SIPquest Announces Enhanced Version of
Mobile Console 2.0
(news - alert) Mobile Console, a software application residing in mobile handsets, delivers personal command
and control of communications services over WiFi or WiFi and GSM or WiFi and CDMA network interfaces. Its single GUI and support for unified numbering consolidates both WiFi and cellular identities to provide a seamless enduser experience. The “Network Aware” feature detects and recommends network connectivity to allow users to place and receive a call over the best available network, optimizing for lowest cost, highest call quality or user preference. The latest enhancements to Mobile Console include three major features: Instant Messaging, presence, corporate directory. Enterprise workers can use these new features to check the availability of co-workers, sort through the corporate directory, enable three-way or four-way conference calls, sort through call logs of their desktop phone and have the IP PBX deliver calls to a temporary number. According to David Hattey, president and CEO of SIPquest, “Mobile Console 2.0 is a major step toward thefulfillment of our vision of giving corporate user complete personal command and control of their communications capabilities - regardless of whether they are in the enterprise or a public setting. With these new features, enterprise users can expect enormous communication productivity gains anywhere, anytime.”
http://www.sipquest.com

RADVISION Announces New Release of Its SIP Toolkit
RADVISION, (news - alert) a provider of multimedia conferencing and communications platforms, announced the immediate availability of version 4.0 of its SIP toolkit for real-time communication applications such as VoIP IM. “The entire telecommunications market is in the midst of a major migration from traditional TDM (timedivision multiplexing) technology to SIP technology,” said Adi Paz, Senior Director of Product Management and Marketing for RADVISION’s Technology Business Unit. “RADVISION’s SIP Toolkit 4.0 is a key enabler for this migration.” Specifically developed to address carrier-side applications, such as soft switches, gateways, andapplication servers, the RADVISION SIP Toolkit 4.0 can support millions of busy hour call attempts (BHCAs) and has been enhanced with carrier-grade features including a reliable and effective Stream Control Transmission Protocol (SCTP) layer and automatic translation between phone numbers and SIP addresses (ENUM). Using advanced design techniques, RADVISION has optimized the SIP Toolkit 4.0 for a small footprint, allowing it to serve efficiently in SIP terminals and IP handsets as well as network-side implementations.
http://www.radvision.com

Citel Unveils SIP-Based Handset Gateways
By Mae Kowalke
Citel (news - alert) released a new series of 12- and 24- port SIP-based handset gateways. The new products allow users to take advantage of feature-rich IP Centrex or SIP-based IP PBX services while still retaining their existing desktop handsets, wiring infrastructure, and data network. Citel’s new product line is designed to help companies of all sizes accelerate their migration to IP telephony services and applications, for both hosted and onsite configurations. The company’s handset gateways connect directly to existing business-grade PBX handsets, which in turn connect to an Internet Telephony Service Provider or IP PBX over Ethernet using SIP.
“Some of the primary inhibitors to a forklift VoIP upgrade are business disruption and total deployment cost of new LAN infrastructure and IP phones,” said Michael Robinson, CEO of Citel, in a press release. “Handset Gateways enable businesses of all sizes to migrate quickly to IP telephony and benefit from next generation feature functionality without buying or installing expensive new phones or performing costly LAN infrastructure upgrades.”
http://www.citel.com

Covergence Delivers First Anti-Virus Solution for SIP-based Apps
Covergence, (news - alert) a provider of unified security and management solutions for applications and services based on the Session Initiation Protocol (SIP), announced that it has integrated technologies from McAfee, Inc. to ship the first complete anti-virus solution for SIP-based applications and services. McAfee’s awardwinning anti-virus solution is now tightly integrated into the Covergence Eclipse family of SIP application management products enabling enterprises to benefit from multi-modal, real-time collaboration while protecting them from the dangers of SIP-borne malware.The proliferation of applications and services based on SIP and its extensions for instant messaging and presence (SIMPLE) represents a new challenge for enterprises concerned with protecting themselves against dangerous malware. One of SIP’s greatest strengths is that it supports multi-modal collaboration applications that can support several different forms of real-time communication simultaneously. While this offers tremendous business benefit, it also creates serious security vulnerabilities. McAfee Anti-Virus for Eclipse addresses these vulnerabilities and prevents the propagation of viruses, trojans, worms, and other forms of malicious software via SIP-based applications. The Covergence solution is unique in its ability to detect and to block malware embedded directly in the SIP signaling stream or in a SIP-associated content stream. The solution can also stop the spread of malware that attempts to propagate by embedding links to infected files in the bodies of instant messages. “The explosion of SIP-based applications and services has created a whole new set of security concerns for enterprises,” said Bob O’Neil, president and CEO, Covergence. “With McAfee Anti-Virus for Eclipse, organizations can finally have the peace of mind that comes with knowing their SIP-based solutions
are protected.”
http://www.covergence.com

Nortel Enables New Rural Market SIP Access Systems
By Johanne Torres
Carriers looking into offering triple play services will have the option of tapping telecom technology provider Nortel to enable new SIP fiber to the home (FTTH) and converged IP access systems that will help them deliver broadband data, video and a set of voice features to residential and business customers in rural areas. Nortel’s (quote - news - alert) new systems are enabled by the company’s DMS-10 softswitch interoperating withproducts from Allied Telesyn and Pannaway Technologies. The bundle gives rural carriers access to a new portfolio of services such as VoIP with E911 lifeline calling, enhanced IP video and video on demand. “As we evolve to VoIP, it is absolutely critical that we have the ability to provide advanced calling and safety features such as E911 lifeline support as well as new applications we can run on an application server. Interoperability between Nortel’s DMS-10 and Pannaway’s SCN will allow us to offer premier video and data services along with support for emerging applications such as HDTV and video on demand,” Joseph Gottwald, CO engineer and ISP manager, Empire Telephone.
http://www.nortel.com

SIP Trunking Simplifies Conversion to VoIP, Saves Money
By Mae Kowalke
A recent survey by CTIA found that 60% of SMBs plan to increase their use of converged voice and data communication solutions during the next 18 months — an additional 13% had already implemented a converged solution. One reason for the growth in converged communications is the advent of SIP trunking — a way for SMBs to cut down considerably on the startup and management costs of switching to VoIP phone service, according to Todd Landry, Senior Vice President at Sphere Communications. (news - alert) Landry explained that until SIP-trunking entered the picture, using VoIP meant purchasing and managing an expensive and complicated system of gateways to convert voice signal from digital to analog and back again — often involving several such “jumps” during the call’s journey from sender to receiver. “The jumps cost money, and they degrade the quality of the voice call,” he noted. SIP trunking eliminates the need for gateways by using software to manage a company’s VoIP service, and by utilizing the carrier’s already-existing network of gateways. When both caller and receiver are set up with internet telephony systems, SIP trunking is especially efficient, because it allows a call to travel the entire way as a digital signal.
http://www.spherecom.com

Sonus Networks Bridges Gap Between 2G and 3G Networks
Sonus Networks, Inc. (news - alert) announced that it has enhanced its IMS-ready SMARRT Wireless solution to include support for legacy wireless service protocols. The Sonus solution empowers wireless carriers with the ability to offer a bundle of services that include their existing services in combination with cuttingedge next-generation SIP (Session Initiation Protocol)- based services. By providing support for key application protocols, such as Customized Applications for Mobile Network Enhanced Logic (CAMEL), into its Gateway Mobile Switching Center (GMSC) solution, Sonus enables wireless network operators to differentiate their service with new SIP-based applications while leveraging their existing CAMEL infrastructure for services such as prepaid calling plans. Sonus’ solution provides an architecture capable of delivering legacy applications over a more cost-efficient IP-based network, while at the same time, providing a foundation for full scale migration to the IP Multimedia Subsystem (IMS). Built on Sonus’ award winning IMS-ready architecture, Sonus’ SMARRT Wireless solution is a family of packet-based wireless switching solutions that allows carriers to cost-effectively deploy and operate wireless networks, increase network capacity to accommodate growing subscriber traffic, and build new revenue streams and customer loyalty through the delivery of packet-based enhanced services. In addition, Sonus’ IMSready SMARRT Wireless solution provides a streamlined migration path towards Fixed-Mobile Convergence (FMC), the integration of wireline and wireless technologies and services to create a single telecommunications network foundation
http://www.sonusnet.com

SIPphone Announces Funding To Expand PC-Based VOIP Service to WiFi And Mobile Phones
SIPphone Inc, (news - alert) developers of the popular VOIP software Gizmo Project have secured $6.0 million in funding to expand their SIP standard service to non-PC devices, such as adapters, routers, WiFi handsets, and dual-mode mobile phones. “The release of Gizmo Project has driven over 400,000 registered users to our PC-based service, but our goals go beyond free calls between PCs,” says SIPphone CEO, Michael Robertson. “We want to seamlessly link the internet and the traditional calling world of landlines and mobile phones and that’s becoming possible as low cost SIP-based devices such as dual-mode mobile phones and WiFi phones become a reality.” SIPphone will use the investment to expand their VOIP platform to portable devices that do not require PCs to make or receive calls and promote adoption of the open standard auto-provisioning system plug-n-dial. The San Diego-based headquarters is expected to triple in size over the next year with hiring primarily in business development and engineering.
http://www.sipphone.com

BEA Named Spec Lead for JSR 289, SIP Servlets
BEA Systems (news - alert) has been named the specification lead for Java Specification Request (JSR) 289 for Session Initiation Protocol (SIP) servlets. BEA is continuing to help develop and evolve the Java-based SIP and Telecom Web Services technologies, which can help facilitate the rapid creation and deployment of next generation IP-based multimedia services. BEA has been named the specification lead for the next release of the SIP Servlet API, version 1.1, as specified in Java Specification Request (JSR) 289. With SIP Servlet 1.1, BEA is set to lead the effort within the Java Community Process (JCP) — a standards organization for community development of Java technology specifications — to help build upon the foundation established in SIP Servlet API 1.0 (JSR 116), and enhance key areas of SIP application development and deployment. JSR 289 is slated to address enhanced control of application composition, invocation and convergence at runtime and deployment time. These enhancements are designed to help further accelerate the creation and deployment of highly rich, flexible, and dynamic IP-based real-time, interactive, multimedia services based on SIP. “The SIP servlet specification is an important part of the overall SIP technology portfolio. Having APIs like SIP servlets is the key to bringing the web development community to telecom,” said Jonathan Rosenberg, a Cisco Fellow at Cisco Systems. “That community is essential for building the kind of innovative services that SIP was designed to support.”
http://www.bea.com

Ixia Tests Triple-Play Networks
By Johanne Torres
Calabasas, Calif.-based Ixia (news - alert) introduced its IxLoad 3.00, a test app for assessing the performance of triple play networks. Ixia’s IxLoad triple play test system is capable of emulating a large number of subscribers, as well as the associated content servers. The system works in conjunction with Ixia’s hardware platform to achieve the scale needed for real-world test scenarios. The new IxLoad release includes features that enable realistic triple play testing. For example, IxLoad can accurately emulate millions of IPTV Broadcast Video and Video on Demand (VOD) subscribers and the millions of video streams they are viewing. Such large scale emulations enable the performance characterization of key IPTV elements, including video servers, multicast routers, and the video delivery network. The required protocols associated with these services, such as IGMP, RTSP and MPEG, are fully supported in IxLoad 3.00. IxLoad tests the performance of Session Initiation Protocol (SIP)-based devices by emulating SIP callers and callees. The system also supports a set of Internet data protocols including HTTP, SSL, SMTP and POP, as well as Distributed Denial of Service (DDoS) attacks and vulnerability attacks. Other new IxLoad 3.00 features include a built-in network impairment to emulate realistic network conditions; over 600 results metrics; capture and replay of custom protocol traffic; rapid adjustment of traffic and support for DHCP, LDAP, IMAP protocols; IPv6 support for HTTP and FTP protocols.
http://www.ixicom.com

More Than a PBX
MKC Networks’ (news - alert) next generation 7000 Communication Server (CS) is a software-only Session Initiation Protocol (SIP) Application Server that pushes the limits of conventional telephony. “Many have a hard time believing that standard computing platforms offer more value and performance than a purpose built product. There are significant benefits to this approach, including the speed of feature deployment, scalability, cost and ease of use,” notes Ben Morris, Sales and Product Marketing Manager of MKC Networks Corporation in Ontario, Canada. The 7000 CS software on CD-ROM loads onto a customer-supplied hardware platform. The combination of open standards, a Linux operating system and SIP call control enables some exciting voice and data communications features such as desk and mobile phone ”twinning,” IP trunking directly to a carrier, phone relocation both internal and external to the corporate LAN, network alarms and flexible software-based licensing. SIP compatibility means the solution is not tied to a single vendor so that customers can choose from “best of breed” solutions, including phones (desktop and soft phones), unified messaging servers, call center applications, conference and collaboration solutions and vertical-specific applications. The cost-effective 7000 CS is designed for a variety of markets, including both small businesses and large carriers since it can be easily scaled with the addition of software licenses. The 7000 CS is available in various license packs (10, 20, 50, 100, and 1,000) on one server, or up to 5,000 users using several servers. MKC Networks reports one carrier in Toronto, Canada is running the 7000 CS software on a powerful Dual Xeon processor, hosting 5,000 users. The 7000 CS can be deployed in global network configurations via SIP-compliant carrierclass gateways.
http://www.mkcnetworks.com

iTalkBB Chooses Covergence
iTalk Broadband Corporation (iTalkBB) (news - alert) has chosen Covergence’s Eclipse solution to address the challenge of reliably scaling thousands of concurrent VoIP connections at the subscriber edge. The subscriber edge is the boundary point at which service providers connect subscribers to their service. At the SIP subscriber edge hundreds-of-thousands to millions of real-time connections need to be managed and secured as they traverse the service boundary. In order to continue to grow its business while maintaining its position as a tier one VoIP service provider, iTalkBB sought out Covergence to ensure service scalability and
performance at the subscriber edge. Eclipse is the first product specifically designed to secure and manage large numbers of active endpoints. The remarkable performance and capacity of Eclipse begins with its hardware and software architecture which were purpose-built for the demands of the SIP edge. Eclipse’s flexible, distributed design supports integrated clustering and load balancing to enable administrators to scale performance to carrier-class levels simply by adding Eclipse appliances to the cluster. With Covergence’s (news - alert) Eclipse(TM) solution, iTalkBB can now offer subscribers the predictable performance and quality they have come to expect from their existing PSTN service.
http://www.covergence.com
http://www.italkbb.com

Verso Rolls Out SIP Gateways with Integrated IP Routers
By Laura Stotler
Verso Technologies, Inc. (news - alert) has launched its Session Initiation Protocol (SIP) Gateways, which include a VoIP gateway as well as an integrated, full-featured IP router. The new models support between one and six T1/E1 digital interfaces and offer a call capacity between 15 and 120 voice channels. The new gateways offer IP and TDM connectivity for carrier and enterprise markets, and provide advanced gateway technology with an integrated access router for seamless connectivity on wireless, wireline and IP networks. They also work on interconnect legacy systems and the PSTN without sacrificing QoS, reliability or performance. “This is a strong niche solution that addresses the market demands of carriers and enterprises alike, designed to work seamlessly on wireless, wireline, satellite and IP networks to interconnect legacy systems and the PSTN,” said Monty Bannerman, chief executive officer, Verso Technologies. “The demand and growth for SIP gateways is a thriving business for carriers and enterprises alike, it is one opportunity that Verso and our award winning NetPerformer solution are prepared to capture.”
http://www.verso.com

Mitel Extends IP Interoperability with 3300 ICP
Mitel (news - alert) delivers native support of SIP and SIP trunking with its latest release of the Mitel 3300 IP Communications Platform (ICP). Release 7 offers a host of new benefits including reduced access costs; easier implementation and facility redundancy; duplicated facilities to multiple providers; enhanced ability to increase (or decrease) capacity and improved disaster recovery capability. Mitel has completed or is in the process of completing SIP interoperability testing with a number of key industry softswitch vendors, including Asterix, Broadsoft, Cedarpoint, Cisco Call Manager, Cirpak, Huawei, Ingate, Newport Networks, Nortel, Radvision, Samsung, Siemens, ZTE, and more. Mitel enables users to fully leverage the power and feature functionality of the 3300 ICP through generic SIP endpoints and innovative use of existing Group and HotDesking functionality. Additionally, SIP connectivity to Microsoft Communicator for softphone functionality and third party call control inter-working with Microsoft Live Communications Server is enabled using SIP / CSTA / XML. Microsoft's interoperability with Mitel is based on SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE). The companies are using the SIP and SIMPLE protocols to enable the federation of presence information and instant messaging (IM) that can be encrypted and authenticated. SIP and SIMPLE provide a rich set of solutions for computer and telephone interaction.
http://www.mitel.com