This article originally appeared in the August issue of INTERNET TELEPHONY
TMC (News - Alert) Labs has enjoyed discovering new and truly unique and innovative products and services within the VoIP industry for several years and awarding them a TMC Labs Innovation Award. Our 12th annual TMC Labs Innovation Awards was certainly no exception. TMC Labs has been testing, examining, and reviewing products since 1994, and one of the best parts of the job is seeing unique and innovative products for the first time.
This year marked several strong contenders in these specific areas: testing tools, video, and unified communications. TMC Labs uses a rigorous selection process when selecting innovative products. This year, TMC Labs proudly bestows 19 companies with TMC Labs Innovation Awards, which are published in two parts to accommodate our in-depth write ups about the winners. The complete winners list is published in both issues; however, the detailed write ups are presented in two pieces, beginning with 01 Commnique and ending with Jabra last month, and starting with Lyrix Inc. and ending with Vocalcom this month.
Mobiso Cloud Based Speech Assistant
Speech-recognition IVRs are popular with financial institutions and large enterprises that can afford to maintain racks of servers, update speech recognition engines, and all the other maintenance involved with a speech-enabled IVR. Some small to mid-sized organizations have dipped their toes in these waters, but it is sometimes just too cost prohibitive to maintain. But what if you could outsource the speech-recognition IVR to the cloud? This would mitigate any upfront hardware costs and result in overall lower TCO. Well, that’s what Lyric’s Mobiso Cloud Based Speech Assistant aims to do. The Mobiso Speech Assistant is a speech-enabled auto attendant that utilizes Lyrix’s patent-winning PeopleFind technology, and the power of SIP. It claims to be the first fully hosted cloud-based speech-enabled auto attendant solution in the marketplace.
Lyrix explains, “Lyrix is a cloud-based service provider with over 10 years of [experience with] public and private clouds. We work with network service providers, PBX manufacturers such as Cisco and Mitel (News - Alert), as well as hosted VoIP providers to establish the SIP connectivity and the exchange of directory records between the customer and Mobiso Speech Assistant. The proven, highly accurate Speech Assistant lets customers reach employees quickly and effortlessly without the frustration of traditional dial by name lookup.”
Mobiso Speech Assistant uses best-in-class ASR technology from Nuance. It’s very easy to VoIP customers to get started with this solution – in 30 minutes either as a partner or end customer via SIP registration or SIP trunking. Once a customer is configured, Mobiso creates a cloud tenancy for the customer and directory adds, moves, and changes flow to the cloud, keeping the customer’s speech directory up-to-date. Once enabled, a user dials the speech assistant through the IP PBX and is passed to the Mobiso cloud over SIP, where Mobiso converses with the caller to determine their destination. The call is brief, and the user is transferred to the extension or phone number stored within Mobiso. The user may speak people, places, product names, customers…whatever names the customer feels is useful. External callers, trying to reach customer’s users, can be serviced by Mobiso the same way, with a company greeting welcoming the caller and then routing the caller to the correct destination; in this way, Mobiso serves as a customer service application in addition to a speech dialer for the customer.
Metaswitch SIP Session Router (SSR)
The Metaswitch SIP Session Router (SSR) provides centralized routing, SIP normalization and load balancing and session management for SIP-based voice, video, instant messaging and multimedia traffic within and between the mobile, fixed line and transit networks of service providers. The SSR addresses scaling problems when session routing decisions become much more complex, requiring a dynamic, real-time routing decision for each individual session for multiple sources and destinations within a network. These sources and destinations are SIP signaling elements such as session border controllers, wireless mobile switching centers, IMS call session control systems, and Class 4 and 5 softswitches. Scalability of the SIP Session Router is achieved by applying N+1 proxy blade scaling coupled with their own SIP load balancing servers resulting in an architecture achieving greater than 500,000 concurrent SIP sessions per instance of the SSR using COTS hardware.
Metaswitch has applied its innovative, extensible protocol interworking technology to a large scale, core network SIP proxy network element with three main objectives: centralized SIP signaling routing, SIP normalization and SIP load balancing. Metaswitch’s innovation for the SIP Session Router is providing the industry’s first XML toolkit, which eases interoperability by enabling the modification of incoming and outgoing SIP messages to accommodate variants. The toolkit is flexible enough to add, replace and modify SIP headers, parameters, and content bodies from one call leg to the other as needed. The key benefits to the service provider are flexibility to utilize the SSR to normalize SIP traffic across multiple network domains and reduced interoperability testing complexity and costs associated with expanding or adding new network nodes. Metaswitch scripting allows free-form header manipulation ensuring future proofing of the solution.
Metaswitch explains, “The Metaswitch SSR XML scripting toolkit advances SIP protocol manipulation and takes it to a new level by leveraging XML technology in combination with the Metaswitch field-hardened Ignite protocol interworking framework/OS, shared with the Metaswitch Service Broker platform. In combination, our XML-based scripting toolkit allows an unprecedented level of protocol manipulation to enable any-to-any SIP variant interworking as opposed to today’s limited hard-coded solutions. The ultimate benefit of this technology advancement is to ensure SIP technology from one manufacturer will work another manufacturer. As LTE and IMS continue to gain momentum, assuring SIP interworking is essential for best-of-breed networks.”
SpectraLink 8400 Series Wi-Fi Handsets
The SpectraLink 8400 Series phone handsets are the fourth generation of the SpectraLink 8000 product line. The SpectraLink 8400 Series could be called the Swiss Army Knife of VoWLAN handsets since it is the only VoWLAN handset that includes 802.11n, HD voice, web browser, and even an integrated barcode scanner. The product’s other key features include an XML API for application support, open SIP platform, industrial-grade durability, advanced noise cancellation, HD voice docking station, and instant messaging and presence with Microsoft (News - Alert) UC (OCS/Lync) platforms. The SpectraLink 8400 Series handsets target vertical markets including health care, hospitality, manufacturing and retail.
These phones are the first to use standard smartphone browser technology (WebKit) with the appropriate enterprise-grade security, quality of service, and management. They also have integrated push to talk for instant group communications, a popular feature in many verticals. The SpectraLink 8400 Series handsets are the first Wi-Fi phones in the industry to support dual microphone noise reduction technology. The combination of dual microphones and the Polycom proprietary spectral processing technique reduce stationary noise (that with a constant background such as HVAC, hum from machinery, etc.) and non-stationary ambient noise (that with rapid or random change, such as a person talking, background music, traffic, or typing). This allows for excellent audio quality even in extremely noisy conditions such as data centers with cooling fans, production floors in the factory, or shipping areas.
Polycom developed the QBC application to provide simple and flexible integration of the integrated barcode scanner in the SpectraLink 8450. The primary use for a barcode scanner is to emulate a keyboard as an input device for the purposes of automating data input. When you scan a barcode, the scanner may be configured to send the decoded information to a computer, just as if you’d typed the information using the keyboard. The decoded information is inserted in the application at the point where the cursor is. Lastly, for improved durability the handsets also feature: rubberized gaskets, removable battery packs for 24x7 usage, an internal magnesium frame to protect the internal circuitry, shock-mounted LCD, and over molding.
Alteon 10000 Application Switch
The Alteon 10000 Application Switch is a high performance, NEBS 3-compliant carrier-grade ATCA platform that delivers on demand, extendable throughput of up to an impressive 80gbps of application delivery capacity. It provides advanced application acceleration capabilities and scalability for carriers, mobile operators, ISPs, and large enterprises data centers that require a high-end ADC solution. Performance metrics include 1.4M layer 4 connections per second, 800K layer 7 connections per second, and 44M concurrent connections.
The Alteon 10000 is built on a modular, ATCA chassis. It features 15 ports of 10GE/GE (SFP+ pluggable optics) and additional 8 ports of 1GE (copper). The 6-slots chassis of Alteon 10000 accommodates four payload blades, each providing 20gbps of throughput, up to a total of 80gbps. All the chassis blades are hot swappable, allowing blade replacements without stopping the entire chassis and ensuring maximum uptime. Alteon 10000 also provides high MTBF with three AC/DC load sharing, hot-swappable power supplies and two hot-swappable fan trays.
One other innovative feature is that the Alteon 10000 takes advantage of the industry-peerless Virtual Matrix Architecture technology. VMA is a fast and flexible architecture embedding multi-CPU and multi-core components, which leverages the entire system’s capacity while providing the parallel performance of distributed processing resulting in linear layer 4-7 scalability.
StreamGroomers and SGM (StreamGroomer Manager)
Business-critical unified communications & collaboration applications from Cisco, IBM (News - Alert), Microsoft and others are making monitoring, controlling, and prioritizing data travelling over bandwidth-restricted WAN networks even more critical. One challenge is getting visibility into your network and ensuring that UCC applications have the bandwidth they need, especially when they run over bandwidth-restricted WAN links. Video traffic in particular is a challenge for enterprise networks because it uses a large amount of bandwidth, but even non-UCC apps like P2P sharing apps can quickly bring a WAN to its knees.
Streamcore helps customers have this visibility and gives them the ability to apply practical business-based UCC policies to manage performance for real-time video, VoIP and collaboration applications, whether delivered over traditional private networks or through cloud-based solutions. Streamcore ensures that organizations can cost-effectively employ IP telephony, unified communications and either desktop or room-based videoconference solutions without any network performance degradation.
The StreamGroomers are plug-and-play appliances that can be deployed in any network location including data centers, headquarters, branch offices or in front of Internet access links. StreamGroomers are positioned in line between the LAN and WAN access router. Streamcore traffic management offers combined network flow analysis and monitoring, QoS enforcement and traffic shaping. Even traffic exchanged with remote sites without a StreamGroomer can be monitored and controlled. This is a key capability for managing any-to-any VoIP/video flows or traffic coming from third-party data centers, such as private or public cloud service providers.
The StreamGroomer Manager (SGM) is a Web 2.0-based centralized platform to manage all visibility and control services provided by StreamGroomers. Streamcore’s centralized management provides unified centralized management on a single server for all operations including, real-time monitoring, performance supervision, reporting, control policies provisioning, StreamGroomer management, etc. It can manage 2,000 StreamGroomers and has support for multi-tenancy.
Streamcore tells TMC Labs, “Streamcore has developed the first product on the market unifying all visibility and control features required to solve the network performance challenge for UCC, with new innovative technologies. What also makes Streamcore unique is its business-oriented management approach, a novel breakthrough in network management. Streamcore’s objective is to empower IT/UCC executives with a set of capabilities which, for the first time, enable them to manage their networks and services in a business-oriented fashion. Given Streamcore's unique combination of core technologies, executives can pragmatically apply business requirements to UCC visibility and control policies and allow Streamcore’s solution to manage all of the underlying complexity.”
The key to this secret sauce is the company’s deep packet inspection. Streamcore’s hierarchical DPI is based on a unique embedded software module that detects all forms of application and UCC traffic on the network: data vs. audio vs. video, codec, clock rate, SSL-encrypted webconferencing, etc. With this technology, customers are able to set up relevant visibility and control policies per UCC traffic type. Highlighted features include real-time troubleshooting (passive audio/video measurements, network performance statistics), network assessment (active audio/video measurements, traffic auto-discovery), traffic shaping, advanced QoS (to manage prioritization between types of applications and UCC traffic), desktop video QoS engine, and WAN load balancing. Lastly, the solution provides performance measurements for any incoming RTP audio/video traffic, including MoS, latency, and packet loss.
Sunrise Telecom RxT Smart Productivity Test Platform with realGATE
The rapid growth of cable, telecom, and wireless services has presented challenges for service providers tasked with deploying, testing, and maintaining new technologies. Having separate testing tools requires significant training and larger capital expenditure on multiple test platforms. Sunrise Telecom developed an all-in-one solution to address these challenges. The RxT is a single handheld testing device that enhances field technicians’ productivity with its innovative removable test modules and ability to handle deployment, measurement, and troubleshooting, reducing the need to carry additional gear in the field. The device features QuickSWAP modules, which can be easily popped in or out. It also makes the device future proof as new technologies, such as new network connectors, are developed.
The RxT is the first test and measurement platform to include GPS-based geo-tagging, providing service providers the opportunity to add location stamps to test records. This feature provides customers with an extra level of documentation and validation (e.g. identify test sites).
It features a large color touch screen, wireless connectivity, long battery life, smartbook functionality, and centralized workflow optimization. In addition, the RxT integrates with realGATE for asset management, report management and workflow optimization helping to reduce capex/opex and helps ensure high-quality subscriber services. Fully integrated with Sunrise Telecom’s realGATE workflow optimization system, the RxT goes beyond testing and offers a complete managed solution for telecom, cable, and mobile operators. This field platform is capable of quickly verifying a wide range of advanced services in a single modular handheld device. Further, the RxT’s ease of use minimizes learning time, quickly boosting productivity, translating into jobs done more quickly and efficiently with more satisfied customers and fewer repeat truck rolls.
Vertical Communications Inc.
Wave ISM 2.0
Wave ISM 2.0 is Vertical’s latest software upgrade for the Wave IP 2500 and Wave IP 500 Business Communications Systems. The Wave IP 500 is designed for branch and small offices, supporting up to 50 users while Wave IP 2500 is designed for medium offices, supporting up to 500 users. It features comprehensive unified communications capabilities, contact center functionality, call recording, reporting, and custom call routing applications.
Additionally, with the 2.0 release Vertical announced the field trial of an integrated fax server and Voice Server 2.0 , an integrated vXML IVR platform, which now offers both automated inbound and outbound applications. A new feature in 2.0 is the ability to have a mobile extension, which can be a user’s cell, home phone, or a softphone. They offer a feature-rich softphone called ViewPoint Phone. Wave Impulse, an integrated, secure and private instant messaging solution adds to the UC functionality. Also part of the UC feature set is that it now has Microsoft Exchange integration with always-on real-time synchronization of contacts and voice mail. The ViewPoint Shared Folders feature lets you share ViewPoint Call Monitor, contacts, voice mail and call logs between users. It also sports powerful find-me/follow-me rules. One unique feature is cascading voice mail, which is a rules-based distribution and escalation of voice mail notifications, so important customers’ messages aren’t ignored. A Wave Client API is available, enabling developers to extend the functionality of Wave as well as integrate with third-party enterprise applications. Unlike many competing systems, call recording is standard on Wave IP. Lastly, there are many voice applications that are offered for Wave IP, including an API client that allows customers to create custom integrations with third-party applications such as CRM, billing, hospitality, pharmacy software suites and more.
Virtual PBX Complete
Virtual PBX Complete is a hosted IP PBX designed for small and mid-size businesses. The company claims to have invented the first true hosted PBX in 1996, and we have no reason to disagree with that assertion. Unique to the offering is that you can use any combination of existing phones, phone switches, analog or IP phone lines, cellular phones, or any other type of phone. Virtual PBX explains, “Unlike other hosted PBX services, Virtual PBX Complete supports complete blending of analog and IP telephony, while simultaneously incorporating open SIP peering. Most traditional hosted services work with analog lines but do not handle VoIP. Most IP PBX providers require IP telephony for all users and only allow PSTN use in call forwarding. Our service embraces both PSTN and SIP traffic at all times and in all places. We also support SIP standards, allowing clients to use phones and VoIP services from other providers when desired, instead of requiring our own proprietary VoIP registration.”
Virtual PBX has other innovations to its credit. It was the first company in the hosted PBX space to offer ACD queuing for call center applications, call routing to a distributed workforce, find-me/follow-me call forwarding, auto-attendant greetings, menus in a distributed environment, and much more.
Important features include supervised call transfers, multi-business support, multi-stage dialing, and automatic routing based on incoming caller ID. Unlike many hosted PBX offerings, Virtual PBX features both powerful hunt groups and powerful ACD queues. ACD options including load balancing, skills-based routing, hierarchical routing, overflow routing, call hold and callers waiting limits. The system includes real-time monitoring of phone system activity by extension or department, including callers on hold, calls in progress, hold times, caller IDs, and more.
It also features Smart Caller ID and Call Preview, which let workers know who is calling, what they are calling about, and how the call entered the system before answering the call. Voicemail interrupt is a nice feature that lets users send callers to voicemail and listen in on the message being left. If the user decides to accept the call, the voicemail can be interrupted and the call connected.
Vocalcom’s Hermes.net V4.1 provides an inbound/outbound solution, which can be used as a standalone contact center application or can seamlessly connect to an existing PBX. If a customer is using a legacy Avaya switch, Vocalcom can provide native integration with outbound predicative dialing and scripting.
The platform offers a seamless CTI layer for the Avaya platform and connects directly to Avaya’s network via AES Server. Vocalcom provides an open database allowing end users to customize the system to their needs. The Hermes platform handles both voice and data with a comprehensive set of management tools. The latest version, V4.1, provides powerful tools such as CTI, IVR, ACD, predictive dialing, scripting, e-mail, chat, fax, CRM integration and recording under a single VoIP/SIP enabled platform.
Vocalcom also offers a hosted option. At the customer’s location Vocalcom only requires a PC with a web browser and bandwidth of 100kbps for each agent to run the application. Vocalcom allows both IP phones as well as a provided softphone as phone options. Vocalcom tells us, “Vocalcom was the first to market with a unified multimedia application – inbound, outbound, fax, SMS, chat, e-mail, print; first to market with the .NET call center application; and truly open SQL/.net architecture.”
Tom Keating is Vice President and Chief Technology Officer at Technology Marketing Corporation, and Executive Technology Editor/SEO Director for TMCnet.com. To read more of Tom’s articles, please visit his columnist page. He also blogs for TMCnet here.
Edited by Stefania Viscusi