Tag Archives: SIP

Nortel SIP-Based VoIP Selected by Adelphia

February 23, 2005
nortelNortel’s PacketCable VoIP and Global Services will support Adelphia’s residential IP telephony.

Nortel (NYSE:NT) announced the selection of its PacketCable VoIP and Global professional services by the cable company Adelphia Communications (OTC:ADELQ) to deliver telephony services to residential customers in selected Adelphia markets.

Adelphia is deploying Nortel’s Communication Server 2000-Compact, a superclass softswitch that uses SIP to provide direct packet connectivity to other softswitches.

SIP will be used to allow the CS 2000-Compact to cost effectively connect to PSTN and also to manage voice and video sessions across Adelphia’s packet network.

Nortel Global Services will plan, build, operate and maintain Adelphia’s VoIP network, for which Nortel is acting as the network integrator to bring together solutions from multiple vendors.

Among the required solutions are the CMTS, MTA, flow-through provisioning, and interconnect components. Nortel will also operate and monitor the network with on-site resources, and act as the single point of contact for trouble resolution.

Nokia: New SIP App Server for Fixed-Mobile Convergence

February 11, 2005

 

nokiaA SIP application server will converge legacy voice and SIP-driven VoIP.

Nokia Corporation announced the launch next week of its Telecommunications Application Server (TAS) which will allow mobile operators to deliver standard mobile telephony services to customers independent of fixed VoIP or mobile cellular access.

These features would include Caller ID and SMS, as well as Intelligent Network features such as Prepaid. A part of the Nokia IMS solution portfolio, the Nokia TAS is built upon the 3GPP Rel 4 compliant Nokia MSC Server system.

Its functionality can be added to any existing Nokia MSC Server with a software upgrade, providing a cost-efficient implementation of VoIP that meets carrier-grade requirements.

Nokia also announced the second quarter release of its Presence Solution 2.0, adding SIP support which allows operators to apply Presence to SIP-based applications, such as its Push-to-talk-overCellular (PoC). Then PoC users will be able to detect the presence of another mobile before initiating the PoC call.

Blackberry Partners with Nortel, 3Com for VoIP over WiFi

February 9, 2005

 

blackberryThe BlackBerry 7270 supports IP telephony to extend enterprise desktop phone functionality to a BlackBerry wireless handheld.

Research in Motion Limited has partnered separately with both Nortel Networks and 3Com Corporation to deliver SIP-enabled VoIP over Wi-Fi LANs to a new Blackberry handheld.

RIM and Nortel are focusing on interoperability and integration between RIM’s BlackBerry enterprise platform and Nortel’s MCS 1500 multimedia communications server, while RIM and 3Com are integrating 3Com’s IP PBX and wireless WLAN switch with RIM’s BlackBerry wireless solution for WLAN networks.

The objective in both cases is to allow seamless roaming of office-based VoIP calls within internal WiFi domains, top enahance the mobility of enterprise voice and data.

Both approaches rely on standards-based interoperability based on SIP. The Internet orientation of SIP readily allows always-on BlackBerry applications such as email, VoIP, browser, organizer, push-based services, security and back-end integration to be tied in to existing office PBX functionality.

The products are being demonstrated now, with general release expected soon.

M5T Releases SAFE SIP Proxy Stack and RTP for Enhanced IP Security

February 8, 2005

 

m5tM5T has developed a complete suite of secure software solutions for multimedia over wired and wireless IP networks.

M5T, a spinoff from Mediatrix Telecom, has released its SAFE SIP Proxy stack version 4.0 as well as its SAFE SRTP (Secure Real-Time Protocol) stack, for enhanced security in IP networks.

The new version 4.0 of M5T SAFE SIP Proxy supports a set of modular services used for building SIP devices, servers and applications, and further strengthens the M5T security line that includes M5T SAFE SIP UA (including back-to-back features), M5T SAFE RTP, and M5T SAFE STUN.

The new product, M5T SAFE SRTP stack, is based on RFC3711 and provides secure transport of multimedia packets, and also addresses media encryption for devices. It can be used with M5T SAFE RTP or integrated with any RTP stack implementation. It uses AES encryption as well as message authentication and integrity checks.

M5T is a wholly owned entity of Media5 Corporation, which also owns Mediatrix Telecom from which M5T was spun off with several IP Telephony pioneers among its senior engineers.

It develops middleware components based on a common SIP framework layer, which forms the foundation for developing secure modular IP communications solutions.

SIPconnect Interface Specification Launched by Cbeyond and Others

February 8, 2005

 

SIPconnectRelease of a new draft “SIPconnect Interface Specification” provides standards-based direct IP peering between IP PBXs and VoIP service provider networks.

Cbeyond Communications and five other IP telephony vendors, Avaya, BroadSoft, Centrepoint Technologies, Cisco Systems, and Mitel, have developed a draft best practices specification to guide interoperability between IP PBXs and VoIP service provider networks.

The rapid deployment of IP PBXs in businesses and the transition to VoIP by service providers has created an opportunity for direct IP peering between IP PBXs and VoIP-enabled service providers which would lower costs, reduce voice latency, and provide for end-to-end SIP signaling for PSTN termination.

The newly released, publicly available SIPconnect Interface Specification leverages the SIP and other IETF VoIP protocols to provide the consistent, industry standards-based approach to interconnection essential to the future of packet-based communications.

Cbeyond Communications, Avaya, BroadSoft, Centrepoint Technologies, Cisco Systems, and Mitel have all indicated their intention to develop products and services that comply with the new draft specification, to the extent that any of their current product lines do not already do so.

They state that SIPconnect specifies a reference architecture, required protocols and features, and implementation rules necessary for seamless peering. It defines a clear and consistent architecture that enables IP PBX vendors and service providers to rapidly and cost effectively deploy VoIP services. They invite the rest of the industry to join and follow them.

VoIP, Inc, Provisions DID Numbers in Real Time for Service Providers

January 12, 2005

 

voipincThe availability of immediate provisioning of Direct Inward Dialing (DID) numbers from VoIP, Inc, for locations around the country will save service providers the expense of having to inventory large supplies of these numbers in advance.

VoIP Inc (OTCBB:VOII), through its subsidiary VoIP Americas, has enhanced its flagship product VoIPDID to include the ability to order and provision DID numbers in real time, eliminating the costly provisioning delays and expense facing many VoIP service providers.

VoIPDID allows next-generation service providers to quickly obtain US local DID numbers which can be terminated on their VoIP gateways anywhere in the world, mitigating the need to establish multiple points of presence in the US, and wait at the convenience of Tier One carriers to provide them with numbers.

The numbers are ordered and provisioned in real time via VoIP America’s web portal and application interface, avoiding lead times that sometimes exceed 90 days. An even tighter integration via XML allows DID numbers to be displayed and provisioned to customers during an online subscription process, further reducing installation costs.

VoIP, Inc, is a solution provider of high-quality VoIP for residential and business customers, and also manufactures products and delivers services, ranging from subscriber-based VoIP, to SIP-based infrastructure, to broadband customer premise equipment and wireless broadband technology, for ISPs, telecom service providers, and cable operators in various countries.

SIPfoundry Deploys Three New Open Source SIP-Compliant Technologies

January 10, 2005

 

Three new open source software solutions fully compliant with the IETF’s SIP standards will accelerate the adoption of the SIP protocol.

SIPfoundry, Inc has announced sipX – the SIP PBX for Linux, reSIProcate – the Reference SIP Stack, and the SIP Interoperability Project, three projects to serve as industry catalysts for the widespread adoption of SIP as the definitive IP communications infrastructure.

sipX, the SIP PBX for Linux, is a 100 percent SIP, 100 percent open source, PBX, voicemail, auto-attendant and SIP proxy, fully interoperable with SIP-compliant media gateways and phones. It is immediately downloadable and is fully manageable via a web-browser interface.

reSIProcate is a SIP stack, unencumbered by any one company’s commercial agenda, that provides the a fully standards compliant implementation of all key IETF RFCs relating to SIP, including enhanced security features, NAT Traversal, Instant Messaging, and Presence.

The SIP Forum Test Framework (SFTF), designed and written by the SIP Forum and hosted by SIPfoundry as the SFTF SIP Interoperability Project, is a growing body of open source test suites that help SIP product users and vendors ensure standards conformance and interoperability – helping to eliminate SIP islands.

SIPfoundry is dedicated to creating software solutions that implement IETF SIP-related specifications, giving end-users and developers one-stop access to a wide variety of SIP application software, tools and resources, all available under open source licenses that allow easy commercial use.

i2Telecom Initiates Private-Label VoIP Push

January 7, 2005

 

i2Telecom launches its Virtual VoIP Network Operator program in an agreement with Clari-net International to resell i2’s VoIP service, augmented by Clari-net’s own local and long distance carrier agreements.

i2Telecom International, Inc (OTCBB:ITUI) announced a global business development partnership with Clari-net International as part of its program to accelerate and extend the penetration of i2’s VoIP service worldwide, and enhance the sale of its platform products.

Clari-net will resell i2Telecom’s InternetTalker MG-3 VoIP microgateways across Southeast Asia, the United Kingdom and elsewhere. As part of the agreement, Clari-net has placed an initial order for 1,000 microgateways. Clari-net will develop its own custom products and rate plans.

By standardizing on i2Telecom’s InternetTalker MG-3 VoIP microgateway and back office systems, Clari-net and other VVNO plan partners will leverage their existing base of international wholesale partners to build share as the VoIP service market develops in their service territories.

Clari-net currently operates in nine countries on three continents, and plans to launch a mass market VoIP service in Australia by the end of Q1 2005 to complement its existing VoIP services in Italy, Hungary, Peru and Costa Rica.

i2Telecom’s proprietary technology platform is built to SIP, and its revenue includes prepayments from sales of InternetTalker integrated access devices, recurring monthly subscriptions, call minute termination fees, and original equipment manufacturer royalties.

Net2Phone Local Number Availability Exceeds 80% of US, Also Offers Canada and UK

December 29, 2004

 

net2phoneLocal phone numbers in more than 80% of the United States, as well as numbers in cities across Canada and local toll-free numbers in the UK, are now available to VoIP service providers worldwide from Net2Phone.

Net2Phone (NASDAQ:NTOP) today announced that the reach of its US local phone number inventory now exceeds 80% of the US, and that it now offers phone numbers in Toronto, Montreal, Calgary, Vancouver, and Hamilton in Canada, and toll free numbers in the United Kingdom, all for use anywhere in the world.

Net2Phone says it is the only provider of both SIP-based and DOCSIS-compliant VoIP solutions, with its SIP-based VoiceLine for DSL and DOCSIS-compliant CableLine for cable service providers. By supporting multiple platforms and architectures, Net2Phone can conform to any operator’s particular marketing, technical, financial and operational needs.

This expansion of number availability enhances service providers’ ability to deliver VoIP service in their regions with local phone numbers, as well as makes more options available to international service providers looking to offer a broader choice of phone numbers.

Net2Phone’s solutions embody features such as call waiting, caller ID and voicemail, and also seamlessly integrate front and back office systems and billing platforms into the operator’s infrastructure, enabling a unified bill for video, high-speed data, and voice services.

Ingate Releases New Tools for SIP Real Time Communications Through Firewalls

December 17,2004

 

Ingate’s new Firewall 1600 and SIParator 60 high-capacity products enable SIP-based VoIP, IM, presence, and other realtime person-to-person IP communications in large enterprises.

Ingate Systems AB announced two new products for organizations with high demands for capacity, throughput and reliability of SIP-based IP communications: the Firewall 1600 and SIParator 60.

SIP (Session Initiation Protocol) traversal of firewalls is an ongoing hurdle for the rapid adoption of this protocol for realtime communications in enterprises, and one which all of Ingate’s security products address.

Both of these products deliver IP-PBX functionality, SIP-enabling technology, and support for Microsoft Office LCS (Live Communications Server) 2005, combined with network security. Both are 1 U high and feature six interfaces, two of which can be used at Gigabit speed, and each supports up to 360 simultaneous RTP sessions (e.g. VoIP calls). Both are managed via an HTML-based graphical user interface.

The Firewall 1600 includes 10 SIP user licenses and five SIP traversal licenses, while the SIParator 60 has 50 SIP user licenses and 25 SIP traversal licenses. The SIP user licenses can be used for the registration of SIP user agents such as phones and soft clients, while the SIP traversal licenses allow one call per license to traverse related firewalls at the same time. Additional licenses are available optionally.

The 1600 and 60 are designed complement an IP-PBX, and support functions such as call transfer, call hold and voicemail. Their compatibility with Office LCS 2005 allow voice, video, IM and presence applications to work outside the LAN.

Ingate Systems is based in Stockholm and Linköping, Sweden, with a wholly-owned subsidiary, Ingate Systems Inc, in Hollis, New Hampshire. Ingate Firewalls solve the NAT traversal issues inherent in using SIP.