CommuniGate Systems Announces Availability of SIP Farm to Deliver Industry’s First Highly Resilient Carrier-class and Scalable VoIP Framework

CommuniGate Systems is launching its new SIP Farm, patent-pending technology on the CommuniGate Pro Dynamic Cluster application server platform. The technology enables large scale deployments of SIP-based VoIP to telco-level capacities. The multi-node, all-active SIP Farm technology makes it possible to host millions of subscribers with either a consolidated cluster or distributed geographic placement of cluster members while providing 99.999% uptime. CommuniGate Pro 5.0 SIP Farm removes all barriers to adoption of SIP as a standard, making it possible to scale to tens of millions of subscribers with redundant signaling. The complex nature of real-time communications has strained the database-centric architectures of the past. Carriers have found it hard to manage systems not designed for the sheer load that IM presence status information and multiple endpoints per user on the system.

“We’re introducing technology today that eliminates the pains of scaling SIP technology for sites with tens of millions of subscribers. The lack of redundancy or susceptibility to signaling failure is the result of limited architectural approaches in most SIP systems,” stated Thom O’Connor, Director of Product Architecture, CommuniGate Systems. “With CommuniGate Pro SIP Farm, redundancy and capacity is expanded in one or even multiple locations acting as a single cluster, with the ability to add or remove nodes from the cluster and providing immediate re-allocation of sessions in the event of system failure.

www.communigate.com

SIPfoundry Releases Version 3.0 OF SIPxPBX

SIPfoundry, a non-profit organization for the development of open source Session Initiation Protocol (SIP)-based communication solutions, released its latest open source PBX: sipXpbx. This release delivers a fundamentally new approach to addressing the real-time communications needs within the enterprise market. This new release of a complete SIP PBX that handles real time communications is available to users as a single file download. Building on the sipXpbx platform, developers at SIPfoundry have expanded its functionality and capabilities to include: an advanced scalable architecture, a presence server, support for multiple and nested auto attendants, advanced call control, forwarding identification support, and interoperability support tested with dozens of SIP-compliant phones and gateways.

In addition, the system configuration and management tool was re-architected to make it more robust, efficient, and extensible to support larger enterprise installations and make it easier for third-party manufacturers to integrate features and functionality. Manufacturers of endpoints can now easily add support for their products in just a few hours, to enable complete provisioning and management of their devices. In this release, complete provisioning and manageability of AudioCodes gateways and Polycom, SNOM, and Grandstream phones were delivered.

Interpeak and TeleSoft to Offer SIPNET

Interpeak and TeleSoft International announced a collaboration focused on the promotion of combined technologies for the consumer electronics, mobile handset, and mobile multimedia markets. The companies will create a new product for the embedded market, SIPNET, which will combine TeleSoft’s CompactSIP and Interpeak’s IPNET. This joint effort leverages each company’s strengths and expertise: Interpeak’s networking, security and wireless technologies and TeleSoft’s flexible, full-featured and compact SIP. The horizontal platform will have potential across a range of embedded products — including set-top boxes, handsets, consumer electronics, PDAs, and other multimedia devices — and applications, including instant messaging and both wireless and wired VoIP. As networks converge on an IP infrastructure, due to the widespread availability of broadband access and advent of rich multimedia-based communications, SIP will become the primary protocol technology for signaling, session capability negotiations, and the establishment of multimedia-based communications. SIPNET will stand ready to meet the increased SIP traffic, with the requisite secure and mobile components as well.

www.interpeak.com
www.telesoft.com

Antepo Integrates Presence and Telephony With VoIP Upgrade for OPN System

Antepo, Inc. unveiled the first preview of its upgrade for OPN System, code-named “Rivoli” which adds voice and video capabilities to its award-winning EIM and Presence Management server. The Rivoli release further establishes OPN System’s native support for SIP (Session Initiation Protocol) to offer broad integration with SIP-enabled services, devices, and applications. With OPN System Rivoli, enterprises and service providers can seamlessly integrate IP-based telephony capabilities with enterprise-wide Presence management across a range of multimedia communications systems and services. The module implements a complete SIP registrar and proxy server, so users can leverage a range of SIP and SIMPLE-based endpoints to make multimedia calls over the Internet — and to generate and manage Presence through OPN System. For service providers, Rivoli works with CounterPath’s award-winning eyeBeam client to offer a robust, scalable Presence engine to deliver voice-centric communication offerings. For enterprises, Rivoli offers the first and only alternative to Microsoft Live Communications Server for the company-wide integration of IP-based telephony systems, and the addition of advanced Presence management capabilities to a full range of voice, video, and IM communications.

www.antepo.com

Brekeke Software Announces Coming Release of OnDO PBX v1.5 With ARS Failover

Brekeke Software, Inc. announced the coming release of their OnDO PBX version 1.5, an enhanced IP-PBX software telephony system, which now supports Automatic Route Selection (ARS) failover. With ARS failover, OnDO PBX seeks an alternate route if the specified route is unavailable, and makes outgoing calls via the best route based on the situation. The benefit of ARS failover is improved efficiency and reliability of your telephony system. Additional benefits include increased ease in creating connections between OnDO PBX and SIP products and services. Setting up routing rules enable ARS failover to select from all available routes. ARS Failover also allows for SIP products and services which are unavailable, to be considered inactivated for a specified period. The route is re-activated based on automatic recovery settings or by manual re-activation through the administration tool. “Future opportunities for VoIP technology using SIP continue to grow. For example, if a natural disaster disabled PSTN lines, VoIP communication by satellite has been recognized as a method for re-establishing communication systems,” states Shin Yamade, President and CEO for Brekeke Software.

www.brekeke.com

Excel’s Integrated Signaling and Media Gateway Enhanced to Allow Any-to-Any Voice Network Connectivity

Excel Switching Corporation announced the availability of SIP and ISDN functionality in an enhanced version of its popular IMG 1010 integrated signaling and media gateway. For service providers that are planning to bridge TDM and IP networks, or connect two IP networks, the IMG 1010 provides new levels of density, integration, and transcoding on a future-proof platform that can evolve as services evolve. The IMG 1010 supports SIP, H.323, SS7 and ISDN simultaneously, all-in-one compact 1U design that can accommodate up to 768 channels, for the ultimate in flexibility. As a result, the IMG 1010 sets a new standard for simplicity, versatility and cost-effectiveness in the media gateway market, offering a solution that empowers service providers to seamlessly transition from first generation, network-based transport-oriented services to the subscriber-driven multimedia SIP applications found in next-generation IP-based services.

www.excelswitching.com

TotallyFreeConferenceCalls Selects Vapps SIP-Based Platform to Deliver Next Generation Conferencing Services

Vapps, a leading global provider of audio conferencing systems, announced that TotallyFreeConferenceCalls has chosen the Conference Bridge 1000 (CB1000) platform as the foundation for delivering next-generation conferencing services. Together with MetaSwitch’s fault-tolerant Class 4/5 softswitch architecture for bridging legacy circuit switched and IP Multimedia Subsystem (IMS) networks, TotallyFreeConferenceCalls can now integrate service across multiple network types to reach new markets and expand its customer base. A VoIP-native IMS application server network element, the CB1000 is a SIP-enabled, carrier-grade conferencing platform that delivers robust, seamless reservation-less conference calls on both legacy and IP-based telecom systems. CB1000 supports up to 18,000 total conference participants in multiple simultaneous conferences and offers both standard and advanced calling features including: participant lists, volume control, mute, multiple languages, call detail records and web-based on-demand feature control. “CB1000 was selected based on the platform’s rich features, SIP interoperability and carrier grade reliability,” said Todd Zweig, General Manager of TotallyFreeConferenceCalls. “By allowing us to bridge traditional PSTN calls with the ubiquity and flexibility of IP communications, we are able to offer customers exciting new services with lower associated operation costs.”

www.vapps.com

TelTel Launches High-Quality PC to Any Phone, Anywhere Calling Via TelTel’s Managed Peer-to-Peer SIP Backbone TelTel, a provider of SIP-based global Internet telephony, today announced the launch of TelTel-Out, which delivers high-quality PC to any phone, anywhere calling, and TelTel Call-Forwarding, a new call forwarding service that allows forwarding of incoming calls to any PSTN number or SIP URL. TelTel-Out and TelTel Call-Forwarding are both now available to TelTel’s over 1.3 million users and through service providers via its popular SIP Virtual Network Operator (SVNO) program. TelTel-Out provides better quality calls than traditional PSTN line calls. The new services use TelTel’s managed SIP-based peer-to-peer backbone to optimize call routing and deliver crystal clear calls to virtually anywhere in the world. TelTel’s popular SIP Virtual Network Operator (SVNO) program allows service providers and enterprises to immediately offer VoIP services like TelTel-Out and TelTel Call-Forwarding that are second to none in quality of service. This program has been successfully deployed worldwide and has proven to be a strong revenue generator for both enterprises and service providers.

www.teltel.com

Linksys Announces Sip-Based IP PBX, Desktop Phones, and Gateway for Internet Telephony Service Providers

Linksys, a Division of Cisco Systems, Inc., the recognized leading provider of voice, wireless and networking hardware for the consumer, Small Office/Home Office (SOHO), and small business markets, today announced a new line of SIP-based telephony products for Internet Telephony Service Providers (ITSPs) targeting large residential, SOHO, and very small business customers. The new line of IP communication solutions includes an IP PBX/Key system, a wide range of IP desktop phones and an Analog Gateway for connection to the Public Switched Telephone Network. Used together with an ITSP voice service, they provide a complete IP telephony system for up to 16 users.

“With the new Linksys SIP-based IP communication offerings, ITSPs can offer residential and small businesses a voice service with many of the features found in large business voice IP networks, such as multi-line service, music on hold, auto attendant, and more at a more affordable price,” said Jan Fandrianto, vice president of voice engineering at Linksys. “The new IP PBX and IP phones bundled with a service provider offering will make the deployment of voice networks easy to install and simple to use at a price small businesses can afford.”

This new solution will complement the recently announced Linksys One solution. Linksys One is an ideal solution for small business with 5-100 users needing a complete communications solution that addresses voice, video, and applications. The LVS series was developed to address the residential and very small business of 1-4 users that may grow to 16.

www.linksys.com

Cedar Point Expands Safari C3 SIP Capabilities to Enable Deployment of Advanced Voice Features

Cedar Point Communications, the leader in integrated packet-based voice switching technologies for the cable industry, today announced new capabilities of its SAFARI C3 Multimedia Switching System that will enable cable system operators to deploy Session Initiation Protocol (SIP)-based multi-vendor advanced voice communication features. The expanded functionality will enable interoperability between SAFARI C3 and a wide range of feature servers, allowing cable system operators to leverage such third-party applications as messaging, IVR and call attendant. “Competition among traditional wireline carriers, third-party VoIP services and cellular providers has created an increasingly cluttered telephony landscape,” said Dave Spear, executive vice president, strategy and market development for Cedar Point Communications. “By implementing SIP triggers to support third-party call control, we will allow our customers to provide real differentiators in their service offerings, ranging from day-to- day voice features to fixed-mobile convergence.” “One of cable’s voice strengths today is its unique ability to combine both carrier-class PacketCable services that equal or exceed the quality and reliability of incumbent providers and SIP-based features that provide operators and providers with unparalleled flexibility,” said Andy Paff, president and CEO of Cedar Point. “As we and the industry move toward an IP Multimedia Subsystem (IMS) architecture in the future, operators will be able not only to implement third party services, but to combine those services to create entirely new offerings. The expansion of SIP capabilities allows us to ease the migration to IMS networks.”

www.cedarpointcom.com

SIP-Based Software Phone Brings Theora Open Source Video Codec to VoIP Community, Developers

Ecotronics Ventures LLC announces a new release of its Kapanga Softphone, now supporting Xiph.org Foundation’s Theora, the open source video codec based upon On2 Technologies’ VP3 codec. Kapanga Softphone extends the set of tools available to the VoIP community by integrating Theora into its SIP-based video capabilities. Kapanga Softphone is the premier VoIP softphone that integrates voice, video, and fax over IP into one software phone. Using the Session Initiation Protocol (SIP), Kapanga Softphone interacts with any VoIP infrastructure, such as VoIP phone service providers, PBX systems for enterprise and small businesses, and other VoIP-based systems. “Since our launch this year, many Kapanga users have used our SIP client to extend their open source PBX deployments”, said Martin Cadirola, President and VP of Business Development for Ecotronics. “We believe we can help speed up the development of open source PBX video functionality by providing a Theora-capable VoIP-based software phone”, he added Kapanga also features other open source audio codecs, like Global IP Sound’s iLBC and Xiph Foundation’s Speex.

www.kapanga.net