Seeking a Level Four Skype Interconnection

Written by Skype Journal on October 10th, 2006 with no comments.
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PacketGen VoIP testing tool

GL Communications has enhanced its VoIP stress testing tool called PacketGen, a software based VoIP testing tool that features real-time VoIP bulk call generation (including SIP signalling & RTP generation), stress testing, and analysis of VoIP networks and equipment.  The latest version, PacketGen 3.0 supports SIP call generation (individual calls as well as bulk calling) and RTP/RTCP traffic generation. With PacketGen you can send/receive voice files, user-defined digits and tones, as well as create noise impairment and real-time voice traffic. TMC Labs has tested their testing products over the years, including their flagship product, the (Digital Central Office Switch Simulator (DCOSS), which we reviewed here and here.

Using PacketGen, calls can also be made to IP phones and to Analog Telephone Adapters (ATAs). PacketGen can be used to test basic functionality and verify proper protocol implementation in SIP based equipment such as SIP phones, as well as Proxy Servers, SIP Registrar servers, as well as PSTN and Media Gateways. One of my favorite features is its integrated impairment functionality. You can generate automatic impairments over the RTP for any (or all) established calls. These impairments include latency (Fixed, Uniform, Nominal), packet loss (Periodic, Random, Burst with burst probability and burst size), and packet effects (Out of order, Duplicate Packets).

Here is a list of their new features:

Continue reading PacketGen VoIP testing tool...

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Tags: bulk call generator, gl communications, sip, testing tool, voip
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Written by VoIP & Gadgets Blog on September 28th, 2006 with no comments.
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With SIP trunking, VoIP ain’t just for residential, my friend

Rich just wrote about an interesting conference session titled SIP Trunking - Realizing Rapid ROI Today, taking place at Internet Telephony Conference & Expo (ITEXPO) in just 2 weeks. Although I ditched my residential Vonage VoIP service, due to price and QoS issues, that doesn't mean I don't whole-heartedly support SIP trunking in the corporate enterprise -- where voice is even more critical. Most corporations have better QoS Internet/IP connectivity than residential Internet offerings, thus SIP trunking with "guaranteed QoS" is a viable option to an expensive voice PRI/T1/E1 line. Some service providers offer voice VPNs for added security. If your attending ITEXPO, definitely check out this session to see how SIP trunking can save on telecom costs.

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Tags: internet telephony, sip, trunking, voip
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Written by VoIP & Gadgets Blog on September 26th, 2006 with no comments.
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Adobe flashes on VoIPifying the web

Written by Skype Journal on September 22nd, 2006 with no comments.
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ADTRAN adds New Multiservice Access Routers to NetVanta family

ADTRAN today announced the introduction of the NetVanta 3400 Series of Multiservice Access Routers. The series delivers up to two T1s of wire-speed performance, even with advanced services like Firewall, Access Control Lists (ACLs) and IPSec Virtual Private Networking (VPN) enabled. I've always been a fan of ADTRAN's "all in one" networking solutions, which often give you not just a router, but PoE, firewall, VPN, built-in SIP server, CompactFlash for backup, and more. Back in May, I wrote about ADTRAN's 7100 IP-PBX, which I thought was one of the most comprehensive, plug-in-play, all-in-one IP-PBXs in the market. Similarly, I was impressed with ADTRAN's Total Access 900, which is also a comprehensive solution.

Well, the new NetVanta 3400 series includes the NetVanta 3448 and 3430, each a modular, 1U-high, rackmountable metal chassis that also follow Adtran's "all in one" model. These products are targeted at the Small and Medium Business (SMB) or enterprise applications requiring high-performance throughput for bandwidth-intensive applications. Each offers a single-slot to house any of the NetVanta Network Interface Modules (NIMs) for WAN access and Dial Backup Interface Modules (DIMs) for disaster recovery applications, two 10/100Base-T Ethernet Local Area Network (LAN) interfaces that allow for broadband backup or DMZ, CompactFlash for configuration backup and storage of multiple firmware revisions, and support for up to 500 simultaneous VPN tunnels. Each is RoHS compliant making it ideal for both domestic and international applications.

The NetVanta 3448 includes the addition of a fully managed, non-blocking, eight-port 802.3af-compliant Power over Ethernet (PoE) Switch delivering a full 15.4 watts of power per port. This allows you to power any PoE device, including security cameras, SIP phones, or wireless access points.

These products also offer a five-year warranty and free firmware upgrades for the life of the unit.

“As the demand for bandwidth-intensive applications continues to grow so does the need for greater throughput and processing speed,” said Rick Schansman, senior vice president and general manager, ADTRAN Enterprise Networks Division. “The NetVanta 3400 Series clearly addresses these needs by allowing businesses to fully utilize a complete arsenal of security and performance features without degrading throughput performance.”

The NetVanta 3430 has a list price of $895, while the NetVanta 3448 has a list price of $1,045. Both models are currently shipping. The Power over Ethernet option for the NetVanta 3448 has a list price of $345 and will be available in Q4. Enhanced VPN capabilities can be activated through a software upgrade for $395.

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Tags: adtran, adtran 3430, adtran 3448, netvanta, sip, voip
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Written by VoIP & Gadgets Blog on September 19th, 2006 with no comments.
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How SIP (Session Initiation Protocol) Works

Note:  Nice little article explaining SIP in a basic sense.  Informative Read

Have you ever wondered why long distance calls cost so much? In part the reason is because telephone lines cost so much. When driving, you might occasionally see a telephone crew maintaining a telephone line, but what you may have never considered is that there are literally thousands of individuals working around the clock to maintain our telephone lines.

 

The telephony system works via a cog and wheel setup. What this means is that every long distance call you make is routed along a telephone wire to a central station, where your voice is routed to another central station, which is finally carried to the person with whom you are trying to communicate. For the call to be maintained, the entire time you are speaking, a space along all the lines in between you and the person you are talking with must be completely devoted to you. Because millions of people are talking at the same time, the little space along the telephone lines becomes rather desired property. And like all things desired, the price is high. Before recent innovations, however, there were no alternatives, so everyone grudgingly paid the often costly long-distance telephone bill.

SIP, or Session Initiation Protocol, has turned the telephony world upside down. Specifically, SIP refers to a protocol that allows computers to talk to each other without going through a central station. Practically, what that means for you and me is that it is no longer necessary to pay for expensive telephone lines to complete our calls. SIP technology is a relatively new development in which calls are made on a peer-to-peer rather than cog and wheel network. What that means, is that you are now able to call people directly from your SIP enabled phone to theirs. This ends up being radically cheaper than the old way of calling.

The SIP system does not require a central computer and operators like the old telephony system did. Rather, your computer, or SIP enabled phone, does all the routing for you.

SIP has been around for a number of years, but only recently has it begun to go mainstream and take off in popularity. This quick increase in interest over SIP is due to companies like Mobalex, who were aware of the fact that over the generations we have come to expect certain tones, buttons, and protocols from our phones. So what they have done is to transpose those functions onto the SIP system. Rather than forcing users to communicate in a completely new way, what these companies have done is to provide a calling experience which from the user’s perspective is completely identical to traditional telephony.

SIP is typically offered in two formats, computer based and hardware based. Computer based SIP is a system that allows you to make calls using your computer as the router and communicating via a headset on your computer. The more practical and popular version, however, actually provides you with new SIP enabled telephone handsets or converts your existing phones to SIP. By eliminating any technical requirements, modern SIP providers have made using the system as easy, or easier, than using a traditional phone. I say easier, because many companies are able to take advantage of the fact that the system is internet based to provide you with some very unique benefits. These include the ability to adjust your plan, change your calling options, and even pay your bill from the same website.

SIP technology is quite revolutionary in the world of communication. By creating a peer-to-peer network, SIP has been able to radically undercut the prices of traditional telephony, take advantage of the Internet, and still maintain the ease of traditional telephony. It is merely a matter of time before we are all using SIP for all of our telephoning needs.

Source: Click Here 

 

Written by Dal on September 13th, 2006 with no comments.
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Truphone SIP client for mobile phones with free calling

Truphone today announced availability of a free beta VoIP software download for Wi-Fi-enabled, mass market mobile handsets. The first download is available for Nokia's latest E-series handsets, with the N-series handsets to follow imminently.  Support for additional Wi-Fi-enabled phones, including those built on the Windows Mobile platform will soon follow.

Calls between 'on net' Truphone handsets are free worldwide. However, in the footsteps of Skype's free U.S./Canada SkypeOut promotion (amongst others), Truphone has a launch offer to the end of the year (Dec 31st 2006) that gives USA users free calls to USA/Canada and UK users will be able to call some two billion landlines in major countries for free. All other Truphone calls from the user's mobile will be charged at low VoIP rates. (e.g. UK 2.7c/min, China 2.8c, Mexico City 3.1c, Japan 4.8c)

Provisioning, installation and use are a breeze. Users send an SMS with the word 'TRU' to a designated number, provisioning takes place 'over the air' and the software installs itself quickly and easily on the mobile handset. Truphone integrates seamlessly with the phone using the existing address book, so there's nothing to re-type. Once set up to use a Wi-Fi hotspot, the handset connects automatically next time the phone is in range.

Dialing is a breeze using this software since it is seemlessly integrated Simply dial or select a contact as you normally would and press the Call (Green) button. Call behaviour is identical to using your handset over your usual GSM/Cell provider. The handset will attempt to make a Truphone call before trying a GSM/Cellular call. If you want to manually select the call type, dial or select a contact as you normally would and select Options>Call>* Call where * = Voice, Video or Internet.

Support for Truphone SMS messaging is imminent, while presence functionality will also follow. The SIP-based software incorporates several leading-edge techniques, including the self-healing, small-footprint AMR codec to ensure optimal call quality even under less than optimal conditions; and, unlike other Wi-Fi applications, Truphone uses smart, low-level coding to minimise battery usage.

James Tagg, Truphone's chief executive, said: "Truphone is the natural evolution of VoIP. First people talked to a PC, then they were tied to fixed-line handsets and now Truphone takes VoIP mobile. Free calling has arrived on mass-market handsets."

To get Truphone:
- Users in the UK should send an SMS with the word TRU to shortcode 60030
- Users in the USA should send an SMS with the word TRU to +44 7624 000 000
- Users elsewhere should send an SMS with the word TRU to +44 7624 000 000

Truephone is certainly leveraging some good open-source stuff, as well as several popular Linux components.

The major telecoms components are:
- OpenSER: Open source version of the SIP Express Router
- Asterisk: Open source SIP PBX

Other components that they use are:
- Apache Tomcat Servlet Container
- MySQL Databases
- Debian Linux Build
- Spring, Hibernate, JSF, Shale - Clay, Acegi for the web site build
- Roller for blogging
- Nagios and Monit for status monitoring
- OTRS for trouble-ticketing
- Jira for bug tracking

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Tags: free calls, mobile phone, nokia.e-series, sip, truphone, voip
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Written by VoIP & Gadgets Blog on September 12th, 2006 with no comments.
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Flextronics Software Systems Ranked Number 1 in Worldwide Market Share for SIP Stacks

 

FSS is the clear leader in the SIP stack market with 32% share, according to VDC’s Telecom Core Infrastructure Market Intelligence Service 2005-2006. SIP is an application layer control protocol that can establish, modify and terminate multimedia sessions or calls.

“SIP is a dynamic and rapidly growing software sector where scalability and interoperability are critical,” said Robert Johnson, Senior Telecom Analyst of Venture Development Corporation. “Hardware manufacturers, software vendors and service providers have increasingly high expectations of companies providing SIP protocol stacks and toolkits. FSS’ top ranking is a very strong statement from the market,” continued Mr. Johnson.

“FSS’ products’ proven ability to reduce time-to-market and their superior scalability, interoperability and quality are the top factors our customers cite when they choose our SIP products,” according to Suresh Kabra, Assistant Vice President and Head of Products of Flextronics Software Systems. “We will continue to accelerate the development of our customers’ products and services through FSS products that include SIP-powered IMS offerings, SIP User Agent Toolkits, our SIP Server Framework, our Back-to-Back User Agent and our SIGCOMP stack,” added Mr. Kabra.

FSS protocol stacks, frameworks and toolkits have been critical to the delivery of products from over 200 Original Equipment Manufacturers (OEMs) worldwide. Flextronics Software Systems offers innovative software products for:

-- wireless, wireline, broadband and satellite infrastructures;
-- carrier-grade and enterprise class communications equipment; and
-- devices, server-side communications applications and the network core.

Source: BWI 

Written by Dal on August 30th, 2006 with no comments.
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Articulation - VoIP on your Palm PDA

Palm users have been clamoring for VoIP on their Palm devices for quite some time. I've mentioned VoIP on Palm (including the Treo) in the past. mobiVoIP was the first to offer VoIP on a Palm device.  However, mobiVoIP only works with their VoIP service plan. Many Palm users simply want a SIP softphone client that works with any SIP-based termination service provider. I also wrote about Talkplus, but they don't support SIP either and they too use their own phone service plan.

Well, today, Hampton Software Limited announced the release of a new Palm VoIP product called Articulation for PalmOS, that supports the SIP standard and therefore works with any VoIP service provider that supports the SIP standard. I should mention that it only works with Palm OS 5. Alas, the software doesn't yet work on the Treo 700p mobile phone, but should work on the Treo 650. It is connectivity-agnostic supporting WiFi, Bluetooth, EVDO, and EDGE/GPRS Internet connectivity.


Screenshot of Articulation for PalmOS



It isn't the prettiest interface in the world, but hey, if I can get my SIP on, I'm a happy man!

Features:
- Ability to make PSTN calls with your PDA
- Select the SIP VoIP provider of your choice
- Lookup phone numbers direct from you contacts
- Touch-tones (DTMF) for 'phone menus'
- Supports WiFi, Bluetooth and EVDO connections
- Call timer
- Silence suppression (only transmit your voice)
- Echo cancellation for 'speakerphone' use

Technical Features:
- Secure account authentication with SIP MD5 Authentication
- Supports GSM and G711 codecs (GSM is suited to low bandwidth connections)
- NAT support through RFC 3581 and ability to fix ports for symmetric NATs
- Symmetric RTP support
- Low latency
- Low memory usage

Good stuff! You can download and try it out for free. Still wish it supported the Treo 700p though.

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Tags: articulation, palm, sip, treo 650, treo 700p, voip
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Written by VoIP & Gadgets Blog on August 23rd, 2006 with no comments.
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Ingate Showcases SIP Trunking Solution

Ingate Systems, which develops firewall technology and products that enable SIP-based live communication for the enterprise while maintaining control and security at the network edge, showcases traversal and security solutions for enterprises connecting to SIP trunks, with Ingate Firewal and Ingate SIParator products at the VoiceCon show.

 

SIP trunking is a service offered by Internet Service Providers (ISPs) which permits businesses to adopt Voice-over-IP (VoIP) with its attendant benefits and remain connected to others who rely on the PSTN. For the enterprise SIP trunks can offer significant cost savings, as they eliminate the need to purchase either the local PSTN gateway or costly ISDN BRIs (Basic Rate Interfaces) or PRIs (Primary Rate Interfaces). The local IP-PBX can be connected to the service provider's PSTN gateways over the Internet.

Ingate Firewalls and SIParators solve the Network Address Translation (NAT) traversal issues that are faced by businesses using a SIP trunk. All voice traffic (as well as data traffic) must traverse the enterprise firewall/NAT. However, SIP traffic cannot traverse traditional enterprise firewalls and NAT devices. This is because traditional firewalls do not differentiate between SIP traffic and unwanted traffic. As a result, the firewall/NAT device blocks all SIP traffic, which includes VoIP. Ingate products resolve this issue, enabling enterprises to utilize SIP trunks.

Ingate's fully SIP-capable, proxy-based Firewalls and SIParators also provide flexibility to interoperate with carrier-specific requirements like numbering plans and authentication. They establish least cost routing rules, which offers businesses the ability to use multiple service providers, and to switch between them depending on which offers the best possible rates (which may vary by time, day or location). Long distance calls cost the same as a local call, reducing expenses for businesses as well as their customers, partners, etc. trying to reach, for example, the corporate sales force.

SIP trunks further reduce costs as they eliminate the need for separate voice and data connections, expanding the potential for communications convergence using both voice and data together. SIP trunking also offers scalability so that, as a company grows, all necessary infrastructure to handle additional voice/data traffic is already in place.

Security Over The Public Internet

Ingate products are specifically designed to leverage all the security benefits available with SIP communications. Ingate's enterprise-class Firewalls secure data and SIP traffic, while SIParators secure SIP media while leaving the traditional firewall in place (working in parallel to the SIParator) to secure data traffic.

Both products feature Ingate's full SIP proxy technology, which allows for advanced filtering, verification, authentication and routing, as well as dynamic control of the opening and closing of media ports. Encryption of the signaling is done using Transport Layer Security (TLS) and of the media (voice, video, etc.) using Secure RTP (SRTP). With encryption, the sessions are kept private with no chance of eavesdropping.

They also secure full VoIP redundancy, as traffic can be routed to a back-up carrier if the primary carrier is unavailable. And Ingate's proxy-based solutions can be used to provide local call management if the service provider cannot be reached; and, if the enterprise has a local gateway installed, outgoing calls can be routed to the PSTN if all connections to the hosting provider are unavailable.

"SIP trunking offers enterprises the benefits of converged communications and saves substantial expense by eliminating the need to purchase their own PSTN gateway," said Steven Johnson, President, Ingate Systems. "NAT/firewall traversal is a critical issue in any SIP trunk deployment. Ingate's SIP-capable Firewalls and SIParators offer an elegant solution to this problem, while also securing voice traffic."

Connecting Remote Workers To The SIP Trunk Without VPN

SIP trunking, when used in conjunction with Ingate's Remote SIP Connectivity software module, allows remote users to traverse most SIP-unaware residential firewalls and NAT devices and use all the IP-PBX functions installed in the enterprise. Remote SIP Connectivity complements SIP trunks, as remote workers can use the software to interface with their corporate IP-PBX and, if they were making calls to outside the enterprise, do so using the SIP trunk.

For more information, please visit us at Ingate's VoiceCon booth #639, or online at www.ingate.com.

Written by Dal on August 22nd, 2006 with no comments.
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Bandwidth.com Launches SIP Trunking Solution

Written by Dal on August 8th, 2006 with no comments.
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Asterisk basics: in six quick slides

I am not  a technical expert on Asterisk, but being fully aware of its gathering momentum as an is a free software / open-source software implementation of a telephone private branch exchange (PBX), I decided to hang out with some Asterisk code jocks for the better of an afternoon. I was much looking [...]

Written by Russell Shaw on July 30th, 2006 with no comments.
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VONaLink SoloRecord VoIP call recorder

There are a few general purpose VoIP call recording solutions that I've written about. Arcosoft Inc., is the latest to announce a VoIP call recording solution released today called VONaLink SoloRecord. SoloRecord works with any VoIP phone system based on the open SIP standard, such as Vonage, to record phone calls and to provide screen pops.
 
With traditional phone systems, calls are recorded with either analog equipment or expensive, proprietary products from the phone company. With the latest VoIP systems built on open, standard protocols, calls can be recorded by monitoring network packets.
 
VONaLink SoloRecord works with any SIP based VoIP system. Soft phones, hard phones, or analog phones via an analog telephony adapter (ATA) are supported since they are converted into IP packets which can be captured. The call is recorded as a stereo WAV or MP3. An inaudible watermark can be added to the recording for later verification that the file has not been changed.
 
Using the caller ID of the incoming call, SoloRecord searches for the caller in Microsoft Outlook contacts, or launches custom applications to search the web or your company database. If the caller is found, the information is popped on the screen, thereby increasing the efficiency of the customer service agent.
 
VONaLink SoloRecord runs on Windows XP, 2003, and 2000. The price is $99 USD per user, available now for evaluation download from www.vonalink.com

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Written by VoIP & Gadgets Blog on July 28th, 2006 with no comments.
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I’d like a pizza with pepperoni and a sprinkle of VoIP and some GPS too

Recently, I received an email from a pizza delivery solutions provider seeking my assistance in an interesting application that involves VoIP, GPS coordinates, Bluetooth, and of course pizza. This is not the first time I have linked pizza with VoIP. No siree bob! In fact, my Vonage VoIP line resulted in me getting a cold pizza. Almost sued Vonage over my damn cold pizza too.

I know what you're thinking. What kind of VoIP application does a pizza delivery company need, right? Well, the best way to explain the application that ties VoIP, pizza, Bluetooth, and GPS together is to include the request they sent to me.

We sell software services to pizza chains.  Our system tracks drivers as they drive around town.  Since we know where they are and where they are going, when they are about 4 minutes from the customers’ door, we want our computer to initiate a telephone call to the customer to say the driver will arrive momentarily.  We imagine some kind of a softphone application running on the PC connected to a VoIP access point to the phone system.  Our software already has the .wav files prepared to “speak” to the customers.

Any suggestions on how to configure this or who we should be talking to that might sell the necessary hardware software?  Is open source stuff available?

PS  Our system is written in C#, so is pretty flexible.

Pizza Pilot recently announced the completion of a multi-store test in Boston-area Domino's Pizza Franchises.  Pizza Pilot was successful in reducing labor, mileage theft and lost drivers.  New tools for measuring each driver's "Smart Hustle" factor as well as data mining techniques to identify and prioritize target customers with Platinum, Gold or Silver service levels were also introduced.

Pizza Pilot is a software-based product that works with any POS-including Pulse and TMS/National Systems.  In combination with Bluetooth and GPS-enabled cell phones, it tracks the location of each delivery driver every 60 seconds.  Pizza Pilot mapping software determines the optimal dispatching and automatically assigns orders to drivers-allowing managers to focus on inside operations.

Due to its ability to track each driver's progress on their way to each destination, Pizza Pilot can determine the moment actual deliveries are made and to update drivers' estimated return times as they return to the shop-making subsequent dispatches more accurate.  Maps or on-board navigation is also available which reduce lost drivers, while real-time tracking identifies and discourages unauthorized stops.

I'm most impressed with the fact that they want to call you when the pizza is 4 minutes away. Imagine that -- a pizza company that is courteous enough to call you when they are about to arrive. How many times have you been told the pizza will take 30 minutes only to see 90 minutes roll by; then when you call to check on your pizza they simply tell you the pizza guy is on the way. Really they have no idea where he is and they're simply patronizing you. Well, with this pizza delivery company, they track all their vehicles so they can tell you exactly where the vehicle is and on top of that they plan to call you using an automated dialer using a VoIP line.

Now if that isn't a hot application for VoIP, I don't know what is. Mmmm. hot pizza... Ahhhh. GPS & Bluetooth gadgetry... Ooooo... VoIP coolness... Ahhhhh...  All I need now is some cold beer & ESPN -- and I'll be in heaven. Maybe I'll launch SightSpeed 5.0 and watch some ESPN from work.

I suggested to Pizza Pilot that there is C# code out there to initiate SIP calls and they he may want to consider looking at Asterisk, the open-source IP-PBX. I also referred him to Erik Lagerway and Ward Mundy over @Nerd Vittles, an Asterisk blog. Erik suggested that since he was using C#, that something in Microsoft LCS might do the trick - or paid kits like those offered by Counterpath. Ward Mundy said he could code something for him and asked for more details on the size and scope of the project.

I'll keep you posted if the pizza/GPS/Bluetooth/VoIP application ever goes live. GPS VoIP pizza delivery just might be coming to a neighborhood near you!

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Tags: asterisk, bluetooth, c#, gps, open source, pizza, sip, voip
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Written by VoIP & Gadgets Blog on July 25th, 2006 with no comments.
Read more articles on VoIP and GPS and Asterisk and SIP and bluetooth and pizza and open source and c#.

BandTel increases SIP redundancy

BandTel announced this morning it is the first to solve the throughput and redundancy problems on high capacity SIP-based networks with its new N-Plus architecture. The solution lifts the burden placed upon servers by creating a clustered architecture that eliminates the need for numerous IP addresses on numerous SIP proxies and eliminates any single point of failure.  Of course, TMC has a N+1 RAID-5 Exchange Server, but that didn't stop 2 simultaneous hard drive crashes last week that broke the RAID array bringing down the Exchange Server. RAID-5 redundancy my %@$^&*! On a usually reliable Dell server no less... Probably a SCSI controller failure.

In any event, BandTel's N-Plus network is based on several pairs of DNS servers that direct the SIP calls to SIP Signaling Transfer Points (STPs), which in turn direct those SIP call on one of "N" SIP proxy's in the BandTel SIP proxy matrix. You can check out the full BandTel news here...

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Tags: bandtel, raid, redundancy, sip, voip
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Written by VoIP & Gadgets Blog on July 24th, 2006 with no comments.
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Paragon hipi dual-mode GSM phone

Paragon Wireless released the world first commercial GSM/VoWLAN dual-mode smart phone, hipi, in March 2006 and was my pick to be a winner in this years' TMC Labs Innovation Awards. With a tri-band GSM/GPRS (Class 10) radio and an IEEE 802.11b WLAN chipset, hipi enables users to enjoy broadband multimedia services at WLAN covered homes, offices, hot spots/zones as well as reliable GSM/GPRS service anytime anywhere. Featuring a 2.4 inch TFT touch screen, QVGA, with 260k Colors, the hipi also supports STUN-based NAT traversal, the SIP standard, as well as G.711, G.729a/b, and G.723 codecs. It even has some gadgety-bling via it's built-in MP3 player and a QVGA/QCIF camera. The hipi can perform SIP-based seamless handover between GSM/VoWLAN. Importantly, it utilizes a unified phone book for both GSM and VoWLAN dialing and a unified GUI for the main applications i.e. phone, E-mail, QQ (IM), and browser.

EXCLUSIVE! The current hipi is based on Linux, but I asked about a Windows Mobile 5 version since there are many popular 3rd party applications written for Windows Mobile and they said that a Windows model was indeed in the works - with some beta trials currently and plans for release later this year. You heard it here first - the first dual-mode Windows Mobile 5 phone out later this year! According to Paragon, the Windows-version will have comparable battery/performance characteristics of their current Linux model.

The hipi has excellent performance in power management, mobility management, security, mobile VoIP, and voice quality. hipi has passed most of regulation certification programs and has done interoperability testing with over 40 VoIP service providers, system integrators, and infrastructure equipment vendors worldwide. Standby time >100 Hours (GSM on, WLAN on) >200 Hours (GSM on, WLAN off). Talk time on VoWLAN is 3.3 Hours and for GSM it's 7.8 Hours. With such excellent battery talk times and standby - even with two radios operating - hipi is an ideal device for fixed mobile convergence. Check out the specs...

Hardware Specification (Linux model)
• Intel PXA271 processor with embedded Linux
• 2.4 inch TFT touch screen, QVGA, 260k Colors
• Built-in speaker/microphone, 2.4mm stereo and headset
• 1.3M pixel CMOS camera
• USB slave
• Mini SD
• 1100 mAh Li-ion battery

GSM Specification
• Frequency bands: 900/1800/1900 MHz
• GPRS Class 10
• SMS, MMS, WAP applications
• FTA/CTA certification
• FCC/CE certification

WLAN Specification
• IEEE 802.11b
• RF channels: US: 11, ETSI: 13, Japan: 14
• High-gain internal antenna
• WEP 64/128 bits, WPA, 802.1x
• EAP PSK/LEAP/PEAP/TTLS/SIM
• Power saving modes
• Fast roaming between access points

VoIP Specification
• SIP: IETF RFC 3261
• Codec: G.711, G.729a/b, G.723
• Acoustic echo cancellation
• Dynamic jitter buffer
• Voice activity detection
• Stun-based NAT traversal

Input Methods
• Handwriting Recognition > English > Chinese > Numeric characters
• Soft Keypads > Qwerty > Standard phone dialpad > Symbol

Power Management Features
• Standby time >100 Hours (GSM on, WLAN on) > 200 Hours (GSM on, WLAN off)
• Talk time > VoWLAN: 3.3 Hours > GSM: 7.8 Hours
• MP3 play time > 5.8 Hours (GSM on, WLAN on) > 6.2 Hours (GSM on, WLAN off)

Fixed Mobile Convergence Features
• Simultaneously activated GSM and WLAN air interfaces
• Handling simultaneously GSM and VoWLAN incoming calls
• SIP-based seamless handover between GSM/VoWLAN
• Automatic/manual switch for out-going calls between GSM and VoWLAN
• Automatic/manual switch for data applications using GPRS or WLAN
• Unified phone book for both GSM and VoWLAN.
• Unified GUI for applications (phone, E-mail, browser, QQ)

Call Features
• Call hold
• Call waiting
• Call mute
• Call forward
• Call transfer
• 3-way conference
• Voice mail
• SMS over SIP
• Phone book - (1000 entries with photos)
• Incoming call prompt with picture
• View phonebook during call
• Enter sketch pad during call
• Adjust volume during call
• Auto-answer/flip answer
• Quick silence
• Turbo dial
• Manual/Auto/Earphone redial
• Call history (20 entries) Data Application Features
• POP3 E-mail client (SSL support) > 100 full E-mails with attachments up to 200KB > Document viewer for MS-Office and PDF files
• Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0
• Instant messaging: QQ Multimedia Features
• Video format: MP4, 3GPP
• Audio format: MP3, WAV, MIDI, AMR
• Picture format: WBMP, BMP, JPEG, GIF
• Camcorder: QVGA, QCIF
• Media Player > Audio: MP3 player > Video: up to 30 frames/second QVGA MP4/3GPP PIM Features
• Calendar
• Schedule management
• Alarm clock
• Voice recorder
• World time
• Currency converter
• Anniversary Other Features
• English <-> Chinese dictionary
• Calculator
• World time
• Notepad
• Sketch pad
• File transfer
• Counter
• Timer

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Tags: dual mode, hipi, linux, mobile phone, sip, voip, wifi, windows mobile 5, wireless
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Written by VoIP & Gadgets Blog on July 23rd, 2006 with no comments.
Read more articles on VoIP and Mobile Phones and Wireless and WiFi and SIP and Linux and mobile phone and dual mode and hipi and windows mobile 5.

CommuniGate Demonstrates SIP Farm for 10M VoIP Subscribers

 

CommuniGate is claiming that the demonstration is the first milestone in building large-scale, standards-based communities of VoIP subscribers, since traditional telecommunications, cable, and wireless providers often support over 50 million subscribers. The benchmark simulation was conducted to provie cost-effective scaling by just adding notes, using either virtualization or physical additions of servers to the running cluster as needed.

Call load generation topped out at 1,000 SIP calls per second with up to 192,000 unique registered user-agent enpoints driven by a Navitel device for inbound calling. Simultaneously, the open source "sipp" SIP load generator was used to generate additional outbound calls.

Using a virtualized cluster on a single Superdome system, the benchmark team discovered that the CommuniGate SIP Farm and F5 supplied hardware continued to perform within required QoS until no further load could be generated by the test environment available. Future tests will use a fully-equipped Navtel chassis to increase available SIP load capability. A full whitepaper report of the test demonstration is available at: http://www.communigate.com/content/whitepapers.htm

Written by Dal on July 18th, 2006 with no comments.
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Cistera Networks’ New SIP Presence Engine Enhances VoIP Deployments

 

Demand for PresenceManager is strong across all vertical markets, and successful deployments have already been made for large customers in the education vertical market including Corpus Christi and Katy Independent School Districts.

Presence Engines allow IP phone users and administrators to tailor their environments based on multiple factors, including physical presence and availability. The Cistera ConvergenceServer™ uses PresenceManager to appropriately deliver, record and playback calls, to present notifications and events correctly and to allow a more comprehensive real-time directory view of an organization.

"In order for customers to transition from traditional telephony to VoIP, we need to ensure the availability of features they have come to depend on. What used to be called 'Do-Not-Disturb' or 'Time-of-Day Restrictions' are actually early implementations of what is now known in the industry as 'Presence,'" said Cistera Networks CTO Greg Royal. "We took those initial features and brought them into the 21st century, while adding additional functionality that is available in a VoIP environment."

Cistera is one of only a handful of vendors that can deliver both "local" presence, where users set their own presence, and "global" presence, where the administrator can set the presence of large numbers of phones. "We have one customer that can set the presence of over 6,500 phones on a daily basis. The Cistera ConvergenceServer manages and routes over 10,000 calls a day for that customer through Cistera PresenceManager," said Royal.

"Because we natively support Session Initiated Protocol (SIP) on our platform, we can extend it to include SIMPLE (SIP for Instant Messaging and Presence Level Extensions) platforms such as the Microsoft Live Communications Server (LCS) 2007, and Cisco's Communicator products," continued Royal.

The Cistera ConvergenceServer (CCS), with its robust suite of application engines, offers customers IPT applications that are easy to integrate and install. The CCS adds critical competency and features such as text and audio broadcasting, messaging, recording and content streaming within Cisco and Nortel Converged Communications environments.

Source: Cistera 

 

 

Written by Dal on July 13th, 2006 with no comments.
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Tutorial shows how to configure P2P SIP dialing on your Asterisk PBX

Barrett Lyon has created a don't miss resource that shows you how to configure P2P SIP URI (Uniform Resource Identifier) dialing on your Asterisk-based PBX."There are thousands of people that operate their own Asterisk based PBX systems, yet they do not enable any method to allow for p2p sip URI dialing," Barrett explains. "These [...]

Written by Russell Shaw on July 12th, 2006 with no comments.
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The Perfect VoIP Softphone and IM client

With more softphones on the market than you can shake a stick at, each with their own "island" of users that cannot bridge to other islands, I've become a bit disillussioned with the VoIP softphone market. Sure, there is talk about interoperability, support for SIP/SIMPLE, XMPP support, etc. but we still don't have a single unified client that can speak to Skype users, AOL/AIM users, ICQ, Google Talk, MSN Messenger, Yahoo! Messenger, etc. Although, software such as GAIM is a multi-protocol client that aims to unify all of these IM/softphone clients, it too is incomplete. For one, it was designed mainly to handle instant messaging (IM) and not voice or video. Although, a branch of the open-source project called GAIM VV was started to work on adding voice and video support, but it appears to be a dead project since the last posted news was October 2005.

This lack of a single unified client means I have to have like 4-5 IM/softphone clients running on my PC, which uses more memory, requires more time spent keeping each client up to date, uses more hard disk space, Registry bloat, and other inconveniences. It got me thinking though - what features would I like to see to create the perfect VoIP softphone and IM client?

And so, here is my "wish list" for the perfect softphone/IM client...

Continue reading The Perfect VoIP Softphone and IM client...

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Tags: google, im, messenger, microsoft, sip, skype, softphone, streaming, video, voip, yahoo
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Written by VoIP & Gadgets Blog on June 19th, 2006 with no comments.
Read more articles on Skype and VoIP and Google and Microsoft and yahoo and SIP and video and im and softphone and messenger and streaming.

ATS DECT 6.0 cordless VoIP phones

Tomorrow, American Telecom Services, Inc. (ATS) will announce the release of its first DECT 6.0 cordless multi-handset Internet phone. The master units (E6501) are expected to retail at a sub $50 mark (likely $49.99) and the extensions (E6502) will each retail for sub $30 (likely $29.99).

The E6501 includes ATS’ patent-pending Digital Clear functionality and is expandable to up to five total handsets.  ATS claims this is the first cordless phone to use DECT technology that integrates a router and a SIP VoIP platform in the charging base of the master cordless phone unit. Uniden makes a cordless phone system, which I recently reviewed - the UIP1869V, that supports SIP, but it's not DECT and its locked to Vonage, so the ATS phone system is the first cordless phone system (DECT or otherwise) that is an open SIP system to use with any VoIP provider that provides you with SIP credentials. Nice thing about that is if you change VoIP providers you aren't left with a "bricked" phone system.

The ATS E6501 and E6502 will be sold by American Telecom under the ATS brand bundled with service through major retailers (online and offline) in the U.S. They are not announcing which service providers and/or retailers will be carrying it at this time, however if you look closely in the photo you can see SunRocket displayed, a well-known VoIP service provider. The phones will also be available wholesale to service providers who wish to distribute them through their own channels.
 
Basic specs are as follows:

• DECT 6.0 communication with a range of more than 650 feet
• Expandable design up to 5 total handsets
• Integrated adapter and router
• SIP compliant VoIP platform
• LCD display
• Page/locator button on base
• Intercom between handsets
• Speakerphone on all handsets
• Headset jack on all handsets
• 60 number phone book
• Auto answer
• 10 hours of talk time
• 10 selectable ring tones
• Hearing aid compatible

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Tags: american telecom services, cordless, dect, sip, voip
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Written by VoIP & Gadgets Blog on June 13th, 2006 with no comments.
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Linksys WIP330 Wireless-G IP Phone Review

Ratings	Score
Installation
Documentation
Features
Usability
Value
Performance
Overall

The Linksys WIP330 Wireless-G IP Phone is one of the coolest gadgets I've tested in awhile. Linksys sent me a WIP330 for a product review two weeks ago and I played around with it quite a bit, but just never found the time to write the review. Well, here goes, an exclusive look at the Linksys WIP330 Wireless-G IP Phone...

The Linksys WIP330 Wireless-G IP Phone enables VoIP service through a Wireless-G network connected to the Internet to make low-cost VoIP calls through your Internet Telephony Service Provider (ITSP). The handset features peer-to-peer dialing, speed dial, 3-way conferencing, call waiting, call transfer, and call forward, mute, hold and selectable ringtones. The Linksys WIP330 measures 46.7 x 135.2 (142.0 with Ant) x 18.8 x mm (1.84 x 5.32 (5.59)x 0.74 in) and weighs 0.119 kg (4.20 oz). It uses a 3.7V 1250mAh Lithium Battery, and includes a 5V 1.0A AC Adapter, but can also be charged using a mini-USB plug.

It's worth mentioning that Linksys makes a similar model, called the WIP300. The WIP300 and WIP330 are similar in that both models let you do VoIP from 802.11b/g WiFi networks and support SIP and the SIP v2 standards. The similarities end there. The WIP330 is the higher-end model, sporting a larger 2.2" LCD screen vs. 1.8" LCD for the WIP300. The WIP330 unit adds hotspot support and the ability to browse the Web on its 2.2-inch color display. The WIP300 on the other hand doesn't have Web capabilities, but it's also less expensive ($219.99 vs. $369.99). The WIP330 runs on the Windows CE 4.2 operating system which enables it to run the CE-version of Internet Explorer for access to just about any web page content.  Although, I wasn't able to view my Linksys webcam which uses an ActiveX control from this mini-IE version. Linksys claims that you can receive live video on the WIP330 Wireless-G IP Phone when you access any web camera, including the Linksys WVC54GC Wireless-G Internet Video Camera. I actually just got my hands on a cool Linksys Wireless-G PTZ Internet Camera with Audio WVC200 that I am reviewing but wasn't able to get it to work with the WIP330. When trying to connect to my home webcam I receive an authentication dialog box and the number keypad won't let me enter in my username/password - it just ignores my key presses as the cursor flashes slowly as though mocking my attempts to access my home webcam. I'll have to contact Linksys tech support on that one.

Installation
In any event, installing and setting up the WIP330 was a breeze and in fact I didn't need to read the manual to do it - always a good thing. Linksys is known for easy, turnkey networking solutions, so considering the complexity of WiFi combined with configuring VoIP SIP settings, I was impressed with how easy the phone was to install. One caveat - Linksys pre-configured the SIP settings for me, but I checked them out and could have just as easily entered the info myself.

WiFi Settings
After first turning on the phone, I simply added my WiFi router's SSID and the WEP encryption key and the WIP330 acquired an IP address, DNS settings, etc. from the DHCP server. Next, I made my first test call by dialing the phone number that Linksys pre-configured on the device. I dialed 1-415-762-XXXX and the WIP330 rang and displayed the CallerID of the caller. It doesn't support CallerID with Name, however if you add the phone number to the Phonebook and associate a contact record, it will display the name of the contact. The handset stores the last 20 call history records and can save 250 phone book entries. When entering contacts using the 10-digits on the keypad, you have to have fast fingers when entering in the alphanumeric characters or risk it moving onto next letter. If I hesitated for a split second, it moved to the next character. The delay should be user-configurable.

The sound quality was excellent, which is attributable to the G.168 echo cancellation, jitter buffer control, and packet loss concealment. The comfort noise generation (CNG) also helps the voice quality perception. The phone supports the usual codec suspects, including G.729ab, G.711u, and G.711a. While you are on a call, you can mute the call, put the caller on cold and even answer a second call on Line-2. In fact, I initiated a 2nd call to 1-415-762-XXXX from another phone and I heard a series on tones on the ear-piece indicating a 2nd inbound call. I was able to take that call which automatically put the first caller on hold. Simply by pressing the softkey button underneath "Switch" on the LCD caused it to toggle between the two calls. Next, I pressed the softkey button underneath "Options" and was able to select "Conference" which then conferenced the three phone devices together (2 landlines & 1 WIP330 IP connection). In addition, there are two types of transfer features available. You can use the Transfer feature when you want to introduce the caller you are transferring or you can use the Blind TFR (Transfer) feature when you want to transfer the caller without introduction.

Profile of the phone:


Profiles
Up to 10 hotspot/WiFi profiles can be stored and you can designate the order of the profiles saved by the IP Phone. The next time it is powered on, it will use the first profile to automatically connect to the profile's wireless network. If that network is not available, the IP Phone will try the next profile. This will continue until it has connected to a wireless network. It can connect to "open" WiFi networks or secured WiFi networks - both WEP (64/128) and WPA-PSK encryption are supported.

Phone Customization
You can customize various settings of the IP Phone including the Ring Option, Keypad Tone, Wallpaper, Date & Time, Language, and Phone Password. You can also upgrade the firmware of the phone over an IP connection. There are five different ring types - ring, ring once, silent, vibrate, and ring & vibrate, as well as (they claim) seven customized ring tones in the documentation, but I only found 3 available. Perhaps they send me an early production model?

Web browser
The web browser is the Windows CE version. Entering in URLs using just a number pad is a bit of a pain, but with a little practice I was able to quickly navigate some sites on the 2.2" LCD screen. Although you can browse the web on the phone, really the browser's primary purpose is to allow you to logon to hotspots, hotel WiFi networks, and other WiFi networks that require some sort of Web-based authentication - often used to ask for billing info. I tested the browser and it worked pretty well, but you'll definitely want to add your commonly visited sites to the Favorites, since I already stated, typing on the number pad is tedious.

I couldn't figure out way to add a currently viewed webpage to Favorites. The toolbar didn't display an icon for adding the current page to Favorites which would make it much easier. Instead, I had to exit out of the browser, go to separate screen and manually type the URL. This is a pain considering how difficult it is to type a long URL along with special characters (: and /) using just a number keypad. Also, if it's a long URL path you may not even remember the exact URL to key in.

When surfing the WIP300, it has an auto-scroll feature so you don't have to repeatedly press the navigation button. You simply press and hold down the center selection key for two seconds until you see an orange navigation cursor. Then you can use the navigation pad to scroll up and down the webpage. To use the cursor as a page up and down button, you simply press and hold down the center selection key for an additional two seconds until the cursor turns blue with a "P" in the center and then use the navigation pad to move up and down the webpage.

Here are some of the menus on the WIP330:


Here's the phone's web admin:

Here are the specifications:
• Support SIP v2 Standards
• Compliance with IEEE 802.11b/g Wireless Standard
• Powered by Microsoft Windows CE, with IE Web browser
• High Resolution Color LCD screen
• Support QoS (Quality of Service) to ensure best quality voice
• Enhanced Power Saving Design for Extended Standby and Talk Time
• 50 hours standby time, 3 hours talk time on average
• 3-Way Conferecing, Call Hold and Resume, and Caller ID
• Fast Hotspot Authentication
• Supports auto-provisioning using HTTP or HTTPS for configuration and upgrades
• Outdoor up to 300m via Embedded Antenna
• One mini-USB Socket, One Stereo Ear Phone Jack
• QVGA TIF 2.2 inch LCD (240*320 pixels) with 65K colors
• 32MB Flash, 64MB SDRAM
• Multiple Access point Registration Support
• TCP/UDP/IP, IPV4, DNS, SDP, ARP, ICMP, DHCP Client, Static IP
• WEP (64/128), WPA-PSK Encryption
• ToS
• SIP v2 Session Initiation Protocol (RFC3261), SDP (RFC2327)
• SIP Session Keep Alive
• G.711( A-law and μ-law), G.729 A
• In-band, Out-band (RFC2833)
• G.168 Echo Cancellation
• Jitter Buffer Control - (default 180ms, max 900ms)
• Comfort Noise Generation
• Packet Loss Concealment
• Speaker and Microphone Volume Control
• VAD - Voice Activity Detection
• 3-Way Conferencing
• Peer-to-Peer Dialing
• Call Hold and Resume
• Caller ID Presentation
• Caller ID Presentation Restriction
• Dial by Phone Number
• Call Forward
• DTMF Tone Detection
• Consultation Hold and Transfer
• Call Waiting and Retrieve
• Mute
• Speed Dial
• Last Number Redial
• Volume Control
• Ringtones: True Tones
• Phone Book (250 records)
• Call History (20 Records)
• Language (English/Spainish)
• Vibrator (Silent mode)
• Password Security
• Date & Time (NTP time synchronization)
• Internet Web Browser (Microsoft IE)
• AES or SSL Encryption
• Firmware upgrades using HTTP, or HTTPS
• Configuration change using HTTP, or HTTPS
• Embedded Web configuration interface (with password protection)
• Power-up Diagnostic

Pros
- thin and lightweight
- Web browser surfing capability
- Phone itself had web admin interface
- support for 10 WiFi profiles and built-in support for T-Mobile, Telefonica, and Hinet hotspots
- SIP v2 support
- Pressing +/- during a call will adjust handset speaker volume
- Can sync/connect to the device over USB

Cons
- No POP3 or SMTP email support (though can always use web browser to send or check email)
- You have to have fast fingers when entering in alphanumeric characters or risk it moving onto next letter
- Pressing +/- while not on a call doesn't adjust ringer volume - have to go into settings
- No speakerphone
- Mystery button on its left side that does nothing. Perhaps for future firmware release?
- Only supports 27-bit WPA key
- No Skype support
- Pricey @$369.999 - you can buy a Windows Mobile 5.0 phone or PocketPC for this price

Conclusion
This Linksys WIP330 is great for businesses using any IP-PBX that supports the SIP standard, which is pretty much every IP-PBX these days. If you recall, I had an "exclusive" on Cisco finally adopting the SIP standard (Cisco Unified Communications) over their proprietary Skinny protocol, thus you can even use the Linksys WIP330 on a Cisco Call Manager system. Of course the Linksys WIP330 will work with the ever-popular Asterisk phone system as well. No Skype support is surprising, but I did notice some Skype config files on the phone itself, so perhaps this capability is coming. Certainly Windows CE has the ability to run a Skype client. Even with its built-in browser and its ability to logon to authenticated hotspots, due to its price ($369.99), only executives with money to burn, or gadget freaks that just have to have the latest and greatest gadgets, will find the Linksys WIP330 attractive enough to purchase.

Gadget freaks who just have to have the most feature-rich WiFi phone available today can get the Linksys WIP330 WiFi VoIP phone from Amazon as well as from VoIPSupply.com.

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Written by VoIP & Gadgets Blog on June 12th, 2006 with no comments.
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IP-PBX market numbers continue to shine

Interesting IP-PBX stats within Infonetics's Enterprise Telephony report that shows IP-PBXs continued growth even with a generally slower 1st quarter. They also list the current IP-PBX market leaders (Cisco is listed as #1). I'd be curious the stats for "hosted IP-PBXs" that use SIP trunking with no customer premise IP-PBX equipment. My bet is that market is growing as well.

Overall PBX/KTS revenue down 2%, IP PBX up 1% in 1Q06

Despite a seasonally down quarter for the overall enterprise telephony market, the slow and steady move from circuit switching technology to packet switching technology remains evident, with worldwide TDM system revenue falling 11% and IP PBX revenue inching up 1% between 4Q05 and 1Q06, says Infonetics Research in its latest Enterprise Telephony report.

Combined, worldwide TDM and IP PBX systems revenue dipped 2% to $2.1 billion in 1Q06, but is 15% higher than a year ago. Annual revenue is forecast to grow to $11.4 billion in 2009, driven by strong IP PBX sales worldwide as more organizations move to voice over IP. Between 2005 and 2009, IP PBX revenue is forecast to jump 82% while TDM revenue plunges 88%.

"The overall enterprise telephony market was not immune to the first quarter blues, but the IP PBX category managed to eek out a small quarterly gain" said Matthias Machowinski, directing analyst at Infonetics Research. "We recently talked with 450 companies in North America about their voice infrastructure plans, and the results clearly indicate a steady move to VoIP, which will put this market on a nice steady growth trajectory over the next few years."
 
1Q06 Highlights
   -  In EMEA, the top IP PBX system line shipment vendors are Alcatel, Siemens, and Nortel
   -  The top IP PBX system vendors in North America are Cisco, Avaya, and Nortel in a very close race: Cisco was just in 3rd position the previous quarter, and the difference in 1Q06 market share from 1st to 3rd is less than 2 points
   -  Cisco leads the IP phone market, with 39% unit market share; the next closest competitors are 3Com and NEC, who are tied for 2nd
   -  Hybrid PBXs account for 63% of PBX line shipments; by 2009, they will account for 78%, up from 61% in 2005
   -  45% of PBX/KTS systems revenue comes from EMEA, 30% from North America, 19% from Asia Pacific, and 7% from CALA

Infonetics' report tracks IP deskphones, IP softphones, TDM PBX/KTS systems, and hybrid and pure IP PBX systems and IP PBX shipments by system size (2-40, 40-100, 101-400, 401-1K, and 1K+ lines) in North America, EMEA, Asia Pacific, CALA, and worldwide. Companies tracked include 3Com, Aastra, Alcatel, Avaya, Cisco, Ericsson, Inter-Tel, Mitel, NEC, Nortel, Polycom, ShoreTel, Siemens, snom, Sphere, Swyx, Toshiba, Vertical, Zultys, and others.

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