New entry-level VoIP telephone snom 300 works against eavesdroppers, data theft and spam

March 31, 2006 by Garrett Smith

Category: Press Release, German Article
February 1st, 2006

snom technology AG to show the new SIP-based VoIP telephone snom 300 for secure use in SMEs and home offices at the CeBIT and Spring VON 2006

CeBIT 2006, Hanover, 9th-15th March 2006 (Hall 12, Stand D57)

Spring VON 2006, San Jose (CA), 14th-17th March 2006 (Stand 1534)

The new starter model of the snom VoIP telephone family, the snom 300, is all about user friendliness and security. Six free, user- or administrator-configurable (or carrier-preconfigurable) function keys can easily be allocated to security-related menu functions, or assigned to multiple lines. The snom 300 comes factory-equipped so that two of its six programmable keys can be configured as line appearances, and upgrades are available from snom that lets you configure (up to) all six function keys this way – flexible enough to suit the needs of every user.

“Due to its rapid growth, Internet Telephony is becoming increasingly enticing for illegal eavesdroppers, data thieves and advertising calls; so with the snom 300 we have placed particular emphasis on security features like SRTP and SIPS,” stresses sales and marketing director Dr. Michael Knieling. “It is precisely in the price segment around 100 Euros that most VoIP telephones are lacking effective protection mechanisms like these.”

In addition, the snom 300 offers all necessary office functionality, e.g. choice of trunk line, status indicator, group lines, busy option and call-pickup. High audio quality, ease of use and interoperability make the snom 300 highly suitable for SMEs, home offices, private users or ISP applications.

Most important features:

Two-line display (2 x 16 characters)

27 keys, 7 LEDs

6 programmable function keys

2 Ethernet ports

2 multi-line registration, or option of 6

Headset connection

SIP RFC3261

Security: SIPS/SRTP

STUN, ENUM, NAT, ICE

Codecs: G.711, G.729A, G.723.1, G.722, G.726, GSM

Photographs to download: http://www.snom.com/press_download.html

About snom:

snom is recognized for its high quality Voice over IP (VoIP) business telephones. Our customers are companies who use interoperable technologies for their enterprise communications. Together with our partners, snom offers solutions that reduce the total cost of ownership, maximize the overall return on investment into telephony infrastructure, and make our customers independent from a single vendor. Founded in 1996 and based in Berlin, the company was amongst the first to adopt the session initiation protocol (SIP), which today is recognized as “the” future telecommunication standard. snom is privately held and incorporated under German law.

press contacts:

snom technology AG

Maria Schnake

Gradestr. 46

D-12347 Berlin

Tel: +49 30 39833-103

Fax: +49 30 39833-111

[email protected]

www.snom.com

Maja Schneider

PR für die Informationstechnologie

Tel: +49 30 79708771

Fax: +49 30 79708987

[email protected]

www.it-publicrelations.de

USA

HowPhonesWork.com

John Jainschigg

3943 47th Street

Sunnyside, NY 11104

Tel: 718-701-4106

Mobile: 917-405-3116

Fax: 718-504-6125

[email protected]

www.howphoneswork.com

More from: VoIP Hardware Garrett Smith

Switchvox Launches PBX Solution for the SMB Budget at Spring VON 2006

March 23, 2006 by Garrett Smith

SAN JOSE, Calif.–(BUSINESS WIRE)–March 14, 2006–Small- and medium-sized businesses (SMBs) now have all of the advanced features of an enterprise-level phone system at a fraction of the cost, with the launch of Switchvox SMB. The new, full-featured, easy-to-use IP PBX solution starts at $2,495.

Switchvox SMB is the only IP PBX to offer a Call Event Notification Application Programming Interface (API) for companies to integrate their IP phone system with other business applications or databases, such as contact directories or order tracking software. The product also includes Switchboard 2.0, the industry’s most advanced real-time call control panel that leverages the benefits of unified voice and data while providing greater visibility into the phone system. For example, users can easily direct calls to other extensions in real time, view if other employees are currently on a call, and even display relevant information from other applications as calls are received.

Switchvox is built from open source software and uses open standards. It works with all Session Initiation Protocol (SIP)-compatible hardware and software phones as well as standard analog handsets, rather than typical PBXs that rely on proprietary telephones. Voice-over-IP providers can be used to send calls over the Internet worldwide and directly to remote corporate offices by peering Switchvox systems using the SIP or Inter-Asterisk eXchange (IAX) protocols.

In addition to supporting the latest technology, Switchvox can also connect to existing phone lines, whether those lines are analog or T1. Other advanced features in Switchvox VoIP system include:

Outlook Integration – TAPI plug-in for Microsoft Outlook allows users to click to call any of their contacts in Outlook
Firedialer – Firefox users can click to call any phone number on any Web page
Universal Call Recording – Flexible call recording feature gives administrators the capability to define “jobs” that record all calls or only the calls they need
Live Call Monitoring – Allows users with permission to listen in to live calls on the system
Call Pickup – Enables users to answer another extension’s ringing phone from their own phone
On-Phone Status Indicators – Presence indicators on the phone’s display allow users to see when others are on a call

“Before we deployed Switchvox, we were using a traditional phone system with limited features. We often missed calls while we were out of the office and needed a way to make sure we were accessible to customers,” said Jason Wieland, CTO of Mission Bay Vacation Rentals. “Switchvox allows us to be away from the office when showing properties and continue to conduct business as usual, at a fraction of the price that we would have paid for any other PBX system.”

“Our customers and prospects have requested a more feature-rich IP PBX phone system to handle the growing needs of their businesses,” said Joshua Stephens, CEO of Switchvox. “With the launch of Switchvox SMB, companies will have all of the features they require in a phone system, without the high price tag. Growing organizations now have a solution that is easy to install and manage, while giving them the appearance of a larger company.”

Analyst firm The Radicati Group estimates that by 2009, 74% of all corporate telephony lines will be IP lines. Worldwide revenue for hybrid and pure IP PBXs is expected to grow from approximately $1.5 bln in 2005, to $9.9 bln in 2009.

Switchvox products are sold as turnkey solutions that include the server hardware and pre-installed Switchvox software. Switchvox SMB is available immediately starting at $2,495 and the Switchvox SOHO product is available for starting at $995.

About Switchvox

Switchvox is a leading provider of PBX and VoIP phone systems for small- to medium-sized businesses (SMBs). The company’s business VoIP SOHO product enables small and home offices to easily and affordably create and manage their phone system, using traditional analog lines, as well as VoIP services. The Switchvox SMB product is a full-featured, more advanced IP PBX system for larger organizations. Based on Linux and other open source software, Switchvox has created software products that fit business and consumer needs.

Linksys Announces SIP Based VoIP PBX SPA-9000

March 9, 2006 by Garrett Smith

Linksys Announces SIP-Based IP PBX, Desktop Phones, and Gateway For Internet Telephony Service Providers

IP PBX and Phones Enable Service Providers to Deploy Residential and Small Business Phone Systems with Big Business Features and Functionality
IRVINE, Calif. – January 5, 2006 – 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.
According to AMI research, there are more than 35 million small businesses world wide. 56% of these businesses have 1 to 4 users, and another 11.8% have 5 to 19 users. The new Linksys IP telephony products target this market segment. ITSPs focusing on the higher-end of the residential market, or the lower end of small business market can deploy services on this platform to help customers save money by using the Internet to make telephone calls.

“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 VoIP phone system. 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.

The Linksys SIP-Based IP Products include:

Linksys IP PBX: SPA9000
The Linksys IP PBX lies at the heart of the telephony solution, delivering advanced, multi-line features commonly found in enterprise telephone networks. With the Auto Attendant feature, small businesses with up to 16 users can customize their voice network to direct calls to the right person or department and program the system to allow on and off business hour greetings. A promotional message can be created to greet or provide information to callers placed on hold, or pipe in the music of choice. The system also allows users to park calls, fax over the Internet and use the intercom for paging. There are over one hundred telephony features built-in. Out of the box, the SPA9000 supports up to four (4) users. Via a software license key upgrade, up to sixteen (16) users can be supported. Details can found at www.linksys.com.

Software Functionality

Configuration Server
SIP Proxy and Registrar – Up to 16 Users
Application Server
Media Proxy
TA via 2 Onboard RJ11 FXS Ports
Hardware Interface Functionality
- Two Phone Ports:
- Analog Telephone
- Fax Machine
- Two Ethernet Ports:
- LAN Connectivity
- Call Traffic and Signaling Routing
  - IP PBX Features
  - Automated Attendant
  - Reduce Call Load On Receptionist (No Receptionist)
  - Professional Welcome for All Callers
  - Single Number Access to All Employees
  - Guaranteed Call Completion
  - Fast, Easy Set-Up and Maintenance
  - Automatic Call Distribution / Routing
  - Bridged / Shared Line Call Appearance
  - Call Transfer
  - Call Forwarding
  - Local and Corporate Directory
  - Call Pickup
  - Group Paging
  - Intercom
  - Call Hunt Groups
  - Direct Inward Dialing (DID) and Voice Mail Integration with ITSP
  - Music on Hold
  - Do Not Disturb
  - Three Party Conference Calling
  - More features can be found at www.linksys.com

Linksys IP Desktop Phones: SPA901, SPA921, SPA922, SPA941, SPA942
This new line of affordable IP desktop telephones provides a wide variety of phone options for small businesses. Choose phones with one or two Ethernet ports, 1, 2, or 4 extensions, optional Power over Ethernet functionality, or the cost-conscious, durable no-display model. These telephones work with all SIP-based IP Telephony solutions, but when used with the SPA9000, the phones install in minutes, automatically configuring themselves with business features set by the Service Provider. With an optional Wireless-G Phone Bridge (WPB54G) users can install phones in hard-to-reach places without running cables. The line of Linksys VoIP phones can also be used in conjunction with leading SIP application server platforms to provide an IP Centrex service model for service providers looking to provide PBX/Key system features to small businesses.

Models

SPA901: Single Line, 1 RJ45, No Screen
SPA921: Single Line, 1 RJ45, Monochrome Screen
SPA922: 2 Lines, 1 RJ45, PoE, Monochrome Screen
SPA941: 2 Lines, 1 RJ45, Upgradeable to 4 Line Appearances, Monochrome Screen
SPA942: 2 Lines, 2 RJ45, Upgradeable to 4 Line Appearances, PoE, Monochrome Screen

Linksys Analog PSTN Gateway + Phone Adapter: SPA3000

If power is lost to the SPA9000 or Internet service is down, calls can be redirected to a traditional PSTN carrier using the SPA9000 Analog Gateway so service continues without interruption. The SPA3000 can be remotely provisioned and supports dynamic, in-service software upgrades.

One Line FXO Port Provides Call Routing To and From PSTN
Back Up in Case of Internet Access Problems
Back up for Emergency E911 Service
Transparent Migration from Legacy PSTN Service
Optional Routing of Local, Toll Free Calls to PSTN
One FXS Port Provides Connection for Analog Phone or Fax Machine
Service Providers Supporting the SPA9000 Platform
ITSPs currently developing services using the SPA9000 Platform include:
- North America EMEA APAC
- VoicePulse Telio (Norway) Engin (Australia)
- RNK Telecom

About Linksys
Founded in 1988, Linksys, a division of Cisco Systems, Inc., (NASDAQ: CSCO) is the recognized global leader in Wireless, VoIP and Ethernet networking for consumer, SOHO and small business users. Linksys is dedicated to making networking easy and affordable for its customers, offering innovative, award-winning products that seamlessly integrate with a variety of devices and applications. Linksys provides award-winning product support to its customers. For more information, visit www.linksys.com.

Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and products are trademarks or registered trademarks of their respective holders. Copyright © 2006 Cisco Systems, Inc. All rights reserved.

Content from Linksys Press Release – January 2006

More from: VoIP Hardware Garrett Smith

Sangoma Enters Analog Telephony Market with Release of Remora A200 Analog Cards

by Garrett Smith

Sangoma has recently announced their A200/REMORA FXO/FXS Analog Telephony support System. It is now available at VoIPSupply.com

The A200 and REMORA system together comprise the FXO/FXS version of Sangoma’s range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic.

The A200 and REMORA cards support up to a total of 24 FXO and/or FXS connections. A single PCI slot host connection for all ports ensures common synchronous clocking for all channels. The base AFT architecture is shared with Sangoma’s A101, A200 and A104 cards ensuring proven 3.3v/5v, high performance PCI compatibility.

Like all the Sangoma AFT Series, the A200 and REMORA system is field upgradeable to take advantage of the hardware and software improvements as they become available. Also, the A201 supports Sangoma’s echo cancellation and voice enhancement board so that hardware voice enhancement is available for analog as well as TDM voice.

Architecture
The A200 consists of a REMORA daughterboard mounted on the AFT PCI card . The REMORA card has two sockets each of which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively.

Up to four additional REMORA daughterboards can be mounted in empty slot positions beside the A201 assembly, connected to the A201 by a backplane bus connector.

Technical Specifications

Support for the Asterisk™ , Yate™ , OPAL™ PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications.
Single synchronous PCI interface for all 24 FXO/FXS ports.
Four RJ11 4 ports per REMORA card.
Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis.
Short 2U compatible mounting clips included for installation in 2U rackmount servers.
32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention.
Autosense compatibility with 5v and 3.3v PCI busses.
Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper interrupt sharing.
Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available.
Power: 800mA peak, operational 300mA max at +3.3v or 5v.
Temperature range: 0 – 50C.
Optional DSP card on the A200d
G.168-2002 echo cancellation in hardware
1024 taps/128ms tail per channel on all channel densities
DTMF decoding and tone recognition
Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction.

Wiring Connections
The A200 and Remora cards incorporate four, 4 pin RJ11 narrow jacks such as used in telephone handsets. Each A200/Remora is shipped with four 2M cables terminating in a narrow RJ11/4 plug at one end and a telephone-standard RJ11/6 plug at the other.

For those who need to hard wire the A200 system, Sangoma has a kit available consisting of 24 RJ11/4 plugs and a crimping tool.

(1) Sangoma Website, A200 / Remora Datasheet, 2005, http://www.sangoma.com/datasheets/p_a200-specs (Nov 2005).

Hardware Echo Cancellation – Sangoma and Digium

by Garrett Smith

In recent months, both Digium and Sangoma have released T1/E1 Interface cards for use with Asterisk Open Source PBX that provide hardware Echo Cancellation.

Echo can negatively impact QOS (Quality of Service) on a VOIP call and lead to an unsatisfactory user experience. There are two sources of echo on voice communication networks. The first, and most common cause of echo is impedance mismatches anywhere in the circuit-switched phone network. When telco cabling is spliced or terminated in connectors, or in the conversion of 4-wire phone circuits to 2 wires, a discontinuity occurs that causes an impedance mismatch on the phone circuit. The greater the extent of impedance mismatch, the more leakage of transmit audio on the receive side of the line when it terminates on a 2-wire phone circuit or device. The main cause of this impedance mismatch is the 2 to 4 wire hybrid that converts a 4-wire audio path to a local 2-wire loop. This tip-and-ring loop is the most-used type of telephone connection made to subscribers from your telephone company central office, and directly drives common analog telephone devices.

Acoustic coupling between the microphone and speaker of a telephone device leads to Acoustic echo, the second most common cause of echo on voice communication networks. Received caller audio can often “leak” to the microphone in cheaper speakerphones and hands-free cellular phones, because of sound pressure from a speaker or earpiece device

This inherent echo led to the development of Echo Cancellers, in order to improve the quality of voice communications. Echo cancellers are very complex digital signal processing devices, and the algorithms that drive them are produced by a handful of technically proficient companies. Naturally, as telecommunications evolves from the legacy PSTN network to the VOIP network, echo cancellation is still a very necessary component for ensuring QOS.

Digium offers hardware echo cancellation on their TE406P and TE411P telephony boards, as well as their new TDM2400 series full length analog PCI cards. Digium’s hardware echo cancellation provides 64ms across 32 channels; however, when it scales over 32 channels it is reduce to 16ms per channel across all channels.

Sangoma offers hardware echo cancellation on their A104D Quad T1 card, as well as on their as yet unreleased series of analog telephony boards. Unlike Digium where available ms of echo cancellation decreases as active channel density increases, Sangoma hardware echo cancellation provides a full 128ms of echo cancellation on all channels, regardless of density.

By providing hardware echo cancellation which extends to a full 128ms on all channels at full load, Sangoma provides a more robust compensation for echo which is more effective under extreme conditions. The echo tail length represents how long the effect of an echo extends after the time of the echo source. If a ping is transmitted at time zero, 16ms (16 ms = 128 taps at 8 samples per millisecond) of tail will deal with any echo from that ping that occurs within 16 ms. The echo itself dies away in about 8ms, so as long as the delay between the ping and the start of the echo is less than 8ms, then the echo canceller will work well. The problem arises when the signals get delayed. For instance traveling through one Telco switch adds about 5ms of two-way delay, so it is easy to see how a bit of extra switching could delay the echo right out of an echo canceller’s
range.

The above image illustrates the echo from a “ping” on a real line where there are
switching delays. In cases of extreme echo as pictured above, normal 128 tap echo cancellation would be ineffective at cancelling the echo present on the line.

Software echo cancellation works quite well for electrical echoes that are within the 16ms/128 tap range. Where hardware echo cancellation really has benefit is where the delays are longer. The Sangoma echo canceller is a carrier-grade device with 1024 taps (128ms) of tail to handle the most severe echo problems.

FCC Deadline for VOIP Provider E-911 Compliance Passes

by Garrett Smith

Providers of Voice over IP (VOIP) services may be restricted from enrolling new customers after the FCC specified date for E-911 compliance is past.

The FCC had previously imposed a 120 day window for VOIP providers to deploy reliable, enhanced 911 services in all their service areas by Monday, November 28, 2005.

Since the deadline passed, little public disclosure of E-911 compliance has been offered by the FCC or the affected VOIP providers. It had been previously estimated by the VON Coalition, a VOIP industry consortium, that about two-thirds of VOIP users would have access to E-911 services by the deadline.

Linksys Announces WBP54G 802.11G WIFI “Dongle”

by Garrett Smith

Linksys has a cool new product on the market that should appeal to owners of the popular Linksys/Sipura line of Analog Adapters and SIP phones. The WBP54G adds 802.11G Wireless capabilities to any compatible Linksys/Sipura device, eliminating the need to run RJ45 network cable to network such devices.

Now you can locate a Linksys Phone Adapter or SIP Phone almost anywhere, without the cost and hassle of running network cables. The WBP54G was specially designed to convert a SIP phone or adapter into a wireless device, so it can connect to your home network without an Ethernet cable. This lets you put your phone where it’s most convenient for you, and not be constrained to the area around your Internet connection.

“To make installation even more convenient, the Wireless-G Bridge shares electrical power with the Phone Adapter, so only one AC Adapter (and wall plug) are needed. To get connected, just plug your existing Phone Adapter’s power jack to the Wireless-G Bridge, and the Bridge’s attached power and data cables to the Phone adapter. The included Setup Wizard makes it easy to configure the Bridge to your wireless network’s settings. To protect your data and privacy, all wireless transmissions can be encrypted with WEP or industrial-strength Wi-Fi Protected Access (WPA/WPA2) security.

More from: VoIP Hardware Garrett Smith

New SIP PBX Appliance from D-Link – DVX-1000

by Garrett Smith

D-Link has developed a SIP based PBX appliance called the DVX-1000

The DVX-1000 fulfills many of the phone system requirements of the SMB market, including popular features such as call forwarding, call hold, find me-follow me, and voicemail. Inbound calls are routed through the integrated auto-attendant and hunt groups to assist callers to their destinations. The DVX-1000 utilizes standard phone lines via an external phone line gateway or cost effective Internet Telephony
services.

One DVX-1000 VoIP PBX appliance can support up to 25 extensions, which can be located in any physical/geographic location provided that location has Internet access. Multiple units can be peered to increase the number of extensions or unite a company that has many locations under a single PBX system. Additional extensions are added via license codes that are obtained via your reseller or directly from D-Link.

The DVX-1000 IP PBX is fully user configurable via a user friendly web configuration tool. The administrator assigns each extension a profile of telephony features, which allows the best match for a users job function. Each user can fine-tune their assigned profile via the web to match their daily business schedule.

In the world of traditional PSTN telecom, conferencing is typically an expensive external hardware or service. The DVX-1000 includes a phone conferencing bridge, which eliminates the need for an external conferencing server or service provider, and adds distinct value to the D-Link offering. Users are able to schedule and invite parties to conferences via a simple web configuration tool. Conference Notifications are sent out by e-mail, which includes the conference phone number and access codes.

The DVX-1000 uses advanced security features to protect your voice network from unauthorized access. To prevent hackers from breaching the system, the DVX-1000 uses MD5 SIP authentication encryption encoder software. The DVX-1000 also includes an integrated firewall for intrusion detection and protection against denial of service attacks.

The DVX-1000 features a fanless solid-state design offering years of non-stop operation. The compact housing can be easily fastened to the wall of your distribution closet or stacked with your existing Ethernet switches or PSTN Gateways. The DVX-1000 is designed with dual processors for supporting up to 25 simultaneous calls. Its class leading performance allows a 1-to-1 extension to phone line mapping, allowing it to scale with your business.

More from: VoIP Hardware Garrett Smith

Digium Ships New TDM2400 Series Full Length Analog PCI Cards

by Garrett Smith

Digium has begun shipping their new TDM2400 Series Full Length Analog PCI Cards to VoIP Supply.

The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium’s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system.

Starting with a base PCB Board that is available with or without Echo Cancellation, the TDM2400 can be configured with FXS or FXO Resources, added in 4 port increments, up to a maximum of 24 total ports. This allows for a wide variety of FXS/FXO combinations, and in many cases eliminates the need for an external gateway or channel bank altogether.

The Digium TDM2400 Series boards are designed for use with Asterisk Open Source PBX

UTStarcom F3000 – First 2nd Generation WIFI Phone?

by Garrett Smith

UTStarcom has been one of several successful entrants into the WIFI VoIP handset space in recent months. Their initial product, the F1000, set the preliminary standard in terms of usability, feature set and price point. Other vendors including Hitachi-Cable and Zyxel have also brought wireless SIP handsets to market based on the 802.11XX WIFI standard.

The initial crop of wireless VOIP phones all seemed to have their own individual strengths and weaknesses. With the upcoming release of the F3000, UTStarcom seems to have built upon the successful foundation established with the F1000, and have set about addressing some of the limitations and inadequacies of first generation WIFI VOIP handsets.

The UTStarcom F3000 has added support for the current encumbent 802.11G Wireless standard.

The UTstarcom F3000 promises increased security with the addition of WPA encryption, which utilizes the temporal key integrity protocol (TKIP). TKIP utilizes a hashing algorithm to scramble the keys and, by adding an integrity-checking feature, ensures that the keys have not been tampered with.

User authentication, which is generally missing in WEP, through the extensible authentication protocol (EAP). WEP regulates access to a wireless network based on a computers hardware-specific MAC address, which is relatively simple to be sniffed out and stolen. EAP is built on a more secure public-key encryption system to ensure that only authorized network users can access the network.

The preliminary spec we have seen for the UTStarcom F3000 promises to address one of the major factors which has limited the mass-adoption potential of first generation Wireless VoIP handsets, namely Handover/Roaming between different AP and SSID. The ability to “roam” between various access points and SSID within a WIFI network, without losing call connectivity of SIP registration in the process, will seriously bolster the case for WIFI VOIP adoption.

Finally, some aesthetic enhancements are evident in the new F3000, including a larger, color LCD screen and a clamshell form factor typical of current mobile/cellular/GSM phones.

More from: VoIP Hardware Garrett Smith

New entry-level VoIP telephone snom 300 works against eavesdroppers, data theft and spam

Switchvox Launches PBX Solution for the SMB Budget at Spring VON 2006

Linksys Announces SIP Based VoIP PBX SPA-9000

Content from Linksys Press Release – January 2006

Sangoma Enters Analog Telephony Market with Release of Remora A200 Analog Cards

(1) Sangoma Website, A200 / Remora Datasheet, 2005, http://www.sangoma.com/datasheets/p_a200-specs (Nov 2005).

Hardware Echo Cancellation – Sangoma and Digium

FCC Deadline for VOIP Provider E-911 Compliance Passes

Linksys Announces WBP54G 802.11G WIFI “Dongle”

New SIP PBX Appliance from D-Link – DVX-1000

Digium Ships New TDM2400 Series Full Length Analog PCI Cards

UTStarcom F3000 – First 2nd Generation WIFI Phone?

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