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May 2006
Volume 1 / Number 3


VoiceXML Gains Acceptance as Industry Standard
By Mae Kowalke

Microsoft (quote - news - alert) announced that Speech Server 2007 - the latest version of the company's speech and telephony platform - will provide full support for VoiceXML. VoiceXML is an open standards-based language used for developing speech applications, managed by the World Wide Web Consortium (W3C). Microsoft said the addition of VoiceXML compatibility (Speech Server continues to support Speech Application Language Tags, or SALT) will "enable customers to choose the development standard that will work best in their environment." According to Clegg Ivey, Vice President of Operations at Voxeo, Microsoft's adoption of VoiceXML is a good indication that the language has matured into the industry standard. "When Microsoft signs on to a standard, you know you're in good shape," he said.

Because of the VoiceXML standard, companies can deploy speech products knowing that when they upgrade or switch vendors down the road, they won't have to re-write all their applications. VoiceXML can be used to create any type of interactive speech application, including voice-enabling of call centers, interactive tracking applications, and outbound notification.

Back to the Future: MindSpring Redux

Can a blast from the past ring in the future? EarthLink (news - alert) thinks so. Bringing back one of the 90's most successful Internet brands, the company relaunched MindSpring. The new MindSpring is EarthLink's softphone client that combines Internet voice and instant messaging on a consumer's PC. Integrating popular Internet applications and Voice over IP (VoIP), MindSpring delivers high-quality, crystal-clear voice calling capabilities and real-time text (instant messaging) and access to free voicemail and EarthLink Web Mail. This combination allows MindSpring users to have one identity for e-mail, instant messaging, and phone calls over the Internet. MindSpring is built on the SIP (session initiation protocol) Internet standard, a platform that makes it compatible with any other open SIP-based network. MindSpring and Google Talk are also interoperable for instant messaging.


Juniper Develops Tool for MSOs to Secure VoIP, SIP Services

Juniper Networks Inc. (news - alert) has unveiled a new product, the Cable Dynamic Threat Mitigation tool, which helps cable MSOs protect VoIP and SIP services against network security attacks. The Cable Dynamic Threat Mitigation solution works with any PacketCable Multimedia-compliant cable modem termination system. It integrates Juniper's routing platforms, intrusion detection and prevention devices, and SDX Service Deployment System so MSOs can identify and isolate security threats, then notify operations staff and infected customers. The intrusion detection and prevention device monitors for network attacks on a per-user, perapplication, or on-demand basis, and detects offending traffic or a DDoS attack. It then notifies the SDX-300, which sends the appropriate policy to any Juniper Networks routing platform and/or third-party PacketCable Multimedia-qualified cable dynamic threat mitigation or metro routing platform. According to the nature of the threat, MSOs can choose to rate limit or filter traffic, drop the call, or redirect the user to a Web portal, and automatically re-establish service once the threat is eliminated.

Juniper's Cable Dynamic Threat Mitigation tool also can provide security for SIP-based services, such as online gaming, video conferencing, instant messaging, and push-to-talk.


Woize International Ltd. to Launch Symbian Client

Woize International Ltd., (news - alert) the digital communication service provider today announces plans on launching a Session Initiated Protocol (SIP)- based Symbian Woize client for the benefit of worldwide users during Q2. Symbian is an operating system based on open standards for smartphones. By launching a SIP-based Symbian 9.1 client, Woize creates an opportunity for worldwide advanced phone users to benefit from the Woize services, thereby reaching even further into the opportunities of the market of free mobile phone calls enabled by WiFi networks.

The Symbian Woize client is planned to have similar functionalities as the SmartphoneWoize client, allowing users to make free calls to other Woize, PocketWoize, SmartphoneWoize, and Symbian Woize users, making inexpensive calls to users outside of the Woize community, and receiving phone calls to the client anywhere in the world.


Dialexia and Grandstream in SIP-eroperability Testing Deal
By Erik Linask

VoIP solutions provider Dialexia Communications (news - alert) and Grandstream Networks, (news - alert) which provides IP telephone hardware, have successfully completed comprehensive interoperability testing between Dialexia's IP-PBX Dial-Office and Grandstream's GXP2000 SIP IP Phone. The partnership is set to allow future service providers and enterprise customers easy VoIP deployments. With the need for connecting distant branch locations, while maintaining the functionality of existing PBXs, customers are looking for an edge, which often is found in ease of implementation - and cost, of course.

Dialexia's flagship product, Dial-Office, is one of the lowest cost VoIP solutions available to enterprises. It is designed to easily integrate with legacy TDM and IP networks, thus offering customers the dramatic savings that often drives them to a VoIP solution. Grandstream Networks' GXP-2000 is a next-generation enterprise IP telephone based on open SIP standards. Built upon unique and proprietary technologies, it features superb audio quality, rich functionalities, and excellent manageability at affordable prices.

The Grandstream GXP2000 IP-Phone combined with Dialexia's Dial-Office brings a rich feature set and extraordinary value to both consumer and corporate markets. With adherence to the SIP standards the products are well suited for enterprise users of IP PBX systems, remote office users, and many other cost saving and productive uses.


Ubiquity Launches Ubiquity Developer Studio to Speed Deployment of SIP Applications

Ubiquity Software, (news - alert) developer of standards-based SIP (Session Initiation Protocol) application servers, announced the launch of Ubiquity Developer Studio (UDS), a development and debugging platform for its SIP Application Server (SIP A/S). UDS provides a simplified set of tools that enables SIP developers to build highly modular applications and services that run on top of the Ubiquity SIP A/S.

UDS is designed to work in conjunction with the Ubiquity Appcelerator product set, an optional add-on to SIP A/S. Appcelerator provides a unique SOA including a Service Oriented Object Framework (SOOF) with reusable code components. And by using Eclipse, developers get an enterprise or "IT" perspective on real-time telecommunications applications software development, which makes it easier to develop converged applications that work with carrier services or within the enterprise. Applications created with UDS can be built tested and debugged in Ubiquity Developer Edition, so you can test and debug right on your desktop.


Genesys Readies Enterprises for IMS World with Release of 7.2
By Robert Liu

With its release of the newest version of its flagship contact center software, Genesys Telecommunications (news - alert) Laboratories is betting that enterprises will eventually fix their IP communications switching infrastructure on the Session Initiation Protocol (SIP) and incorporate elements of the IP Multimedia Subsystems (IMS) architecture. The centerpieces of the Genesys 7.2 release are the Genesys Customer Interaction Management (CIM) platform, a framework that adapts to any media channel, and the Genesys SIP Server, which acts as a software-based automatic call distributor (ACD) to direct calls to up to 30,000 IP agents through multimodal customer contact channel options that include voice, e-mail or chat.

This new standards-based approach to call control and signaling means Genesys 7.2 can freely communicate with media gateways and softswitches without having to conform to a myriad of different variations on the Media Gateway Control Protocol (MGCP). In addition, the company has incorporated the Real-time Transport Protocol (RTP) to manage the media elements within the network, as outlined by the IMS specifications. Meanwhile, the Customer Interaction Management (CIM) platform features an Open Media software component that tightly integrates IP into key elements of the suite, such as the Genesys Voice Platform to support voice selfservice, video and voice-enabled applications. As such,

Genesys 7.2 is able to incorporate business process routing (BPR) capabilities by applying intelligent real-time routing, tracking, reporting and management capabilities to work items and interactions over any channel inside the contact center and across the enterprise.


DiVitas Networks Announces Interoperability with Trapeze Networks' WLAN

DiVitas Networks, (news - alert) a provider of seamless, unified mobility solutions for enterprise networks, announced that it has successfully tested the interoperability of its platform with Trapeze Networks, (news - alert) the provider of the wireless LAN (WLAN) Mobility System, MP-372 access point, and new MX-200 and MX-216 series of switches to enable enterprise campus mobility - and the industry's first demonstrable Fixed Mobile Convergence for WiFi and Cellular.

DiVitas and Trapeze successfully tested a VoIP-over-WiFi application, handing off VoIP calls from the WLAN to the cellular network and back for seamless roaming over the best available network. Leveraging cellular networks, Internet/Wi-Fi technologies, and dual-mode handsets, DiVitas delivers mobile voice and text on a single platform, both within the enterprise and outside it. The DiVitas solution also serves as an application platform for enhanced communications capabilities at extremely attractive cost points. Above all, the DiVitas solution is controlled by the enterprise. Administrators can now centrally manage mobile communications independent of the network that delivers them.


RADVISION Releases New SIP Toolkit With Java APIs

RADVISION, (news - alert) a provider of multimedia conferencing and communications platforms, has released a Java version of its Session Initiation Protocol (SIP) toolkit for real-time communication applications, such as VoIP and IM. The new toolkit significantly shortens development cycles, thereby enabling even faster time-to-market for a variety of new and innovative services. Java programming language, which is rapidly gaining popularity among developers, can be used to write software on one platform and run it on another, create programs that run within Web browsers, develop server-side applications, and write applications for cell phones and other consumer communication devices. Java SIP is a fundamental building block for Web-based endpoint-oriented communication applications.

The Java SIP Toolkit can be used to develop clickto- talk applications, as well as a variety of applications for web call centers, and soft phones for laptops and PDAs. It also can be used as a major building block for innovative and time-critical advanced telecom services on the server-side. Similar to all of RADVISION's SIP toolkits, the Java SIP toolkit addresses carrier-side applications for softswitches and application servers. The Java SIP Toolkit supports operator-grade loading of busy hour call attempts (BHCAs) and has enhanced carriergrade features, including advanced SIP signaling functionality. The small footprint allows efficient operation in SIP terminals and IP handsets, as well as network-side implementations.


UPC Broadband Signs One Millionth VoIP Customer
By Erik Linask

UPC Broadband, (news - alert) the European broadband division of Liberty Global, Inc., whose networks pass more than 11 million homes and businesses in Western Europe is using solutions from both Nortel and ARRIS, as well as several others, in growing its European network. In fact, over the past 15 months, UPC's telephony deployment has doubled and it has now signed its one millionth VoIP customer. UPC is also one of the first cable operators to deploy SIP (Session Initiation Protocol) -based VoIP services, laying the foundation for new, feature-rich voice and multimedia services, like presence, instant messaging, video and mobility, and taking the first step towards the converged services.

"As consumer and business demand for multimedia services grows, operators want scalable, reliable and open VoIP solutions that deliver services in a timely, reliable and cost-efficient manner without abandoning existing network investment," said Steve Pusey, executive vice president, Nortel and president Nortel Eurasia.

UPC provides high quality VoIP service that is attractively priced to offer a real alternative for consumers to their existing telephone service. Its deployment of VoIP-based equipment - like softswitches and gateways from Nortel and customer premise equipment from ARRIS - allows UPC to provide enhanced triple play services to its customers. There are plans to develop a VoIP/SIP-based converged product portfolio over a single IP network to both fixed and wireless networks. The goal is to introduce integrated fixed/mobile services in Switzerland and other key markets based on these SIP softswitches.


The New PacketCable: SIP ... and More

The new specs announced this week by CableLabs Inc. (news - alert) for its PacketCable system bring the cable industry closer to adopting SIP, but still refrain from specifying SIP as the only option for cable operators deploying VoIP services. The current signaling system used by cable operators, called NCS, does not allow the portability that SIP supports. Unlike SIP, customers cannot plug their IP phones or terminal adaptors into the Internet anywhere and get service. Therefore, the services closely resemble the PSTN, with a few extras, such as Web interfaces to configure features.

The changes to PacketCable signal that the leading operators in the cable industry will continue their relatively conservative approach to VoIP in the short term, but the specs also open the door for future services, such as WiFi VoIP, and combinations of voice and other media, that take advantage of the portability of SIP and its support for multimedia. Also included are select aspects of IMS, chosen for their relevance to the cable industry.

Cable operators now have a mechanism to do both voice and video telephony. They can combine services across platforms, for example, sending caller ID to a set-top to display on a television. There are also ways to integrate fixed and mobile services. "It gives operators ways to integrate across all those services," Miller says. The specs developed by CableLabs, the technology development center for the North American cable industry, also deal with some of the knotty deployment issues of SIP VoIP, namely security and network address translation (NAT) firewall traversal. Other new features address QoS and device provisioning.


App Server Vendors Prep for Imminent Combination of IT, Telecom
By Robert Liu

Application server vendors are accelerating their rollout plans to prepare for the imminent blending of information technology with the telecom world, brought on by the influx of IP Multimedia Subsystem (IMS)-based architecture. Oracle (quote - news - alert) has outlined its roadmap to deliver a standards-based Service Delivery Platform (SDP) for the telecommunications industry to transition from current silo-based network investments into a service-oriented architecture (SOA) to help reduce time and costs to deploy nextgeneration communication Internet Protocol (IP) voice and data services. Oracle SDP plans to extend its Oracle Fusion Middleware platform with technologies obtained through its recent acquisition of HotSip, adding elements like Session Initiation Protocol (SIP), Presence Server, Proxy Registrar, and Location components.

Oracle's announcement underscores its desire to buy its way into the lucrative telecom market, which is witnessing an economic upswing as standards like IMS and Linux enable the convergence of delivery platforms. "IT-standards-based service delivery platforms offer compelling value to operators as the basis for developing interactive, media-rich, next generation data services," said Philip Marshall, director, Wireless/Mobile Technologies, Yankee Group. "Service delivery platforms that enable immediate ROI through out-of-the-box services and integration with [operational and business support systems] through standard interfaces are likely to be particularly compelling. Vendors who are able to provide a broad portfolio of products that are stable, mature and carrier-grade will have a definite edge over the competition."


Cisco Expands Unified IP Phone Portfolio
By Sandra M. Gustavsen

Cisco Systems (quote - news - alert) has introduced a number of new or enhanced IP telephone models, including the 7911G (Global) and 7941G-GE and 7961G-GE (Global and Gigabit Ethernet) models. All support IEEE power and advanced SIP capabilities. Recently, Cisco also announced Cisco Unified CallManager 5.0, a new software version of the company's IP PBX that runs on the Linux operating system and adds native support for SIP (SIP line and trunk support) in addition to Cisco's existing protocol, Skinny Client Control Protocol or SCCP. With native support for SIP, Cisco customers and channel partners will have a choice of endpoint devices from various vendors for a customized solution. Further flexibility is available to Cisco customers since they can choose devices based on SIP or Cisco's SCCP

Cisco's IP Phone 7911G is the company's newest basic IP phone with the same hardware design of Cisco's earlier IP Phone 7912G, including one line and four soft keys for access to popular telephony features and functions. Designed for business employees with low to medium telephone usage such as those in retail, school or manufacturing environments, the 7911G adds enhanced memory, IEEE 802.3af Power over Ethernet and advanced security. XML support enables text-based applications (including double-byte and unicode characters) to be viewable on the monochrome display. The 7911G works with Cisco's CallManager 3.3 software (or higher) and CallManager Express 4.0.


Openwave Joins MobileIGNITE

MobileIGNITE, the leading industry association fostering collaboration to accelerate fixed-mobile convergence (FMC), today announced that Openwave, a leading independent provider of open software products and services, is joining MobileIGNITE. Openwave has agreed to the group's tenets of working on open standards-based interoperability and leveraging SIP (Session Initiation Protocol) with a commitment to IMS (IP Multimedia Subsystem), the industry's leading blueprint for convergence.

MobileIGNITE's Interoperability group is working to define common use cases, interoperability best practices, and test cases from the baseline open standards on which FMC and other IMS-related applications are based. "As the line fades between mobile and VoIP networks, carriers will be able to give consumers adaptive messaging services that are independent of network, device, or protocol used," said Richard Wong, senior vice president, products and solutions, Openwave. "We look forward to becoming active in the MobileIGNITE association to ensure industry interoperability and accelerate development and deployment of converged services."


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