassis and is also available in a NEBS-compliant version.
It also supports an optional TDM interface, which extends IP media services to PSTN networks. The SnowShore IP Media Server allows customers to bridge TDM-originated calls, with reliable SIP and RTP interoperability between the media server and media gateway. Customers now have the flexibility to deploy flexible IP-based media processing resources that also connect to TDM environments.
CallWave Mobile Call Screening/Mobile Call Transfer
CallWaveï¿½s history in the convergence of voice and the Internet goes back to 1999 when they launched their first product, the Internet Answering Machine, which currently has more than 800,000 paying subscribers today. Now CallWave has a new convergence product that leverages VoIP, call screening, and mobility via CallWave Mobile Call Screening/Mobile Call Transfer. Essentially, CallWave provides subscribers with a new, local phone number that rings on their existing cell phone. When a caller uses that new CallWave number, the subscriber will be able to screen calls and even transfer them to a regular landline phone, if they so desire.
With Mobile Call Screening, a CallWave user can perform call screening similar to that of a home answering machine. As a caller is leaving a message you can listen to it in real time and then press ï¿½1ï¿½ on your cell phone to take the call, or simply let the message go to voicemail. Mobile Call Transfer allows the user to seamlessly transfer a live call from their mobile phone to a designated landline phone by simply pressing ï¿½2ï¿½ on the handset. This feature helps stretch mobile minutes and is also convenient when the user arrives at work and wants to switch to the office line, or when they get home and the cell phone reception is poor.
Unlike customers of other VoIP companies, CallWave subscribers donï¿½t need to buy a Pocket PC or lug around a laptop to gain mobility. CallWave takes advantage of VoIP technology that is unique from any other VoIP provider. CallWave is using VoIP to separate applications from transport to provide value-add services to customers and allowing them to take advantage of these new services on the communications devices they already own. CallWave has a proprietary softswitch and they claim to be the only company thatï¿½s currently using softswitch technology to offer these mobile phone features to the mass market. According to CallWave, ï¿½The technology behind allowing voice calls to be moved between landline, mobile and Internet actually requires extensive relationships with technology partners like Pac West and Level 3.ï¿½
These applications are sold on a subscription basis and you can be live in seconds once youï¿½ve ordered the service. While this product utilizes the traditional PSTN as the last mile to the consumer and not VoIP, CallWave responded, ï¿½As the cost of phone service continues to plummet, the VoIP transport layer is becoming irrelevant as a differentiator. Consumers are also becoming more sophisticated, leading them to demand more from their mobile and Internet-connected devices. The real market opportunity lies in the development of value-add applications such as the ones CallWave currently offers.ï¿½
COVAD VoIP with Voice-Optimized Access (VOA)
COVAD VoIP with Voice-Optimized Access (VOA) targets businesses with 10ï¿½200 employees per site. Covad VoIP with Voice Optimized Access (VOA) establishes two permanent virtual circuits (PVCs)-one for voice, one for data. This enables the network to allocate bandwidth dynamically and prioritize voice traffic over data traffic at the ATM layer (Layer 2) rather than the IP layer (Layer 3), eliminating the need for packet fragmentation, resulting in reduced jitter, improved bandwidth utilization, improved reliability, and superior voice quality ï¿½ an absolute necessity in the business world.
Covad claims to be the first and only access provider in the industry to offer this type of prioritized service. It is available now on Covad T1 and symmetric DSL (SDSL) lines to offer customer an array of access mediums. Covadï¿½s VOA is available in 384 kbps, 768kbps, 1.1 Mbps and 1.5 Mbps SDSL, and full 1.5 Mbps T1.
The company leverages its nationwide broadband network to provide a managed VoIP service through their VOA technology. Covad told TMC Labs that they are the first company to offer nationwide business-class VoIP with managed quality to ensure that businesses get high quality, reliable communication services. According to them, ï¿½Before Covad VoIP with VOA, VoIP users lacked the ability to prioritize and segregate voice traffic from data on a single broadband line while automatically being able to dynamically adjust bandwidth between voice and data so that VoIP phone users never experiences jitter, echoes, etc.ï¿½ Covad added that, ï¿½Call quality is one of the top concerns keeping businesses from adopting VoIP, with this major issue resolved more businesses can choose Covad VoIP with peace-of-mind that they will be getting a true telephone company replacement product.ï¿½
Covad VoIP with VOA is a unique VoIP service because it can overcome the barrier of last mile QoS by not only segregating and prioritizing voice traffic from data but also dynamically adjusting bandwidth according to real-time usage. Covad claims to be the only provider that can do this nationwide. The product also features Web administration and the ability to call contact with a click of a mouse. TMC Labs was quite impressed with the innovative features of Covad VoIP with VOA.
Contact centers are continually looking for alternatives to help preserve capital investment and outsource only telecom and data infrastructure. IP Hosted Contact Center solution providers are one arena where call centers can outsource their telecom and datacom infrastructure to reduce expenditures. EagleACD is a product of EagleIP, headquartered in New York City ï¿½ that has been designed to eliminate high entry-cost barriers that organizations are faced with when deploying new products and services.
EagleACDï¿½s unique approach allows organizations of any size to offer voice ACD with skills-based routing, Web chat, and e-mail-ACD over either PSTN or VoIP ï¿½ at no entry cost. In fact EagleACD makes it very easy to try their service at virtually no risk. Most of their services can be deployed within a matter of days, and without any up-front capital costs.
Early on, EagleACD realized the ï¿½utility-basedï¿½ model for contact centers would be successful. The basic idea is to move computing/networking toward a ï¿½utility pay-for-useï¿½ model in which users can tap into vast pools of computing/networking power and use only what they need, only paying for the bandwidth processing/network connectivity and applications that are actually used. Eagle developed the industryï¿½s first truly innovative ï¿½pay-as-you-goï¿½ pricing scheme. According to EagleACD, ï¿½The idea that you pay for what you use is a big change from the call center norm. Moving from a fixed-cost, asset-based payment plan to a variable non-asset based payment for minutes used is a huge marketing innovation in the industry. We have worked very hard to create a very simple relationship with our clients. You only pay for minutes used. There is not a lot of extra billing for different items, no fixed monthly expenses per agent seat, no minimum. That is a big innovation to provide true-metered services for call centers.ï¿½
Another truly unique innovation is what while there are a number of hosted contact service providers, all of them require minimum monthly payments. However, EagleACD does not. There are ï¿½no expensesï¿½ if there is ï¿½no business.ï¿½ Essentially, expenses are closely tied to the revenue stream. Under their business model, customers do not have any fixed monthly expenses for contact center agent seats. This is a strict interactive usage based business model where customers only pay for what they need. EagleACD told us, ï¿½This is widely used concept amongst utility companies. We are the world innovator for implementing utility concept in the hosted contact center industry.ï¿½
Rather than lump all the features under one pricing scheme, EagleACD allows for a la carte selection of features. EagleACD has developed what they call the EagleACD Utility Grid to serve the contact center market, which currently has these prices:
ï¿½ Online session ï¿½ Web chat - $0.03/min
ï¿½ Live agents ï¿½ voice; $0.06/min
ï¿½ Outbound call ï¿½ Predictive; $0.08/min
One of EagleACDï¿½s references stated, ï¿½EagleACDï¿½s skill-based routing finds the right agent among all the agents connected to the network. This allows us to optimize staffing and increase service quality for better financial results ï¿½ creating a more efficient company and more productive call center. No overstaffing is required due to this high reliability.ï¿½
EagleACDï¿½s unique pricing model makes it a truly innovative solution. Add to the fact that it has a plethora of features including VoIP, Web chat, skills-based routing, multi-media ACD, IVR, predictive dialing, e-mail response management, call recording, CTI-like integration, and more, and TMC Labs can unequivocally state that contact centers looking for a hosted contact service provider should seriously consider EagleACD.
The EdgeView NMS allows network operations and field technicians to better troubleshoot problems that impact the quality of VoIP calls for business customers. Along with detailed call quality statistics including MOS, jitter, latency, and other measurements, EdgeView provides advanced diagnostics linked to an online knowledgebase that network operators use to obtain troubleshooting tips. This capability dramatically reduces the effort and time required to identify the root cause of poor quality calls. The EdgeView product is primarily targeted at service providers, while it provides remote monitoring, testing, and repair for SMB managed VoIP and video applications.
In addition to facilitating problem resolution, EdgeView provides visibility into the overall call quality performance of the network. Trend analysis and proactive notification of poor VoIP call performance enable the operator to identify and resolve issues that would otherwise impact VoIP service delivery. EdgeView also provides several administrative features that enable the scalability of the VoIP service including centralized management, node database backup, and restore and group upgrades. According to Edgewater Networks, ï¿½To the best of our knowledge, the EdgeView was the first to couple these measurements with an online database that explained them and provided troubleshooting guidance.ï¿½
EdgeView takes common passive call quality monitoring measurements and presents them in a unique way that helps network operators save money, improve service, and increase customer satisfaction. EdgeView has several other features including identifying call quality issues with alarms, active call count reporting, and configuration backup and restore.
A customer reference stated, ï¿½The EdgeView enables our NOC technicians to become familiar with the customer experience without a truck roll. The EdgeView not only allows us to troubleshoot, but also provides us with the ability to rectify problems remotely. This saves us and our customers time and money.ï¿½
Having good tools to test VoIP and ensuring the best voice quality is obviously a no-brainer. TMC Labs was impressed with the innovative trend analysis and database comparison to put the VoIP testing results in perspective. What they have done is make VoIP testing and analysis virtually dummy-proof. TMC Labs commends Edgewater Networks for simplifying the rather complex task of VoIP testing.
Hammer VoIP Test Solution for Enterprises
Empirix has been a leader in telecom testing for quite some time, so when they added datacom/VoIP testing to their testing arsenal, there was little doubt Empirix would have an innovative winner on its hands. Empirix is uniquely situated within the VoIP testing arena because it already has TDM testing expertise which they leveraged when developing their VoIP testing products. The Hammer VoIP Test Solution for Enterprises reduces risk and speeds the rollout of VoIP services and IP telephony applications (such as messaging, speech self-service, conferencing, and CTI) by accurately assessing how their infrastructure and applications will perform live, in production. It involves driving a high load of virtual callers, performing real transactions, into a VoIP environment and then measuring how the network and applications perform. Problems are flagged, and users can drill down to determine the likely source and fix them before the system goes into production. The solution is unique because it can simulate a
real-world mix of callers, across both IP and TDM lines.
The solution is made up of three components:
ï¿½ Hammer FXT, a feature test platform spanning IP and TDM that generates synthetic VoIP calls (signaling and media) and evaluates voice quality in real time;
ï¿½ Hammer CallMaster, a graphical scripting and reporting tool for creating test call flows; and
ï¿½ Hammer Call Analyzer, a diagnostics and troubleshooting solution that enables users to visualize and debug signaling and voice quality problems in VoIP and TDM networks.
Hammer VoIP Test Solution for Enterprises provides a comprehensive approach to testing enterprise communications environments that covers both networks and applications, across IP and TDM. According to Empirix, ï¿½This is the first solution to apply Empirixï¿½ patented Hammer technology to enterprise communications networks (versus carrier and service-provider networks).ï¿½
This product is uniquely suited to look at both voice quality (both packet and call level measurements) as well as testing the correctness and performance of applications such as IVR, routing, voicemail, etc. Empirix uses speech recognition technology to assess the correctness of these applications, voice quality measurement technology to connect the network characteristics with the performance of the applications, and a breadth of protocol support (SIP, H.323, MCGP, etc., as well as all major TDM signaling protocols) to cover a breadth of equipment types and interoperability scenarios.
Backhaul Services for Wireless Carriers
Optimizing and increasing backhaul capacity and reliability has become a critical element especially as voice, video, and data application usage continues to climb. FiberTower is one of only a handful of companies to address backhaul requirements, particularly those associated with growing 3G capacity demands. While backhaul is the most critical component of the wireless network for providing sufficient capacity and scalability, it is also the portion of the network that has historically been most limiting and most expensive to wireless carriers. Until now, the backhaul space has been completely dominated by the ILECs, creating a monopoly market that offers no alternative for wireless carriers. Considering that backhaul is one of the fastest growing cost of service (COS) items and accounts for over 60 percent of cell site outages, lack of a viable backhaul alternative has rendered wireless carriers helpless to improve a major weakness in the reliability and performance standards of their networks.
With FiberTower wireless carriers now have a choice in their backhaul service and an opportunity to significantly improve customer service, network management, and cost effectiveness. FiberTower enables carriers to scale to meet next generation capacity demands ï¿½ commonly referred to as 3G ï¿½ and improve their current service standards. With over 170 million Americans using cell phones today and that number expected to increase substantially with the arrival of multimedia convergence devices, performance levels, and cell site outages have become of paramount concern to carriers.
FiberTowerï¿½s backhaul solution was built expressly to scale to meet the growing demand for wireless voice, data, and video applications. By inventing a new way to conduct backhaul traffic and eliminating the legacy copper T1 land lines that carriers traditionally employ, FiberTower not only improves todayï¿½s standards of service reliability, but also positions carriers to handle future capacity demands.
FiberTowerï¿½s solution employs a model currently used in Asia and Europe, which utilizes a hybrid solution of fiber and digital radio links (DRLs) to provide optimal backhaul service. Until recently, backhaul in the U.S. has been dominated by aged legacy copper links, which suffer from limited capacity, and are also responsible for frequent network outages, sub-optimal repair cycles and an inability to scale quickly to meet escalating bandwidth demands. FiberTower leverages a combination of fiber and digital radio links to alleviate the risk of outages resulting from land obstructions.
When we asked for any other unique aspects of their solution, FiberTower responded ï¿½We have managed to extract entirely new levels of performance and reliability out of microwave digital radio links by marrying the proven capacity benefits of microwave technology with the enhanced reliability inherent in a fiber infrastructure. Existing legacy copper backhaul suffers from shortcomings in both capacity and reliability, and FiberTowerï¿½s hybrid solution takes the best of both worlds ï¿½ microwave and fiber ï¿½ to address these shortcomings.ï¿½ Proven microwave technology replaces the aged legacy local copper loop, dramatically enhancing capacity by offering up to 155 Mbps (megabits) of storage versus the 1ï¿½3 Mbps provided by a copper loop.
As high-speed data applications, including VoIP, increase in popularity wireless carriers are under increasing pressure to scale and deliver bandwidth in a reliable, cost effective way. Through its facilities based approach, FiberTower claimed that it reduces backhaul costs by 10ï¿½30 percent based on current tariffs. We commend this unique and innovative hybrid solution for helping to solve the bandwidth problem while simultaneously reducing costs.
IP Contact Center (IPCC)
IP Contact Center (IPCC) is a SIP-based contact center solution with an advanced skills-based and data directed routing engine with text to speech, automated speech recognition and common data speaker fully integrated. Because the solution is server-based, there is no need for specialized or proprietary hardware. IPCC integrates seamlessly into existing infrastructure. IPCC is the first IP-based contact center to provide telephony integration with FrontRangeï¿½s core CRM/service management products, HEAT and GoldMine. It is a cost-effective and seamless way for these users to integrate queuing and routing solutions (among the other product features) without transitioning to and paying for proprietary architecture.
When asked about some unique aspects of IPCC, FrontRange responded, ï¿½Call recording is a good instance of a technology that weï¿½ve replaced. Because we are software in a SIP-based solution, we can offer call recording without any additional hardware or costs.ï¿½ FrontRange also explained that on the CRM side, with GoldMine, they have an integrated SIP soft phone that will speak with multiple vendorsï¿½ proxy providing us interoperability with service providers and customer premise equipment solutions. ï¿½With the tight integration of our VoIP (SIP) solution to our Service Desk and CRM applications, we are creating the VoIP ï¿½Killer Applicationsï¿½ through advanced unified communication and application integration that customers will look to as the benchmark for return on investment analysis.ï¿½
IPCC features some powerful self-service applications including IVR. Within the IVR, FrontRange Solutionsï¿½ IPPC has the ability to integrate text to speech which provides a human-sounding voice for names, addresses, and product descriptions. Self-service is also enhanced by automatic speech recognition and ï¿½common data speakerï¿½ where dates, times, money amounts are all built-in and delivered on a multilingual vocabulary-based basis.
Another unique feature comes from the integration between IPCC and GoldMine for customer relationship management. IPCC has the ability to blend inbound and outbound calls. Outbound campaigns are automatically put ï¿½on holdï¿½ when a matching inbound call takes place. TMC Labs was impressed with the tight integration of VoIP with one of the most popular CRM applications on the market today, as well as the advanced feature-set including TTS and speech recognition ï¿½ making IPCC truly a unique and innovative solution.
Global IP Sound
Global IP Sound is renowned for its excellent Voice over IP codecs. Several years ago, before Global IP Sound was well known, TMC Labs tested their voice codec on a Pocket PC and used Shunraï¿½s network emulator to induce jitter and latency into the VoIP call. Long story short, the Global IP Sound codec handled the extra latency and jitter with no problem. We were very impressed then, and today Global IP Sound continues to impress. Many VoIP softphones embed Global IP Sound technology including Skype and Teleo. Now Global IP Sound has decided to go after the mobile handset market, specifically the ï¿½coolï¿½ dual mode WiFi/3G handsets which are coming to market.
VoiceEngine Mobile is embedded voice processing software that enables application developers, service providers, and OEMs to create mobile products that are both easy to use and that provide substantially better voice quality than previous generation products. The solution is optimized for devices that run on the Symbian operating system, such as smartphones or dual-mode cellular phones, as well as for Windows-powered personal digital assistants (PDAs).
They have very advanced speech processing algorithms that provide the high-quality voice and this is what sets Global IP Sound apart. Global IP Sound was quoted as saying, ï¿½It is the first product to bring such a high level of quality (better than PSTN and current cellular telephony) to mobile communications, and is on the cutting edge by combining VoIP with current mobile technology.ï¿½ Further, they also claimed, ï¿½VoiceEngine Mobile not only provides, hands down, the best possible voice quality available on the market, but also enables applications developers to focus on development without having to worry about integration. The result is a superior product that gets to market more quickly.ï¿½ After witnessing the superb voice quality for ourselves firsthand in their earliest product years ago as well as embedded in Skype, we certainly do not dispute their claim.
The packaged solution handles all of the necessary voice components for VoIP to achieve superior voice quality, even under adverse network conditions. VoiceEngine Mobile manages all of the most challenging problems encountered in mobile IP applications, including packet loss, delay, and jitter. Using the solution with its comprehensive and easy to use API, vendors can quickly develop products that give the end user greater mobility and that incorporate excellent voice quality without in-house expertise.
Hosted VoIP Global Platform
The Go2Call Global Platform enables customers to become VoIP service providers ï¿½ that is if you want to be the next Vonage, you can use Go2Callï¿½s hosted solution. As a fully hosted, turnkey solution, the Go2Call Global Platform incorporates all of the necessary elements to create, manage, and deploy feature-rich VoIP service offerings. The Go2Call Global Platform enables customers to target solutions for the residential, enterprise, mobile, and call shop end user.
Go2Callï¿½s comprehensive hosted platform includes global call termination and origination, an OSS providing Web-based business management tools and pre- and post-paid, multi-tiered billing flexibility in real time. Go2Call claims that their platform is the first to place all of these elements in one hosted package that can be completely customized-from the branding to the service options by a customer.
One of the hottest new trends in software applications is allowing customers to customize the look and feel of the application through the use of ï¿½skins.ï¿½ The Go2Call Global Platform has recently incorporated a customized SIP Dialer V9 into its offerings that leverages SIP and HTML, and it provides fast and reliable PC calling. It works in a wide variety of Internet environments and custom skins permit Go2Call customers to modify its appearance and branding.
Go2Callï¿½s Global Platform includes several unique features. The real-time billing and provisioning systems support multi-level distribution, allowing service providers to effectively manage and sell VoIP services through their distribution networks. Go2Callï¿½s platform supports provisioning and billing for any SIP-based voice service, including PC-to Phone, Device-to-Phone, or traditional calling card services. Go2Callï¿½s advanced yet easy-to-use Web-based tools include powerful features for managing all aspects of deploying a VoIP service, including DID provisioning, rate management, and account management systems.
While offering the customer the benefits of a hosted solution, the Go2Call Global Platform simultaneously empowers the customer with the ability to customize and control the business at multiple levels. Go2Call integrates white- or private- label branding into all layers of a business. With the Go2Call Global Platform, the customer has capacity to customize everything from end user tools, such as the SIP Dialer V9 and IVR, to distributor materials, such as the multi-tiered administration tool and marketing documents. IT
TMC Labs Innovation Award winners will be highlighted in Part II.