Zoiper WEB introduces communication functionality to any web application and VoIP-enables any website making Zoiper WEB beneficial for any business environment.

Zoiper WEB requires no desktop client-software installations, which have to be stored and configured – it simply eliminates any complexity by providing instant access to any services and products upon website visit or upon first use of a web application. This results in increase of the sales conversion rates, and drastically reduces website, shopping cart, or web application abandonment.

Ingate announces that the Ingate SIParator and SIP Trunking software module are compliant with key SIP trunking and IP telephony PBX and Unified Communication solutions from Avaya. The SIParator helps businesses connect IP-PBXs and Unified Communication solutions to Internet telephony service providers' SIP trunk offerings instead of conventional PRI trunks, giving them both lower cost and more features for their telephony connection. The SIParator also makes it possible for home workers and road warriors to use the company PBX and allows Unified Communication to be global, stretching outside the Local Area Networ. The SIParator and Ingate SIP Trunking software module are now compliance tested by Avaya for compatibility with the Avaya Aura Session Manager 1.1 SIP Trunking, Avaya Aura Communication Manager 5.2 and Avaya Aura SES 4.2. This is in addition to Ingate's existing certification with other Avaya IP telephony solutions.

Doddle allows users to make VoIP calls anywhere in the world via Doddle’s web based SIP phone directly from their webpages. With Doddle’s free online phone service, no registration is required and VoIP is as easy as accessing a webpage. Users just begin using.

Doddle’s web based phone allows anyone to add phone capabilities to their web applications, including websites, portals, social networks and blogs such as Facebook, MySpace, Orkut, Blogger and more, so that web visitors can make calls directly from the webpage. Twitter tweets can truly make sounds with an instant call link.

InCharge Systems released a hosted reference system of its ACerted Trust solution, available immediately for interoperability testing. The reference system, will be demonstrated Thursday, October 29, 2009 at the Illinois Institute of Technology 5th Annual VoIP Conference and Expo.

ACerted Trust is a solution for assuring the identity of end users and their operators that originate SIP requests for voice, video, presence or messaging communication sessions, based on the Internet Engineering Task Force standard RFC 4474, “Enhancements for Authenticated Identity Management in the Session Initiation Protocol”

Smoothstone announced that its SIP Trunking Service is compliant with key Internet protocol telephony solutions from Avaya. Smoothstone's SIP Trunking service helps businesses maximize IP-PBX functionality by accessing the rich feature set of Smoothstone's VoIP services, delivering reliable, secure and cost-effective connections that provide greater redundancy and higher aggregation of voice traffic (compared to T1/PRI), without costly equipment upgrades. The application now is compliance-tested by Avaya for compatibility with Avaya Aura(TM) Communication Manager and Avaya Aura SIP Enablement Services.

Panasonic announces the interoperability of BroadSoft's BroadWorks platform with the KX-TGP500 series of SIP DECT cordless telephones, which offers outstanding voice quality and a range of productivity-boosting capabilities in a full-featured desk phone replacement. DECT 6.0 ensures no interference with wireless networks, and the convenient cordless design eliminates the need to run dedicated network wiring to each employee workstation. It is ideal for home and business environments.

Skype announced the selection of Net-Net OS-Enterprise session border controllers from Acme Packet for its Skype for SIP beta offering. Acme Packet is the leader in session border contro solutions and the sole SBC partner for the Skype for SIP beta program.

Skype’s deployment of Acme Packet’s SBC simplifies the interoperability and feature compatibility of the Skype for SIP beta offering with enterprise IP-PBX equipment and next-generation unified communications platforms which utilize the SIP standard. As a result, Skype for SIP will allow those enterprises with an on-premise IP PBX to take advantage of an innovative IP-enabled communications tool and to benefit from end-to-end interoperability. The implementation of an SBC by Skype as part of the Skype for SIP beta program also enables the delivery of high-quality, real-time interactive communications, while minimizing the exposure to risks for those companies who sign up for the trial.

As the market for SIP trunking heats up, service providers are increasingly turning to Metaswitch Networks. Carriers including Birch, LightBound and Fibernetics are finding that the feature richness, field-proven interoperability, and reliability of the MetaSphere platform enable them to differentiate their offerings and win in this increasingly competitive market.

Smoothstone and FusionStorm announce they have formed a strategic partnership that will enable FusionStorm to provide Smoothstone's voice services and on-demand applications to FusionStorm clients. This new strategic alliance will add Smoothstone's SIP Trunking capabilities to the spectrum of IT services and solutions FusionStorm currently offers to their clients. Now, FusionStorm clients can receive reliable, secure and cost-effective voice services that provide better flexibility in their infrastructure and increased scalability as their businesses grow. Additionally, these voice and network-based offerings are coupled with integrated applications that give clients a deeper level of visibility through real-time call reporting and greater redundancy through enhanced business continuity features and controls.

Junction Networks is broadening the range of its popular OnSIP hosted PBX service beyond voice, adding the unified communications features of instant messaging, presence, and phone status. All four components are presented to OnSIP end users through my.OnSIP, an IM-style interface available from a browser.

Through one window, my.OnSIP shows all users which contacts on their hosted PBX are "present," which are on the phone, and which are free to call with a click on their names. It lets users send calls only to those who can answer, avoiding voice mail and phone tag. And unlike consumer IM services that are often banned from the workplace as note-passing distractions, my.OnSIP's chat is limited to those on the hosted system, even though those coworkers and their extensions may be physically located at multiple sites on different continents or at temporary locations.

Skype announces the beta version of Skype for SIP has been certified as interoperable with the Cisco Unified Communications 500 Series for Small Business. This will enable small and medium-sized businesses who manage their networking and communications needs with this affordable UC solution to communicate more efficiently by directing their outbound calls to mobiles and landlines via Skype, while also allowing them to receive inbound calls from Skype users.

TalkSwitch announces that its VoIP-enabled phone systems have been tested and approved for interoperability with the Titan Communications 'SIP Plus' SIP Trunking Service. Titan's SIP Plus SIP Trunking Service provides all the basics, including unlimited local calling, caller ID, directory listing, 911 service and local phone numbers. Additionally, small business customers benefit from virtually unlimited long distance (2000 minutes per month with every SIP trunk anywhere in the continental United States and Canada), along with virtual numbers, virtual local calling areas and unlimited phone numbers (DID).

Skype and ShoreTel announces interoperability between ShoreTel's UC system and the beta version of Skype for SIP. ShoreTel is the world's first UC vendor to achieve interoperability with Skype for SIP. The nearly 11,000 ShoreTel enterprise customers can now register to try Skype for SIP in beta, giving them the opportunity to save money, save time and stay ahead by integrating Skype into their communications portfolios.

The SIP Forum proudly announces that membership in the association continues to be strong across all categories, including dues paying, corporate Full Membership; Academic Institutions; and individual, “Participant” membership. Currently, the SIP Forum has 48 Full Member companies, five Academic Members, and more than 6,500 individual members from around the world.

Companies that have joined the SIP Forum as Full Members since the association last reported membership details include Bell Canada; 8x8, Inc.; Optimum Lightpath, a division of Cablevision Systems Corporation; Cedar Point Communications; Comcast Cable; Cox Communications; Genband, Inc; NET; Nortel; PAETEC; Samsung Electronics Co., Ltd.; ShoreTel Inc.; and XO Communications.

JAJAH is working with Microsoft to provide SIP Trunking services to Microsoft enterprise customers globally, allowing companies to make high quality voice calls over JAJAH's IP Platform in the cloud, without requiring an infrastructure upgrade.

Enterprises using Microsoft Office Communications Server 2007 Release 2 can now connect directly with JAJAH's global carrier-grade IP network to make high quality voice calls. In addition to the significant cost-benefits, JAJAH's extensive range of calling solutions integrate seamlessly into the enterprise, so voice calls can be made from computers and landlines.