TMC Labs 2003 Innovation Awards: Part I

BY THE STAFF OF TMC LABS

The TMC Labs Innovation Awards are all about recognizing unique and innovative products and solutions within the Internet telephony industry. As the fourth annual TMC Labs Innovation Award within INTERNET TELEPHONY� Magazine, we somewhat expected to see slightly less innovative products than in past years, since the market has begun to mature. We were pleasantly surprised however by the sheer number of innovative products that applied for this award.

In fact, with over 200 applications we couldn�t narrow the winners� field down to a dozen or so as we have in past years. There were just too many innovative products to not include in this awards special based solely upon some arbitrary �winners� number. Try as we might, we couldn�t whittle down the winners list to any less than 22 winners -- they were all very deserving.

We attempted to squeeze all 22 winners into a single issue, but our editor said it would require mass trimming of our very detailed descriptions. Rather than cutting the descriptions, we thought it would benefit our readers to keep the more detailed information. Our editor compromised and said we could keep the detailed descriptions, but we would have to break the TMC Labs Innovation Awards into two parts; so without further adieu, here is the first half (11) of the 22 winners (with next month�s issue containing the second half). The list was split up arbitrarily; for a full list of winners, check out the following table.

ARRIS
Touchstone Telephony Modem 202P
www.arrisi.com

Targeting cable operators, MSOs (MultiService Operators) and other Broadband Service Providers, ARRIS� Touchstone Telephony Modem 202P (TM202P) is designed for primary line telephony that can replace the service offered by the current ILEC (incumbent local exchange carrier) or CLEC used by residential customers. The TM202P connects directly to the coax cable coming from your cable service provider and delivers two lines of toll quality VoIP. What is quite unique is that no cable modem is needed because the TM202P is an embedded multimedia terminal adapter so it functions as both the high-speed data interface (i.e., cable modem) and the telephony interface. The TM202P is PacketCable 1.0 Certified and also supports DOCSIS 1.1 and PacketCable 1.1. It currently supports G.711 with G.728 and G.729e in development. ARRIS claims that the TM202 was the first E-MTA with an integrated battery to achieve PacketCable Certification from CableLabs. They also claim that the TM202P has the longest battery back-up time (over eight hours) of any E-MTA. Thus, if there is a power loss, you don�t lose your phone service. In addition, the TM202P has software that contains a configuration file editor called PacketACE that allows operators to adjust various packet telephony configurations when network equipment or parameters change.

Grandstream Networks, Inc.
HandyTone Analog Telephone Adaptor
www.grandstream.com

Many within the VoIP industry are quite familiar with the SIP-capable Cisco ATA-186, which is used by many ITSPs. One of the reasons for the popularity of the Cisco ATA-186 is that it was the only game in town if you wanted single-port analog-to-VoIP connectivity. Well, Cisco�s ATA-186 now has some competition on its hands with the release of Grandstream Networks HandyTone Analog Telephony Adaptor. It�s very similar in functionality to the Cisco ATA-186. The HandyTone analog telephone adaptors allow residential or SOHO users to use their existing analog phones (including cordless phones) or fax machines to place voice or fax calls over the Internet.

One of its innovative features is that it is so small. Unlike the Cisco ATA-186, you can fit the HandyTone in your pocket (slightly larger than a credit card) and take it with you when traveling, thus enabling VoIP connectivity in hotels, or other destinations with a broadband connection. The HandyTone products supports SIP2.0 (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, DNS, DHCP, NTP, TFTP, DIGEST/BASIC authentication with MD5 and MD5-sess algorithms, and STUN (Simple Traversal of UDP thru NAT), Layer 2 (802.1Q VLAN and 802.1p) and Layer 3 (ToS, DiffServ) QoS, and transparent Fax pass through. It also offers easy configuration via keypad, Web browser, or a central configuration file for mass deployment. Finally, with a low retail price of $75 each, Grandstream�s HandyTone telephone adaptors offer an extremely compelling value that is sure to make waves in the Internet telephony industry.

iSoftel Ltd
iRoute Intelligent Routing Management System
www.iSoftel.com

In the past the routing of calls in VoIP networks was either done by configuring the rules directly at the endpoints (media gateways) or by moving these decisions to the gatekeepers. This concept worked well as long as the networks remained small. However, as competition in VoIP grew, there was a need for sophisticated routing features to be built into the network to enable the service providers to choose the best possible route to complete a call. The rules for selecting the terminating endpoint could be based on the time of the day, ingress carrier, destination country, etc.

The reason for not building sophisticated routing features into gatekeepers was that the route selection process can be resource (CPU/RAM) intensive and hence, could prove prohibitive to sacrifice the call handling capacity to sophisticated routing. The other reason for not making gatekeepers too intelligent in routing is that there might be many gatekeepers, thus making it cost prohibitive. Also, the routing intelligence would be scattered all over the VoIP network, which might lead to errors in configuration, which would mean lost revenue for the service provider.

Thus, it would be better to have a dedicated server that makes all the routing decisions, allowing all the gatekeepers within the network to then query this server for the best route for every call. iSoftel�s iRoute is one such centralized routing solution. It is a scalable, fully redundant, and advanced route management system that helps service providers to optimize the routing of network and calls in the VoIP network. It supports a wide variety of routing policies, such as least cost routing, percentage routing, sequential routing, time-independent routing, overflow routing, alternate routing (auto-reroute), and source-based routing. Centralized provisioning and management of the routing information in iRoute is done via a Web browser.

Essentially, iRoute is a software application that enhances the Cisco Global Long Distance solution by enabling advanced, complex routing logic based on business rules. It is part of the Cisco Voice Infrastructure and Applications (VIA) solution. It provides centralized provisioning and control of network routing decisions for Cisco H.323 networks. It works in conjunction with the Cisco Gatekeeper by leveraging the Gatekeeper Transaction Message Protocol (GKTMP) interface. The iRoute provides the ability to optimize the network based on least cost, time-of-day, resource availability, trunk-group, and percent allocation routing.
iSoftel features a few other innovative features. Since jitter, latency, and dropped packets can all affect VoIP quality, iSoftel has a QoS monitoring server that tightly integrates with iRoute and dynamically decides on best route to be used. In addition, this solution can ensure that calls are not routed to a gateway that is not able to handle the call due to maintenance or a lack of available free resources. Another innovative design in this solution is that iRoute stores all the routing data in memory (RAM) and hence does not query the database for every call routing request that it receives. This achieves a very fast response time, which translates to increased call handing capacity. iRoute also has special features such as Call trace and Lawful intercept that aid in government agencies for tracing specific calls.

NetRake
nCite
www.netrake.com

Netrake�s founders took a look at what worked well in IP such as security and analyzing the traffic content (deep packet processing), leveraged their PSTN knowledge, and then applied that to a VoIP environment. The result was the nCite Session Controller, which is an IP Session Controller designed to enable secure IP network interconnection to deliver VoIP services over any network (public or private). NCite, a Layer 2�7 IP platform, delivers secure, network interconnection and offers deep packet processing allowing it to see past the packet header and into the packet payload. This helps deliver a great deal of information including performance metrics to the service provider. It features 10/100 BaseT, Gigabit Ethernet, DS3, OC-3, OC-12, and OC-48 ATM Interfaces, five gigabits of throughput, six million simultaneous sessions/frame, and <50 microseconds of latency through the system.

In order for VoIP to achieve parity with the PSTN, VoIP will require native interoperability, predictable performance, call integrity, and traffic control across network boundaries. VoIP networks are typically built as �island networks� due to complexities associated with IP interconnection or peering (such as address overlap and security firewalls). In addition, many VoIP networks are over-provisioned in order to accommodate peak call times without degrading performance or dropping critical in-band control traffic. This results in bandwidth inefficiencies and unnecessary costs, diluting the benefits of VoIP. Netrake�s nCite solution resolves the peering, latency, quality of service, capacity, and control issues associate with VoIP.

For instance, their solution can recognize voice traffic, prioritize it relative to other traffic, and then route it on explicit paths to minimize latency and jitter issues. nCite classifies traffic by application type using any number of parameters inside or across packets in a session. Once a specific voice call is identified, nCite modifies the packet settings to the highest priority resulting in excellent quality of service.
In addition, nCite solves firewall/NAT issues associated with VoIP by dynamically authenticating, identifying, and allowing outside calls to complete without compromising network security. nCite scans the payload of a VoIP control message to extract the address and port of the called party and associates that address with an incoming call. nCite then dynamically establishes �pinholes� through its firewall for just that one call or session. Once the call or session is finished, nCite closes the pinhole -- all the while maintaining network security. Network addresses, for both control and bearer traffic are �anchored� to the nCite device using NAT and NAPT as well as an innovative payload NAT. These enable the interconnection of separate private domains with public networks, providing peer-to-peer communications across IP network boundaries. By providing performance metrics, QoS, VoIP NAT traversal, and deep-packet processing, Netrake�s nCite will enable service providers to deploy and maintain VoIP services much more easily.

Radcom
The Cellular Performer
www.radcom.com

RADCOM�s Cellular Performer is an innovative testing tool that fits into the Internet Telephony/VoIP space due to the convergence of VoIP and cellular networks. 3GPP Release 4 is a set of specifications defined by the 3G Partnership Project, the standard making body for third-generation mobile telephony, for all GPRS/UMTS network elements. The main purpose of 3GPP R4 is to introduce separation of connection, its control and services for the Circuit Switched Core Network to make it freely scalable. It also endeavors to route a part of voice calls over IP, thus facilitating the transition to all IP networks.

New components, like the Media Gateway and the Media Gateway Controller, defined as part of 3GPP Release 4 standards, use VoIP stack (AMR/RTP/UDP/IP at the user plane and MEGACO at the control plane) in GPRS/UMTS networks. With the introduction of Push-to-talk over Mobile (PoM) � an application that will use the �always on� characteristics of GPRS and enable mobile subscribers two-way-radio-like service using their GPRS handsets, building a Voice/RTP stream on top of the IP connection created by the GPRS session.

So with the convergence of the telecom (cellular) and the datacom worlds (IP), how does one manage and test such a complex integration? Well, RADCOM�s Cellular Performer monitors and troubleshoots both voice and cellular data applications. It can follow the entire call setup signaling at the SIP-based VoIP network domain, MEGACO-based MGW domain, SS7-based CS Core domain, GTP-based PS Core domain and up to the RAN and UTRAN. The Cellular Performer integrates voice simulation and quality-assessments tools with the sophisticated session-oriented �Consultants� to evaluate network performance and reduce troubleshooting time.

3G cellular is a fairly complex network that carries voice and data. Subscribers and session identifiers at different layers are used. Hence a �smart� application is required to automate the decoding of the captured data into meaningful information. RADCOM�s Cellular Performer does exactly that -- it deploys smart decode engines required to reconstruct captured traffic into information useful to network managers.

The Cellular Performer is quite unique in that unlike many of its competitors it offers a complete �soup-to-nuts� solution that includes protocol verification, cell/frame-level analysis, voice call and IP session analysis, streaming media, and voice quality testing for GPRS, EDGE, UMTS, and CDMA2000 technologies.
In addition, it decodes over 550 telecom and datacom protocols, including standard protocols such as ITU-T, ANSI, IEFT, 3GPP, and 3GPP2 as well as country/vendor-specific variants. This solution utilizes RADCOM�s proprietary Generic Analyzer Processor Chip � the GEAR T. This chip is a fully customized analyzer that provides hardware-based, full line rate analysis capabilities at up to 2.5 Gbps. TMC Labs has always been impressed with Radcom�s testing products (they�ve won several TMC Labs awards) and Cellular Performer was no exception.

Siemens Enterprise Networks
Siemens OpenScape v1.0
www.icn.siemens.com

This product is one of the most exciting and innovative Internet telephony products we�ve seen in a while. Siemens� OpenScape is a real-time communications application suite that creates intelligent relationships among all the communications resources at work in the enterprise, which can be managed by rules established by users. The product consists of three primary functional elements, including Communications Broker. The OpenScape Communications Broker is SIP-based middleware that establishes intelligent relationships among various communication resources (voice system, voice mail, e-mail, instant messaging, cellular/wireless voice/data, etc.), and provides the rules engine that users leverage to create presence-sensitive call treatments for all forms of communication.

For example, individuals can designate whether incoming phone calls should be sent to their office phone, cell phone, home office phone, or a colleague�s phone (or other), based on who is calling, the time/date, and the user�s current working context (in a meeting, on vacation, at a client site). The same is true of e-mail and instant messaging. The Communications Broker also enables multi-resource communication by managing session set up transparently to end users. In other words, the user experiences the communications event as a single session, not separate sessions. It further enables all communications capabilities to be accessed via speech, telephone, or graphical user interface (SUI, TUI, GUI).

The OpenScape personal portal provides a single point of control for call treatment rules, presence updates, changing the user�s preferred device for incoming calls, establishment of collaboration sessions, and the OpenScape Contacts List � an enterprise class buddy list. The Contacts list displays colleagues� availability for communication via phone, e-mail or instant message. If the colleague is available, the user does not need to know which device or address to access. Clicking the icon (or accessing a master phone number) will direct their communiqu� to the preferred device/address of the colleague. This eliminates the need to communicate by trial and error, saving time and the cost of unsuccessful long-distance calls, cellular minutes, and message storage.

The personal portal capabilities can be accessed via SUI, TUI, or any number of GUIs, ranging from the OpenScape Personal Portal Client to Windows Messenger Clients to PDAs or applications in which the capabilities have been embedded. Using OpenScape, users can click to set up conference calls on demand, using the conference bridge resident in the OpenScape application. OpenScape users simply select the people they need in conference and click. OpenScape locates the preferred device of the attendees and calls out to them. The OpenScape workgroup portal saves time scheduling conference calls and distributing access numbers, logging in and waiting for others to join, setting up separate voice and data conferences, and synchronizing documents.

In addition, it brings facilitation of most conference calls in-house, reducing the outsourced conferencing costs, as well as leveraging the enterprise�s long-distance rates. We should also mention that OpenScape is one of the first applications to utilize Real-Time Communications Server 2003 (RTC Server) within Microsoft�s Windows Server 2003. Overall, this product is designed to enable something that we all want � the ability to control how and when we are contacted and to which device. TMC Labs is very intrigued by the features of OpenScape and we look forward to testing this product more thoroughly in the near future.

Snom Technology AG
Snom 200
www.snom.com

One of the problems with some hardware-based SIP phones is that they can be just as expensive as their PSTN/digital PBX phone counterparts. Well, Snom Technology has done an excellent job of setting a price point for their Snom 200 SIP phone ($360) that will certainly aid SIP deployment. The Snom 200, running Linux under the hood, supports not only a SIP stack but also supports H.323/H.450. This is critical for any organization wishing to slowly migrate from H.323 to SIP. We are unaware of any other phones supporting both stacks simultaneously. It features a 2x24 character display with graphical field and five programmable function keys. Complementary software enables integration with Microsoft Outlook and other TAPI-based applications.

In addition, this phone offers mass deployment system to providers that want to deploy with Web-based provisioning. It supports phone setup and error tracing via http. It also traverses NAT for registration and service using UPnP and STUN. In fact, Snom claims they were the first to apply the UPnP and STUN standards to hard phones. Snom phones� Web server also offers a complete trace and logging function for both H323/H450 and SIP calls, which is very helpful during interoperability testing. Importantly, Snom phones support the SIP notify function to receive text messages from an IP-PBX in a standard fashion allowing software developers an easy and convenient way to communicate with the phone. The phones come with an extra Ethernet port with VLAN support also enables you to �daisy chain� the phone with your PC without the necessity for another LAN port. One final feature of note is that it has integrated LDAP to enable administrators to set up and use company and global phone books.

Sprint Corporation
Sprint Unified Communications-Total Access
www.sprint.com

Certainly, when TMC Labs thinks of VoIP or especially the open SIP standard, a carrier such as Sprint does not come to mind. In fact, with VoIP eroding the major carriers� market share or at least helping to bring down pricing on billable minutes, we were pleasantly surprised by Sprint�s application submission, for a product that leverages VoIP using the SIP protocol. Called Sprint Unified Communications-Total Access, this product is unique in the industry as a method to unify communications across multiple user technologies (WiFi, video) and multiple network technologies (VoIP, 3G, PSTN). Sprint Unified Communications is a fully network-based voice over IP system that enables a user to create a fully customized communications experience. This experience marries voice, video, wired, and wireless devices into a seamless communications system.

An example of how this network could be used includes handling the communications needs of a physician throughout the day. In the morning, as he arrives into the office, the Sprint PCS Phone he carries on his hip is detected by the system � it �knows� he�s at his desk � and reroutes his PCS calls to his desk phone. Later as he begins �rounds,� he carries a Tablet PC that is both a terminal for Electronic Medical Records and a telephone. The physician could then set his profile to his own preference setting � for example, �On-Rounds.� At this point a predetermined profile is enabled. This profile forwards calls that are designated as low priority directly to voice mail. High-priority calls are routed to the tablet PC over the WiFi network in the clinic. Upon returning to the office, his profile is then set back to �In Office� and his calls are routed to his wireline desk phone. Essentially, this system is quite similar to the Siemens product we mentioned earlier; with the main exception being that Sprint�s solution is a hosted solution and Siemens�s offering is a CPE-based solution.

The overall system includes network-based IP Centrex functionality that is protected by a 3DES IPSec tunnel. The Sprint network hosts all PSTN gateways, signaling infrastructure, and application servers to minimize cost to the consumer. The Sprint wireline and wireless voice and data networks are converged into a seamless user experience in this manner. Using a network-based approach, rather than CPE-based, the customer investment is minimized.

Additionally, applications at the user location can further integrate the network. Sprint has developed augmentations to the network-based applications to modify call controls upon the detection of a specific PCS phone or Secure Access Card. This capability further adds to both the security of the communication systems of customers and the reduction of administrative burdens related to user name and password recognition. It does this through substituting a traditional computer log-on with the access system. Additionally, Sprint recognized the powerful capabilities brought to bear by integrating Wireless Personal Area Network (WPAN) technology with the system. This technology acts as one avenue to manage the location and customer calling experience.

The true innovation of this product is the marriage of SIP, VoIP, WiFi, and 3G technologies. According to Sprint, early in their research, they saw SIP as an intrusive technology, but they quickly realized that the marriage of the legacy and future networks was critical to their future success. Kudos to them for being technologically progressive and not sticking with the status quo!

Voyant Technologies, Inc.
ReadiVoice
www.voyant.com

Voyant�s ReadiVoice enables high-capacity, always-available, totally automated conferencing. Their conferencing technology enables end users to initiate conference calls without the need to make reservations or contact operators. The subscriber starts, controls, views, and ends calls using telephone touch-tones or the Web.

ReadiVoice utilizes Voyant�s next generation carrier-grade conferencing media servers called InnoVox 4000, which grants ReadiVoice the ability to scale much more than previous versions. InnoVox 4000 offers up to an impressive 4,032-port capacity per single bridge. Voyant claims that this makes the InnoVox 4000 the highest density system available in the marketplace. Additionally, they�ve designed it so that it occupies less than 30 percent of a standard 19-inch equipment rack. It supports multiple connectivity options to let customers choose the telephony configuration that best fits their existing network, T1, E1, or VoIP. In addition, Voyant claims that they are the first to offer a T3 connectivity to the bridge. Further, you can deploy multiple bridges to achieve up to 20,000 ports in a single system, using Voyant�s SS7 call routing technology.

Other key differentiating features in the InnoVox 4000 include a lack of any single point of failure and its use of distributed resources to maximize and optimize bridge-wide usage. The bridge has been designed to reduce any single point of failure. Major hardware components, such as fans, power supplies, and cards are all hot-swappable. The cards are all self-managed, ensuring no single point of failure, and facilitating the sharing of resources amongst cards as needed. The Ethernet switches are redundant, ensuring failure with no impact on users, in the event of an Ethernet failure. With an impressive scalable architecture, redundant failover capabilities, and ability to manage conferences from the traditional phone or via the Web, TMC Labs would be hard pressed not to merit Voyant a TMC Labs Innovation Award.

Xten Networks, Inc.
X-PRO (Soft SIP VoIP Phone)
www.xten.com

Xten Networks has released a feature-rich SIP softphone called X-PRO that includes features that TMC Labs has not seen in other open SIP clients, including: six lines, one-button conferencing, call forwarding, multiple SIP proxy support (10 Proxies), and is SOHO NAT compliant for traversing those pesky home NAT firewalls. This software is rather innovative in that it allows users to connect to multiple SIP proxy/registrars simultaneously, which is great for multi-network environments.

X-PRO has a feature called one-button conferencing that allows conferencing across both IP and PSTN networks. With Multiple SIP Proxy support conferencing across IP & PSTN networks is a snap. Other features include touch-tone (DTMF) support, Line hold, Line Transfer, Do Not Disturb, Inbound Call �Ignore�, Inbound Call �Go to Voice mail�, Call Forwarding URI/URL, Voice mail URL, Multiple SIP Proxy Registration [10 Proxies], Dial/Redial/Hangup, Recent Numbers List, CODEC Selection, Caller ID [SIP ID], Call Timer, Silence Threshold, mute, Push-to-Talk [Toggle Switch-PocketPC], and more.

They also offer a free version called X-Lite, which is 100 percent SIP-compliant, as well as a SIP client with 128bit to >2,500bit encryption called X-Cipher for those either paranoid about eavesdropping or those that just plain think it�s a cool feature to have. Overall, TMC Labs was very impressed with the feature-set of Xten Networks� SIP client as well as its low price point -- $50 for X-PRO, $75 for X-Cipher, and free for X-Lite. Finally, take our recommendation and check out their amusing promo piece, which is a cartoon created using Macromedia Flash. Just go to http://xten.com/promo/x-pro_episode1.html or go to their home page to view the cartoon.

Zultys Technologies
MX1200
www.zultys.com

TMC Labs has preached, no, make that PLEADED for PBXs to be more open and using open standards for more than eight years. Well, our begging and pleading has finally been heard; Zultys Technologies has combined the open SIP standard with the open Linux platform to develop an open IP-PBX solution called the MX1200, which is not only an IP-PBX, but it�s also a router, firewall, and Ethernet switch all rolled into one. In fact, 12 of the 28 Ethernet ports on the internal switch can provide power over Ethernet according to IEEE 802.3af. No A/C adapter is required for the Ethernet phones connected to these 12 ports.
Targeting the enterprise IP telephony market, the MX1200 integrates multimedia communications into a compact system that has standard interfaces for all connections. Specifically, it combines the functions of an IP PBX (automated attendant, voice mail, and ACD groups) with an Internet gateway, switch, and router and allows all functions to be managed using a single Windows application.
As we mentioned, this is an �open� IP-PBX, which is demonstrated by the fact that this product works with SIP-based phones from ACT-TEL, Cisco, IP Dialog, Pingtel, and Snom, and can even use the SIP-based Microsoft Windows Messenger as a softphone. In addition, The MX1200 has taken the concept of presence within instant messaging and taken it a step further by tightly coupling presence and availability to call handling.

The MX1200 is powered by a real time version of Linux running on four IBM 440GP processors and can scale up to 1,200 users with no hardware expansion necessary. Expansion is performed using software licenses so additional functions and capacity are available instantaneously. Other specifications include 16 automated attendants, 400 hours of voice mail with 48 simultaneous accesses, 16 operator groups and 64 ACD groups, up to eight T1 or E1 circuits, and up to eight analog circuits for connection to fax machines or other analog devices. TMC Labs can unequivocally say that Zultys Technologies� M1200 is not only innovative, but it�s the most open PBX platform we have ever seen. It�s about time.

[ Return To The July 2003 Table Of Contents ]