Even though VoIP traces its roots back to 1996, it continues to be one of the hottest and most innovative technologies. It has even proven to be a game changer in the mobile phone arena. Viber, Skype for iOS and Android (News - Alert), and other mobile VoIP apps, are very popular. TMC Labs has enjoyed discovering and testing innovative products and services within the VoIP industry for several years, and one of the things we enjoy most is recognizing innovative VoIP products via the TMC Labs Innovation Awards.
2012 marks the 13th annual INTERNET TELEPHONY TMC Labs Innovation Awards, with several strong contenders in cloud, collaboration, unified communications, high bandwidth products and, of course, VoIP.
TMC Labs uses a rigorous process when selecting innovative products. This year, TMC Labs proudly bestows 14 companies with TMC Labs Innovation Awards.
01 Communique Laboratory Inc.
I’m InTouch Meeting
I’m InTouch Meeting allows users to collaborate, host online meetings, and share their desktops with up to 15 attendees while leveraging whiteboard, chat, VoIP, and a PSTN phone bridge. The audio from VoIP and PSTN callers is inter-mixed so VoIP callers can communicate with PSTN bridge callers and vice versa. Importantly, there is no need to have any software residing on either the host’s (i.e. meeting moderator/presenter) or the attendees’ computers since it is 100 percent on-demand browser based.
I’m InTouch Meeting features a shareable (concurrent) licensing model so that multiple users in an enterprise can share the same license. This makes is cost competitive against some of its competitors. It also features file transfer to/from the moderator and attendees and powerful administration and real-time attendee access rights control. Perhaps one of the most innovative features is the remote printing of documents from the moderator to the local printer of an attendee.
In addition, it supports detailed reporting features and robust management tools. You can schedule a group meeting online, search current and previous meetings, re-start adjourned online meetings, customize your invitation e-mail, generate statistical reports, create users, and access and manage your billing account. The solution also offers real-time granting/revoking of different access rights for an attendee (e.g. the right to do file transfer, the right to receive remote printing, the right to see the presenter’s screen, the right to remote control the presenter’s computer, the right to see who else are in the meeting, the right to chat, etc.). Unlike some competing solutions, the moderator can share his or her screen and give control to an attendee. In the last six months the company has added the mixing of VoIP and PSTN real-time audio chat sessions feature, improved the speed in the screen sharing, and added the ability to re-assign the presentation role to any of the attendees.
Net-Net SIP Multimedia-Xpress (SMX)
The Acme Packet Net-Net SIP Multimedia-Xpress (SMX) is designed to handle small subscriber populations from initial service rollouts to large-scale IMS networks. It can start as low as 20,000 subscribers and can scale up to 250,000 subscribers per system. For larger deployments you can create a Net-Net SMX cluster using the Net-Net Session-aware Load Balancer to increase subscriber capacity up to two million subscribers per cluster. The solution is targeted at service providers that want to deploy a more cost-effective and less complex solution. Acme Packet explains, “IMS deployments are often burdened by complexity and cost, because of too many decomposed functions, products, and signaling interactions that slow network migration. The Acme Packet Net-Net SIP Multimedia-Xpress (SMX) combines IMS session management with leading session border control functions to reduce the complexity and cost to deliver high-value, revenue generating SIP multimedia services.”
Net-Net SMX can be used to deliver a broad range of SIP-based services including consumer or business voice (over fixed line, 3G, 4G/LTE or Wi-Fi), rich communication suite (RCS and RCS-e), video chat and hosted unified communications services. The solution minimizes up-front cost and risk while providing a fully-standards-based IMS architecture that leverages the rich features and functions of Acme Packet’s Access SBC, creating an integrated IMS service delivery solution. The company claims the solution reduces the cost of IMS infrastructure to approximately one to four dollars per subscriber.
Through standard interfaces including SNMP, SFTP, XML and SOAP, Net-Net Central also integrates with OSS/BSS ecosystems to deliver advanced service fulfillment, service assurance, billing and mediation. Acme Packet tells TMC Labs, “The SMX solution is the first product of its kind to provide IMS access, call session control, routing and interconnect functions all in a single platform, and is the only solution using a session border controller for the IMS core. With this increased simplicity, Acme Packet’s solution is the most cost-effective and business-oriented product for IMS.” Integrating the IMS core with the session border controller solves 70 to 80 percent of the IMS complexity changes and cuts service provider costs, according to Acme Packet.
In the last six months, Acme Packet has developed solutions that address over-the-top service providers, as well as wireline and wireless providers. The Net-Net SIP Multimedia-Xpress has also expanded services to support the development of voice over LTE.
Optical Networking Edge (ONE)
The ADTRAN Optical Networking Edge (ONE) portfolio enables service providers to bridge the gap between network access and transport, leverage their installed base and extend the bandwidth efficiencies of the core all the way to the optical edge. ADTRAN explains, “In the past, parallel packet or circuit-based SONET networks would be constructed to ensure each new service did not impact the quality of another – a method that is not only expensive to build and manage, but limits scalability. ONE reaches beyond the delivery of high-performance optical services at the edge of the network, enabling service providers to eliminate the need for capital-intensive overlay builds to support 4G backhaul services.” ADTRAN continues, “Through scalable service separation, ONE allows providers to optimize their network assets and cost-effectively grow their overall addressable market opportunity in both mobile and residential backhaul applications. Moreover, in combination with the ADTRAN Advanced Operational Environment (AOE), ONE assures the stringent service level agreements required for high-quality broadband and cell-site backhaul services.”
The web-based SLA monitoring tool and cell-optimized wavelength service capabilities ensure each type of traffic is guaranteed and provisioned as needed. The ADTRAN AOE provides its customers a means for efficient collection, proactive analysis and clear presentation of service and network data used to verify strict backhaul SLAs. The AOE Service Monitor tool provides intuitive, proactive reporting of SLA compliance status including customer bandwidth and ITU-T Y.1731-based performance statistics for one-way frame delay, delay variation and frame loss to sub-millisecond accuracies.
ADTRAN claims that ONE is the only solution on the market that combines legacy networks and services like SONET/SDH with advanced optical access services like gigabit Ethernet, Active Ethernet and GPON, with technologies more commonly found in core networks like DWDM, CWDM, scalable carrier Ethernet, OTN and ROADM. The advanced solutions, typical of core networks, have been edge-optimized so that they can be deployed with less capital and with only minimal training thereby improving deployment time and reducing costs. ONE is the only solution on the market that combines and accommodates packet optical transport system functions with advanced access technologies within a single device, while competing solutions require multiple boxes to terminate transport and access technologies. Further, with ONE’s modular design, instead of a new box, operators can simply add a card into the system when necessary.
Unify ME VNOC Symphony
Unify ME VNOC Symphony is a managed service offering customers the ability to perform scheduling and management of videoconferencing endpoints and infrastructure using Apple iPhone, iPad, and BlackBerry smartphones. Conference hosts, owners and administrators can directly schedule and manage their systems through the touch-screen interfaces and intuitive GUIs. Using your mobile device or PC browser you can view scheduled conferences, view schedules for individual endpoints verifying availability, create meetings with both internal and external participants. You can also select a desired videoconference layout, such as 2x2 or 4x4 even-sized squares, or .uneven layouts such as 1x7 (one large square, seven smaller squares), and many other layout choices. The layout choices enable you to have the speaker featured in a large video square and the attendees in smaller squares.
In addition, you can view usage and performance reports – i.e. system usage, number of calls per system, conference type, and success rate. The platform also features a unique geo-spatial interactive map, which lets users see how many miles would have been needed to be traveled to conduct a specific scheduled meeting in person. Unify ME VNOC Symphony is available to customers using AVI-SPL’s hosted or managed services for their videoconferencing needs. MCUs, gatekeepers, proxys, etc., can sit on premises or in the AVI-SPL VNOC.
AVI-SPL tells TMC Labs, “Many videoconferencing vendors have a management suite for their specific systems, but Unify ME VNOC Symphony is vendor agnostic. As long as the system is standards based, Unify ME VNOC Symphony can manage it. With Unify ME VNOC Symphony, scheduling and management of calendars are now in the hands of the users. No longer do people need to hope a system is working or available when they schedule a meeting. They have this knowledge at their fingertips with a few simple steps.”
The current version of VNOC Symphony 2.5 was recently released with many new customer requested interface changes to include the addition of many VNOC facing support tools like VNOC Maestro, which allows technicians to view over 75-live statistics within a video conference.
Digium practically invented the open source telephony movement when the company launched Asterisk in 1999. The organization has continued to innovate. and made a smart acquisition of Switchvox back in 2007. Five years later, Switchvox 5.5 now has extensive unified communications features, and version 5.5 now works with Digium’s new family of HD IP phones. Importantly, its plethora of features is offered at an affordable price that SMBs require.
Switchvox 5.5 is built on Asterisk and offers a comprehensive communications solution for small and mid-sized businesses. This unified communications system integrates all office communications, including phone, fax, chat and web mashups. You can access call queues, view presence, and see the applications you need right on your desk phone.
The Digium Phone Management is a key component, and it includes auto-discovery, auto-provisioning of the Digium phones as well as simple installation. It offers complete control of the IP endpoint with changes that can occur in real time, without having to reboot the phone, thus saving the customer time and eliminating service interruptions. There is also “application integration” with the Digium phones. Users can see call queue statistics, access parked calls easily, see real-time status/presence information about callers, and completely manage voicemail messages with visual voicemail.
Digium tells TMC Labs, “We’ve integrated a unique web-based Switchboard in Switchvox. It is supported on any browser and allows people access to real-time presence, drag-and-drop transfer, mobile phone numbers, call queues, call recording, monitor, whisper, barge as well as the personalized phonebooks, the internal directory and call parking. Not to mention, users can add other web mashups easily such as CRM integration and more. This is available at no extra cost to every user of Switchvox. Users have access to more than 300 API methods within Switchvox, allowing them to integrate other application with the IVR, call reports, and more.
Ensim Unify Service Provider Edition
Ensim Unify Service Provider Edition enables service providers to rapidly deliver, bundle, and differentiate hosted services at lower cost and higher ARPU. Ensim Unify provides a centralized, relational platform that manages creation, activation, configuration, and administration of an entire hosted application and service offering. It works with or without HMC and MPS components and easily overlays any existing HMC, competitor, or self-built Microsoft-based hosting environment. It supports connectors for over 40 applications and services and the ability to add custom connectors via its SDK. Service providers can easily connect to any billing system or other OSS/BSS/SDP systems via web services (XML/SOAP) and offer hosted applications from their own data center or cloud-based services from third-party providers.
Based on a modular, scalable, extensible, carrier-grade architecture, Ensim Unify is designed to address the most critical operational challenges in deploying and managing hosted applications and services. Ensim Unify service delivery platform enables service providers to centrally create, control, and deliver hosted IP and application services for improved deployment times. With the latest release, Ensim Unify Service Provider is the only control panel that provides full management and provisioning for Microsoft Lync Server 2010 and the Multitenant Pack for Partner Hosting. Ensim Unify offers four fully internationalized web management portals including service provider, reseller, organization, and end user. Lastly, it includes the ability to support a wide range of cloud desktop, computing, and application management applications from Citrix, VMware, and Microsoft.
GXP2124 Enterprise HD IP Telephone
TMC Labs has always viewed Grandstream’s products as being the best bang for the buck due to their advanced feature sets and affordable prices. We might even argue Grandstream has been a major force in driving down the costs of everything from VoIP ATAs to IP phones to IP cameras. The new GXP2124 Enterprise HD IP Telephone is no exception with a list price of $169.
The GXP2124 is the first HD telephone from Grandstream with electronic hook switch support for Plantronics headsets. Users can answer and end calls using only the button on the headset, eliminating the need to touch the desktop phone. Features includes four line keys with up to four SIP accounts, 24 speed-dial/BLF keys, expanded native language support, and five-way conference. It also has context-sensitive XML programmable keys and up to 32 call appearances. It sports a 240x120 backlit graphical LCD with up to eight level grayscale and dual 10/100mbps Ethernet ports with PoE. This truly is an international IP phone with support for multiple native languages including Chinese, English, French, German, Greek, Italian, Japanese, Korean, Portuguese, Russian, and Spanish.
Most mid-range to higher-end IP phones now include HD wideband audio, and this phone is no exception. It also features a full-duplex speakerphone with advanced acoustic echo cancellation. It is able to perform a five-way conference by leveraging the excellent audio performance of DSP Group’s XciteR chipset. One advanced feature is it supports personalized application services such as weather, stocks, currency, RSS feeds, music ring tones, music streaming, and more. The phonebook supports up to 2,000 contacts, and the call history holds up to 500 records, both of which are more than those supported by many IP phones. It also has advanced security protection (TLS/SRTP/HTTPS/802.1x) and auto provisioning (TR-069, HTTPS, and AES encrypted XML configuration file).
Knowlarity Communications Pvt. Ltd.
Knowlarity makes business telephony intelligent and accessible for SME in emerging markets by providing cloud-based telephony solutions. Cloud telephony-based products replace on-premises hardware-based telephony solutions, which require hardware setup, maintenance and up-front costs. Knowlarity’s solutions offer pay-as-you-go pricing and target both enterprise and SMEs. Knowlarity provides a suite of hosted voice applications for call tracking, call notifications, call forwarding, call automation and interactive voice response. These technologies help businesses manage, measure and automate voice communications.
Knowlarity was incorporated in August 2009 and in the short span of two years has grown from a garage startup to a 200-plus people company with revenues in the several million dollar range and with more than 40,000 customers in India.
Its product suite consists of SuperFax, SuperReceptionist (hosted PBX), PhoneAll (instant blast message to group or instant multi-party conference), SuperConference (conference bridge dial-in), and SuperCaller. SuperCaller is a hosted IVR solution linked to an inbound phone number; alternately, it can be used in outgoing calls and can support a large (1,000-plus) number of simultaneous connections for both outgoing and incoming calls.
Customers only require mobile or landlines to use their services, with no setup or hardware installation required at customer premises. This is a major factor in the company’s success in emerging markets.
The platform can generate hundreds of outbound calls and then transfer them to an IVR where it asks questions and stores the answers. Some of the features include Indian accented text to speech, real-time MIS reports, API for integration with software or database, schedule start and end time for a campaign, as well as retry time.
One application the company developed is a police helpline for Madhya Pradesh government, which allows any person from anywhere in the state to register a complaint over the phone. The system senses the location of the person and connects to the police station that has jurisdiction there. On the other side of the call is the police officer responsible for registering the complaint. The complaint gets registered with the voice signature of both the police officer and the person who is making the complaint. This solution allows audio logging of complaints and a definite proof through the voice signatures of the police officer that the complaint was actually registered. M.P. government is using the solution to address corruption and improve turn-around times in resolving complaints.
M5, ShoreTel’s Cloud Division
M5 offers a cloud-based business phone system along with several hosted applications to improve customer service and measure important business metrics. M5 Intelligence is a hosted application that includes phone-based business intelligence and analytics that can be assessed using detailed reports and customized dashboards. M5 explains, “One powerful example is the Live Answer Service Metric. Do you know how often a client or prospect calls your business, but is unable to reach a real person? With most phone systems, this key figure is difficult to uncover outside of the call center. That’s why M5 provides the Live Answer Service Metric as part of our M5 Intelligence package. Customers tell exactly how many calls reach a live person, regardless of which line is called or how many auto-attendants are used to direct the call. This sophisticated, but easy-to-use KPI lets you understand this vital statistic for your whole company or for any individual phone number.”
From the web portal, administrators can manage authorized contacts and assign phone system roles, log and track service request tickets with M5 Support, manage connectivity and continuity, run detailed call reports, edit ring groups, manage 911 addresses, see current invoices and access billing history, and more. From a user perspective, they can specify personal find me/follow me controls to utilize mobile devices and home phones, enable incoming call screen pops to see who is trying to phone and choose to answer or send the call to voicemail, and remotely check voicemail and forward messages to coworkers. One innovative feature allows users to set personal disaster recovery preferences to designate where calls should go in the event of an outage. It also supports click-to-dial dialing of contacts via the web directory.
The platform is able to track and report on each leg of a call, regardless of whether the caller called an auto-attendant, hunt group or direct inbound dial. This technology also tracks calls that were routed outside the phone system to a cell phone or PSTN legs. M5 states, “We are the first to offer this type of detailed reporting outside of the contact center. Our ability to report on calls to get a true measure service, regardless of the call path or number dialed, is the first of its kind for the enterprise.”
Fiber Driver - High Density 10G Optical Transport Solution
MRV’s Fiber Driver – High Density 10G Transport Solution enables service providers, data center collocation companies, Internet exchange carriers, web service companies, cloud and content providers, and large enterprise data centers to meet today’s need for the cost-effective optical transmission of 800gbps over fiber across metropolitan areas. The Fiber Driver has been chosen as the solution of choice by top Fortune 500 cloud providers such as Amazon Web service to connect their data centers. MRV’s Fiber Driver – High Density 10G Transport Solution claims to be the industry’s highest density and power efficient for 10G services per minimal rack space; however, ADVA, Ciena, Cisco and Cyan are considered competitors.
The solution’s primary target market includes content distribution network providers and cloud networking providers looking for high density, 40-80 channels of 10G, achieving the enormous capacity of 800gbps between their data centers. The Fiber Driver is also targeted at users with lower channel counts, including Tier 3 and small Tier 2 ISPs/CLECs; data center co-location (multi-tenant) companies; multi-building enterprises such as utility and health care dark fiber data centers; and college and university campuses.
The Fiber Driver also packs up to 80 high-speed channels of 10G transport in just 11U of rack space, resulting in up to 75 percent space savings and up to 90 percent in power savings. According to MRV, it has an excellent Telecommunications Equipment Energy Efficiency Rating (a Verizon standard) of 10.28 to 10.55 (on a scale of 1-10), while delivering eight times the bandwidth at 25 percent of the power consumption.
Panasonic SIP KX-UT670 Smart Desk Phone
Released earlier this year, the Panasonic SIP KX-UT670 Smart Desk Phone features a 7-inch color LCD touch screen and an open source-based operating system that is programmable in Java. Companies can utilize a variety of business-friendly applications or develop ones that are specifically designed to fit their needs. Applications can be easily loaded onto the phone using the built-in SD memory card slot. It also features web browsing, e-mail, and the ability to check daily calendar appointments with third-party applications. It even supports Adobe Flash web content.
Through the use of the high-quality video (H.264/720P) viewer and up to 16 integrated Panasonic Network Security Cameras, managers in retail environments can discreetly monitor a store from a back office or from home via a desk phone. Users can choose a specific camera feed to view or opt for all feeds to run sequentially on the LCD screen.
In the education vertical market, school faculty and authorities can monitor potential emergency situations in classrooms from a secure area by viewing live video on the phone from Panasonic’s PTZ network video cameras. Users can control integrated cameras’ pan, tilt and zoom functions directly from the phone’s screen. The KX-UT670 is the first SIP-based phone to integrate with 16 network cameras allowing it to truly become a security monitoring device as well as a phone.
Other vertical use cases are in the health care and hospitality fields, where the KX-UT670 phone gives users the ability to develop applications to enhance business operations and could include applications such as nurse call systems and easy ordering from a touch screen for room service.
Features include two Gigabit Ethernet (GbE) ports, Power over Ethernet (PoE), full duplex speaker phone, Bluetooth, electronic hookswitch, 6 SIP accounts, HD audio (G.722), 32 ringtones, 3-way conference call support, phonebook, long handset cords, wall mounting option and a changeable angle stand. Lastly, you can surf the web while talking on the phone, which some competing products do not support.
Radware’s Attack Mitigation System (AMS)
Radware’s Attack Mitigation System (AMS) is a security solution for cloud service provider infrastructure and data center protection. AMS is a real-time network and application attack mitigation solution that protects the application infrastructure against network and application downtime, application vulnerability exploitation, malware spread, information theft, web service attacks and web defacement. Radware’s Attack Mitigation System contains three layers, including a protections layer, security risk management, and emergency response team.
The protections layer is a set of security modules including denial-of-service protection, network
behavioral analysis, intrusion prevention system, reputation engine and web application
firewall – to safeguard a CSP’s infrastructure (billing, DNS, AAA, web portals, VI, etc.), network, and servers against known and emerging network security threats. The security risk management features built-in security event and information management collecting and analyzing events from all modules to provide enterprise-view situational awareness. Lastly, the ERT is comprised of knowledgeable and specialized security experts who provide 24x7 instantaneous services for customers facing a denial-of-service attack in order to restore network and service operational status. Radware ERT brings real-life experience in attack mitigation together with product expertise for best system tuning while under attack, as well as support and training of the CSP security team.
Radware’s AMS is the industry's first fully integrated IT security strategy and portfolio that protects the application infrastructure in real time against network and application downtime, application vulnerability exploitation, malware spread, information theft, web service attacks and web defacement. Radware’s AMS is fully suited for multi-tenancy, self-service, and automated provisioning of cloud computing environments. It supports easy provisioning of security rules per customer by orchestration and cloud management systems, and the generation of security- related reports per customer either via Radware’s SEIM or the CSP’s SEIM. This unique capability facilitates new potential revenues for IaaS providers from selling security services, such as anti-DoS protection and WAF services.
Radware explains, “As more and more service providers rush to become cloud service providers, there is a growing need for them to create service differentiation in order to attract new customers. The attraction of new customers and growth in business may bring the CSP to the attention of attackers, creating the need to protect both the CSP’s infrastructure and its hosted customers’ applications. AMS addresses concerns about ensuring cloud-hosted applications’ availability, SLA, performance and security.”
Symphony Teleca Corp.
Symphony Teleca offers a portfolio of services, platforms and partnerships that help enterprises accelerate their mobility strategies, retain control of their costs, and manage their mobile security. m-Suite, and associated Enterprise Mobility Services, target large enterprises and top-tier service operators. m-Suite is provided as a fully managed, cloud-based service. Customers need only have secure browser access and VPN connectivity. The following cloud-based deployments are used: Microsoft Azure, Amazon Web Services and private cloud. Symphony Teleca’s end-to-end solution gives enterprises a 360-degree ability to address mobility requirements relating to end users, devices, applications and networks.
Symphony Teleca has developed its own proprietary technology stack to optimize cloud deployments, thereby delivering m-Suite from a mobile-optimized cloud. Symphony Teleca was the first to offer a truly integrated mobile device management and telecom expense management platform, including mobile apps for telecom expense management that cover all major the smartphone operating systems including iOS, Android and BlackBerry.
Symphony Teleca states, “The latest release of m-Suite CLM breaks new ground, providing enterprises with the ability to define and deliver complex mobile device and telecom expense manage workflows using a simple graphical user interface. As well as accelerating deployment times, enterprises are further able to tailor their platform usage to address their specific telecom management needs, and consequently make greater savings in their telecom spend.”
In the last six months, m-Suite has been updated with Sybase’s latest Afaria MDM platform (7.0). Of note is Afaria’s new analytics and reporting functionality, including on-device data collection for voice/data and application usage. This gives enterprises the most advanced capability to define, implement and measure their mobile management policies. Also, it added direct integration of the cloud-based Mobile Enterprise Application Platform with the Enterprise Application Storefront.
Virtual PBX Complete with VoIP Anywhere
Virtual PBX Complete with VoIP Anywhere is a turnkey hosted IP PBX business-class phone system designed for small- and mid-size businesses. Unlike other hosted PBX services, Virtual PBX Complete with VoIP Anywhere supports complete blending of analog and IP telephony, while simultaneously incorporating open SIP peering. Leveraging VoIP technology allows customers to use their smartphones and computers in and out of the office. High-quality calls can be placed over 3G, 4G, and Wi-Fi connections through any standard SIP softphone app. The service includes popular features, such as business caller ID for outbound calls, call recording, call transfers, call monitoring on inbound calls, and real-time reports.
Virtual PBX claims to have invented the first true hosted PBX in 1996, and has always used its own platform, consisting of a variety of multi-user operating, billing, voice, control, and telephony applications and services. Virtual PBX espouses its phone flexibility, saying “We can use any combination of existing phones, phone switches, analog or IP phone lines, cellular phones, or any other type of phone. Alternatively, clients can purchase standard IP telephones and VoIP registration services from us, if desired, but there is never a need for capital equipment expenditures for things like switches and servers.”
Other features include ACD queuing for call center applications, call routing to a distributed workforce, find-me/follow-me call forwarding, supervised call transfers, auto-attendant greetings, and menus in a distributed environment. Other features of note include HD audio support, call preview, voicemail screening and interrupt, multi-business support, multi-stage dialing, and automatic routing based on incoming caller ID. Importantly, the product can present the business caller ID, and not the number of a personal cell phone, when making an outbound call. The product also features a greetings manager and library for customized virtual attendant greetings that allows a different custom greeting for each phone number. Administrators will enjoy the real-time monitoring of phone system activity by extension or department, including callers on hold, calls in progress, hold times, caller IDs, and more.
Edited by Stefania Viscusi