Alcatel My Teamwork Land Mobile Radio Conferencing and Collaboration solution (LMRCC)
On September 11th, 2001, one of major problems during rescue operations was that the police department radios couldnï¿½t communicate with the fire department radios. Although this problem has existed for decades, certainly 9/11 brought this issue under closer public scrutiny. As with any problem involving technology, it is the technology companies that will solve these issues. The Alcatel My Teamwork Land Mobile Radio Conferencing and Collaboration solution (LMRCC) claims to have solved these issues by utilizing third-party radio-SIP adaptors to integrate disparate radios (e.g., Police, FEMA, Fire, EMT) with mobile and standard telephones into one seamless audio and data conferencing experience.
The software can run on a PC using a standard Web browser and they also have a ï¿½thick clientï¿½ that runs on smartphones. The application interface looks much like MSN Messenger and has the ability to organize contacts into groups. Also similar to MSN Messenger is its full presence support to display whether you are on the phone, offline, away, etc. The primary target market for the solution is national, state, and local governments. Secondary markets include healthcare and educational campus scenarios; small, medium, and large businesses; and service providers.
Often it is critical for workers to be joined together within a data collaboration session to view and annotate maps, blueprints, weather radar screens, policy statements, and press releases for the public. Workers equipped with PDAs, smart phones, and/or telephones can talk with each other using the Alcatel solution, while also collaborating on important visual and written material.
The Alcatel My Teamwork Land Mobile Radio Conferencing and Collaboration solution uses one or more radio-SIP adapters produced by other companies, coupled to an IP network, such as a LAN or a secured WAN, with the Alcatel My Teamwork unified conferencing and collaboration application software. The My Teamwork LMRCC solution executes software routines on a standard industrial PC so that no special hardware involving DSP chips and boards, high-speed audio busses, etc., is required, maximizing time to market and minimizing total cost of ownership. Importantly, since the solution uses IP networks, it can connect geographically separated emergency workers, agencies, government workers, etc. Finally, the My Teamwork LMRCC solution offers a set of services-oriented application programming interfaces (APIs) that allow it to be integrated with other systems, including the ability to automatically initiate a conference if a certain alarm event is triggered.
Carriers and service providers are always striving to increase bandwidth to deal with the explosive demand by businesses and consumers. Indeed, with increased competition among carriers and service providers and the growing use of bandwidth-hogging applications, such as P2P, VoIP, and streaming video, having visibility into the network to monitor and control traffic and user behavior is becoming even more critical to maintain existing customer loyalty and attract new customers. Thatï¿½s where network management devices, such as the NetEnforcer AC-2500, come into play in order to maintain quality of service (QoS), provide service control, and ensure ROI.
The NetEnforcer AC-2500 is optimal for deployment in high-capacity and fully redundant topologies, commonly found with large DSL, cable, and wireless network operators. In addition to its ability to support throughput rates of up to 5 Gigabits/second, the high-capacity traffic management device supports up to four Gigabit Ethernet lines, giving network administrators the flexibility to integrate with many different network topologies. The AC-2500 models can be used with DMZ (demilitarized) zones or other multiple network segments and are highly suitable for fully meshed network environments.
Leveraging an enhanced network processor array architecture, Allotï¿½s AC-2500 provides full Layer 2ï¿½7 control, the AC-2500 series supports redundancy configurations, and it is the first network management device on the market to support up to 5 Gigabits, placing it among the industryï¿½s most powerful network traffic management solutions available.
Carriers and service providers can manage up to 150,000 subscribers with the AC-2500 series. Network administrators can identify hundreds of applications and protocols to shape/prioritize traffic and optimize traffic flows to maintain and maximize the performance of critical applications. The devices also help mitigate security threats by detecting traffic anomalies and isolating potentially malicious traffic without interrupting regular traffic.
The AC-2520 and AC-2540 traffic management devices also come with Allotï¿½s NetXplorer centralized network management software for reporting and network element configuration. With NetXplorer, network administrators can view traffic trends and drill down to individual devices, users, or applications for real-time troubleshooting. Data can be analyzed in real time or over periods of time for reporting, capacity planning, or usage tracking. NetXplorer also triggers alarms that can be programmed to identify potential security risks.
Atreus IP Service Provisioning Software
As service providers move to offer Triple or even Quadruple Play offerings, the complexity of managing and provisioning those various services becomes more complex, time consuming, and costly. In fact, service providers are expanding beyond traditional Triple Play offerings to offer hosted backup, hosted Exchange, hosted antivirus, hosted conferencing, and other applications. With so many diverse services to manage and provision, service providers are looking for a single platform to provide a unified provisioning and management system.
In fact, carriers and service providers have a strong desire to significantly reduce the time and cost entailed in the development, deployment, and management of advanced services, like Voice over IP and rich IMS-based offerings for businesses and consumers. Atreus IP Service Provisioning Software enables service providers to speed time-to-profit through the automated creation, delivery, and management of just about any service, including VoIP and value-added IP services, such as video.
Atreusï¿½ IP Service Provisioning Solution delivers pre-integrated, automated provisioning and configuration for a variety of devices, enabling the rapid deployment and extensive adoption of VoIP and advanced IP services. With Atreusï¿½ vendor-agnostic solution, service providers and their customers have access to a unified user interface to activate VoIP and complementary IP services.
Besides the automated provisioning engine, the product features self-service portals, allowing the end user to have control over service ordering, feature changes and updates. The benefit to service providers is quite apparent ï¿½ service providers can quickly deploy feature-rich VoIP bundles, while dramatically reducing the time, cost and complexity of adding new offerings, modifying features, and scaling customer growth.
Atreus claims that, in 2002, they were the first to help a handful of innovative carriers launch VoIP with ï¿½self-service portal functionality.ï¿½ As such, Atreus focused on streamlining the provisioning of core components which make up VoIP services ï¿½ specifically the Feature Servers, Media Servers, and CPE devices. Atreus claims that their solutionï¿½s total cost of ownership (TCO) is one third of that of a homegrown solution.
Atreus has continued to make their solution more comprehensive by widening its provisioning support in the VoIP ecosystem ï¿½ from Feature servers (BroadSoft, Sylantro, Sonus, Siemens (News - Alert), NetCentrex, Tekelec (News - Alert), et al) and Messaging servers (IP Unity, UTStarcomï¿½) to CPE (Cisco and Polycom phones) and Analog Telephone Adaptors/Session Border Controllers (Acme Packet (News - Alert) and Edgewater and the like).
Atreus also integrates with external systems, including local number portability (LNP), Inventory, E911, and directory listings, etc. Atreusï¿½ service provider customers enjoy a fully featured turnkey solution that empowers them to quickly launch innovative IP services to residential, business, and wholesale customers today, while supporting any future expansion into other advanced IP services such as video, video-on-demand, hosted backup, and more.
Avaya one-X Quick Edition
One of the most underserved markets for the VoIP industry has been the SMB market with four to 20 users. Many small-to-medium businesses donï¿½t have the IT staff necessary to install an IP PBX themselves or the budget to hire a VAR or reseller to replace their existing phone system.
Fortunately, Avayaï¿½s one-X Quick Edition delivers cost-effective intelligent communications to very small businesses and small branches of enterprises. With SIP-based P2P (peer-to-peer) technology, telephone system set up and installation is virtually ï¿½plug and play.ï¿½ Based on P2P technology Avaya purchased through their acquisition of Nimcat Networks, you can simply plug the telephones into the local-area network and the system configures itself. The phones automatically ï¿½discoverï¿½ each other and provide back-up for one another. In just minutes, all users have access to the most commonly used set of features including voicemail, conferencing, auto-attendant, and call management.
A complete working phone system in just minutes is innovative itself, but this product has a few other tricks up its sleeve. For instance, this solution is the first enterprise phone system to use SIP-based peer-to-peer technology. By eliminating the need for centralized servers, Avaya one-X Quick Edition enables very small businesses to enjoy significant savings on acquisition and installation. Individual telephones participating in the system perform functions previously performed by a central server. By redistributing the workload out to the telephone, costs are reduced and system reliability is increased with the elimination of a single point of failure.
No centralized equipment to purchase, set up, or manage reduces the total cost of ownership ï¿½ lower acquisition, installation, and ongoing costs. A standard secure Web browser enables users or system administrators to manage the phones either locally or remotely. The user interface makes it simple to navigate options and features. As your company grows, additional phones can be simply added to the network without the need for a professional installer or a truckroll.
Most impressive is the fact that the phones only cost from $500 to nearly $600. Many standalone IP phones that still require a centralized IP PBX can cost $300 or more. TMC Labs reviewed Nimcat Networks P2P technology before the Avaya acquisition, so we can vouch for its innovative feature set and dummy-proof plug and play installation.
EdenTree Technologies, Inc.
EdenTree Lab Manager
Telecom labs, R&D labs, and service provider labs are filled with switches, patch panels, and miles of network cables that are constantly being moved by co-workers, making testing equipment a management headache and a frustrating experience when a co-worker unplugs something currently under test. TMC Labs certainly knows a little bit about that. Well, EdenTree Lab Manager has designed a ï¿½lab operating systemï¿½ for managing, scheduling, and tracking connectivity of devices in labs and networks. EdenTree Lab Manager is a client/server software application that controls third-party physical layer switches to which network devices are connected, creating a software-controlled switching infrastructure that replaces manual patch panels.
EdenTree has partnered with over a dozen leading switch vendors, to ensure that the switching infrastructure may be custom configured to accommodate any number or combination of interface types, including POTS/analog, T1/E1, DS3, any rate Ethernet, any Optical, Fiber Channel, RF/Coax, and others. Lab Manager automates the topology reconfigurations, while providing an easy-to-use drag-and-drop graphical user interface for designing, storing and sharing configurations, reserving topologies in either a deterministic or an event-based (queued) schedule, searching for devices, tapping connections for monitoring/analysis, and tracking asset availability and usage.
Without the EdenTree solution, switches could theoretically be used as lab infrastructure, but they would have to be managed by tedious single switch CLIs (command line interfaces) available from the switch manufacturers, or by scripts that must be written and maintained by the user. According to EdenTree, ï¿½With EdenTree Lab Manager, users have a GUI that focuses on the lab devices the user needs, and our system transparently and intelligently connects those devices through interconnected switches that make up the lab infrastructure. No other solution exists to intelligently manage multiple physical layer switches acting as one virtual switch, let alone switches from multiple manufacturers.ï¿½
One truly innovative feature is that this solution allows users to submit a topology and associated executables to a queue. When the resources included in the topology become available, the topology and scripts will be executed automatically. This feature allows increased utilization of devices. Additional features include: APIs for integrated control of the system from test scripts, triggers that allow any script or executable to be launched in conjunction with scheduled configurations, and right-click access to directly control end devices. User permission controls allow fine-tuning of user and group access to devices, and priorities for usage. EdenTreeï¿½s solution is an available on the customerï¿½s platform of choice or an appliance, and they claim typical ROI of zero to nine months.
Diva Server SIPcontrol
Diva Server SIPcontrol is a software adaptation layer that allows Diva Server telephony boards to be used with the Vocalocity VoiceXML Voice Browser. Diva Server SIPcontrol provides a SIP-based approach for interfacing with the Vocalocity platform. It behaves as a SIP User Agent and converts the call control information of the Diva Server telephony board into SIP messages. Voice channels are converted into IP packets and streamed via the RTP protocol into the Vocalocity platform or to another SIP endpoint. Configured in this fashion, Diva Server telephony boards, in combination with Diva Server SIPcontrol, act as an IP/PSTN Gateway and provide an open and standards-based approach that is compliant with the Media Resource Control Protocol (MRCP) and SIP architecture. Itï¿½s interesting in that it can take inbound PSTN-trunk side calls and convert into SIP packets to relay to SIP applications. This is kind of the reverse of most SIP/TDM conversions ï¿½ most boards convert outbound TDM/PSTN calls to IP/SIP
for transmitting to another SIP gateway or device, such as a branch office.
The Diva Server SIPcontrol is the first SIP ï¿½wrapperï¿½ for telephony boards, essentially making it the first TDM product to use SIP messages to make and receive calls. Think of it as a SIP software driver for TDM boards. In addition to supporting SIP, the entire Diva Server product line has been developed with a fully modular design, which allows you to mix and match old, new and future technologies, while maintaining the system, the application, as code-compatible.
Most TDM boards require that application developers use proprietary APIs, which adds a new learning curve and doesnï¿½t leverage existing standards. This is not the case with Diva Server SIPcontrol. With the SIP ï¿½wrapper,ï¿½ developers can simply focus on creating their applications and not worry about the underlying architecture.
It is certainly faster development to include TDM in a SIP-based solution. Also, once the application runs with SIPcontrol, any feature of Diva Server is available to the application without any cost of re-design and implementation. Since it is SIP-based, it can more easily integrate with other SIP-based applications. For TDM connectivity it can use all the Diva Server Adapters (analog, BRI, PRI, etc.) It also features full support of any ISDN protocols, PBX integration with all major PBXs, can run on the same PC as the SIP application (e.g., Microsoft Speech Server), and scales from two to 480 channels.
Envox CT Connect
In this rapidly changing global economy where competition can come from all corners of the globe, rapid application development (RAD) is a key driver to your businessï¿½s success and to stay one step ahead of the competition. Envox CT Connect is a graphical call processing software that provides an open, standards-based method for communicating with over 30 leading traditional and IP PBX models, including those from Alcatel, Avaya, Ericsson (News - Alert), Nortel (News - Alert), Rockwell, and Siemens. According to Envox, it enables CTI capabilities such as intelligent routing and screen pops to over 1,000,000 agents worldwide. Envox CT Connect is also used by leading CRM and contact center product providers including Oracle/Siebel, Witness Systems (News - Alert), Cincom, and Virtual Hold. These companies rely on Envox CT Connect to eliminate complex PBX integration issues, shorten the product development cycles and ensure compatibility with a wide range of customer telephony environments.
TMC Labs has tested Envoxï¿½s products for several years and we found that the Envox CT Connect APIs are extremely easy to work with, which eliminates the need for low level programming. Envox CT Connect was one of the first products to provide an open, standards-based method for integrating with a wide range of traditional and IP PBX models. In fact, Envox offers Envox CT Connect Gateway for Cisco CallManager, a standards-based gateway that allows applications developed with Envox CT Connect to communicate with Cisco CallManager.
This product is one of less than a handful that provides an open, standards-based way to communicate with over 30 different traditional and IP PBXs models with full support for SIP and VoiceXML. This ensures greater interoperability between all call center products. By creating a more open, future-proof call center, enterprises are more able to invest in new enabling technologies such as speech, VoIP and VoiceXML.
Importantly, the software provides instant access to important PBX data (ANI, DNIS, call position, device availability, etc.) to reduce the time, cost and complexity of designing intelligent routing solutions and agent screen pop solutions.
Esna Technologies, Inc.
Telephony Office-LinX enterprise edition is an all-in-one unified communications platform features unified messaging, wireless connectivity, CTI call control, one number Find Me/Follow me functionality, Web access, instant messaging, speech recognition, and text-to-speech for e-mail reading.
Esnatechï¿½s Telephony Office-LinX Unified Communications Platform provides enterprises with enhanced access and control over communications featuring a suite of applications including multilingual speech-enabled auto attendant, unified messaging, text-to-speech, and secure wireless messaging support. The most recent release added new features that include an integrated fax server, speech recognition, Windows 2003 support and more. Having a single vendor with all the functionality rolled into one modular platform has obvious benefits, including better integration, as well as being more cost-effective than purchasing disparate systems and tying them all together.
One innovative feature is Live Reply, which is integrated with the Outlook program to allow the recipient to merely press the ï¿½Call Backï¿½ button on the Outlook toolbar to initiate a call. Similarly, Live Reply is integrated with Esnaï¿½s Web Client program to allow quick return phone calls. Esna also offers UC Mobile, an application for PDAs that enables live call control, instant messaging, and messaging access.
With the latest release, Esna has improved integration with CRM programs through ActiveX components. One really innovative feature is that the 7.0 UC Client Manager allows users of the system to associate any Bluetooth device with their location giving you presence management. Users will now be able to walk away from the computer, and when the UC Client Manager detects that they are out of range, it automatically changes the userï¿½s location to the pre-determined settings.
One final innovative feature of note is that the unified messaging feature will allow playback of messages to be gender specific. This means that the TTS engine will use a male or female engine depending on the gender of the sender.
Global IP Sound
GIPS Border Interface Engine (BIE)
Global IP Sound is best known for their voice engine installed in such popular VoIP softphones as Skype, Google (News - Alert) Talk, and more. Now, Global IP Sound has a new product called Border Interface Engine (BIE), which is designed to allow VoIP solutions to provide consistent connectivity between networks, and enables high-quality conversation in both directions by maintaining call integrity over the IP network. GIPS BIE utilizes GIPS patented codecs, as well as NetEQï¿½s jitter buffer and error concealment module, while the call goes through the IP network. The combination of BIE and NetEQ allows better management of jitter and delay on signals before they are transcoded and sent over the public Internet or PSTN.
Interestingly, the solution enhances the voice quality and hence the user experience on the receiving side, even when a call is terminated in a border gateway or PSTN and isnï¿½t a 100 percent IP-to-IP connection. According to GIPS, ï¿½BIE is the first transcoding/dejittering solution of its kind, at least to our knowledge. It will greatly improve the quality of calls between networks, a problem that has continued to plague VoIP communications.ï¿½
BIE works best when endpoints, like Skype and Google Talk, which employ GIPS technology connect to endpoints that are not using GIPS technology or are on another network, such as the PSTN. The GIPS-enabled end user gets the benefits of using high-quality GIPS codecs, while both sides enjoy reduced jitter and latency.
When NetEQ and GIPS codecs work together within BIE, GIPS claims that better than PSTN quality can be maintained at up to 30 percent packet loss. Itï¿½s important to note that previously this high quality could only be enjoyed by users who ï¿½bothï¿½ used GIPS endpoints, or if the receiving side had GIPS technology. Now, however, by placing BIE at the border of two networks, such as a media gateway between IP and PSTN, both users enjoy a high-quality conversation. Application developers and service providers will surely find this innovative VoIP solution quite useful to deploy high-quality VoIP services.
Grandstream Networks, Inc.
GXV3000 SIP Video Phone
Not all desktop VoIP phones are equal. Sure, most support the SIP standard ï¿½ even Cisco finally adopted SIP support in their phone endpoints, but some desktop VoIP phones give you much more. Grandstream Networksï¿½ GXV3000 SIP Video Phone gives you a large 5.6-inch TFTP color LCD (CIF or QVGA resolution), an advanced VGA resolution camera giving you the ability to have high-quality videoconferences from your desk, rather than making a special trip to the conference room, where high-end videoconferencing equipment is often installed.
The GXV3000 is a next-generation IP video telephone based on SIP standard and the latest H.264 video codec, which is currently the codec of choice for high-quality video. The GXV3000 is the first IP video phone that retails for less than $300 and the first H.264 IP video phone that supports real-time (up to 30fps) high-quality video at very modest bandwidth level (as low as 32kbps, up to 1Mbps). The phone allows nearly all viewing angles via adjustable LCD screens and cameras.
This IP video phone provides three line indicators each of which can support independent SIP accounts. It also features dual 100Mbps Ethernet ports (switched or routed with built-in NAT router), dual USB ports, RCA style audio/video output jacks to TV, and a 2.5mm headset jack. Grandstream claims that this is the first IP video phone that has advanced error protection and picture recovery algorithm against packet loss and network jitter.
The GXV3000 has a unique design that uses a single video DSP chip (from TI) to process audio, video, and all network protocol handshaking. Compared to other designs, which rely on two or three chips to handle the challenging audio/video processing, this innovative design achieves a new record in price performance benchmarks, continuing to make Grandstream one of the primary VoIP equipment manufacturers that comes to mind when TMC Labs thinks of inexpensive, high-quality VoIP products.
Interwise Connect version 7
Interwise Connect may not be as well known as WebEx or Microsoft LiveMeeting, but Interwise can match them feature-for-feature and has its own features up its sleeves that make it one of the best online conferencing solutions on the market. Interwise Connect delivers unlimited voice, Web, and video conferencing for the enterprise. Designed for the unique needs of mid- to large-sized enterprises, they are unique in offering a ï¿½fixed priceï¿½ with unlimited usage pricing model. Using Interwise Connect, companies can consolidate multiple conferencing tools with one product and give every department exactly what they need ï¿½ voice conferences, Web meetings, virtual training, Webcasts, broadcasts, and recordings. Interwise Connect is sold as a software site license for unlimited use by licensed participants. They also offer a hosted service, which is licensed on a fixed price/unlimited use basis.
The voice conferences features pre-scheduled and reservationless meetings. Multi-level security is available (e.g., create a personal conference room just for the employees). You can also seamlessly escalate from voice-only to Web meetings. TMC Labs certainly likes that you can have a full-featured phone conference without per-minute charges or overage charges that some of their competitors charge. Interwise is quite innovative in that their solution allows you to participate via traditional phone calls or using VoIP and multipoint video. Interwise uses SIP to synchronize the audio from multiple devices (PC, TDM phone, IP phone) for voice, Web, and video conferencing.
This product allows you to lead or attend virtual classes with full moderator control and participant interaction features. One innovative feature lets you record the Web meeting and then let participants play back the recording on their own schedule.
Interwise told us that they are the first ï¿½Unified Conferencing applicationï¿½ by pointing out, ï¿½All other conferencing products today offer a subset of voice, Web, and video conferencing and fill in the gaps by bundling technology from one or more partners.ï¿½ They continued, ï¿½Interwise Connect is the first application that integrates all three classes of conferencing ï¿½ voice, Web, and video ï¿½ at both the architecture and data level. For customers, the benefits include lower conferencing costs, seamless escalation from voice-only conferences to Web-enabled meetings and events, easy integration with business applications and IT infrastructures for more effective user access and IT management, higher security for all types of conferencing, and simplified recording and editing of live events.ï¿½
One final innovative feature of note is that Interwiseï¿½s blended deployment option allows both hosted and customer premise approaches to be used together and combined in a single integrated application. Customers benefit from combining the cost savings, enhanced security, and greater control of an on-premise deployment with the rapid start-up and global reach of a hosted system. For enterprise customers, the hosted service can provide automatic overflow and failover protection for the on-site deployment. Because the two models can be fully integrated they can be managed as a single application.
Lucentï¿½s Hosted IP PBX Service
The Lucent Hosted IP PBX Service, from Lucentï¿½s VoIP for Enterprise portfolio, enables service providers to offer a turnkey hosted IP PBX service for their business customers, by leveraging Lucentï¿½s hosted services infrastructure and operations, VoIP professional services for the business premises, and VoIP marketing and technical expertise.
Lucentï¿½s is the first private label, geographically redundant, carrier-grade hosted IP PBX service that scales to serve T2/3 carriers up to Tier 1 service providers. Hosted IP PBX leverages their Global Network Operations Center (GNOC) and Security Operations Center (SOC) to deliver end-to-end network management and security from the WAN into the LAN.
With all the hype about dual mode phones and the ability to use your cell phone connected to your office PBX, Lucent is one of the few vendors to have actually done it. Using Lucentï¿½s Mobile Extension, it enables a cell phone to function as a full-featured office phone on the IP PBX system. This is the first offering of its kind that uses a hosted services infrastructure, allowing carriers to leverage both wireline and wireless assets to provide new productivity enhancing services to their customers.
In addition, Lucent built this using a modular design based on standards, including IMS, which enables them to integrate technologies from multiple third-party partners to support a highly reliable, high-quality VoIP solution that integrates into the carriersï¿½ existing OSS/BSS environment. Sprint and BellSouth (News - Alert) are just two customers of this hosted IP PBX solution.
One unique aspect of this solution that Lucent points to is its strong application ecosystem. Lucent stated, ï¿½This offer gives providers a highly scalable solution to deliver not only VoIP, but provides the service provider access to an ever evolving hosted applications ecosystem to drive new value to businesses and support growth in network-based services for providers.ï¿½
Lucent also told us, ï¿½Lucent has a market leading portfolio and vision for the future of network-based services driven from our IMS architecture. We are working closely with our customers to provide the solutions and expertise that will help them achieve their business objectives in the market so they can lay a foundation with VoIP today, and be well positioned to leverage that investment to new services, and new markets going forward. Lucentï¿½s Hosted IP PBX Service can evolve to a full IMS solution over time.ï¿½ IT