Meet Our Newest Addition to The VoIP Insider!

January 23, 2007 by Garrett Smith

Meet Arlyn E. Pifer

arlyn pifer

Garrett Smith, approached me about creating a blog that had to deal with VOIP in the government and education sectors. I am not now, nor will I ever claim to be the expert in this particular filed, or any other field for that matter, but if you care to come along for the journey, maybe we can identify additional opportunities or learn something together. It is my intention to provide editorials and commentary and to locate and announce some useful websites. In addition, any articles or commentary of interest will be posted. I’ll also relay some of the cool implementations of voip in those sectors. Feel free to join in and trust me on this one, I have been accused of having Teflon skin more than once in my life – if you disagree or have a varying point of view – express it – just use some semblance of support for it!!

I am currently involved as an VoIP PBX Consultant with PBXSelect, LLC. One of my other roles is to develop the GovEd opportunities for our company. PBXSelect is a sister company of VoIPsupply.com, but focuses on the IP-PBX product offerings. Prior to joining this company, I have held various sales and sales management positions in the information technology industry. Even had the opportunity to work at stops of the supply chain as well. Manufacturing, wholesale distribution, reselling, (VAR and Retail) and have also been an end-user, (since 1984). My undergrad degree is in Accounting from Valparaiso University and I received my MBA from Colorado State University.

Skype Announces New Calling Plans

January 18, 2007 by Garrett Smith

Skype’s New Pricing is Anything But Disruptive

In a move to better monetize their service, Skype today announced a new connection fee for international calls, and a new plan, SkypePro, that offers a flat rate monthly fee for free nationwide monthly calling. Touting their new pricing structure as “distruptive,” for most Skype users the only thing distruptive about these new calling plans is the additional money that will need to be spent on connection fees.

While Skype is still one of the cheapest ways to make a phone call, the PR spin on charging customers more for the same service is, well, laughable. There are times for spin and there are times to tell the truth. Everyone knows Skype is a business and everyone knows they are in business to make money. Be up front about it. Don’t tout “disruptive pricing” and “Taking Internet Communications One Step Further With New Pricing Strategy”, tell us, the consumer, that the gravy train is over and we need to start to pay. Be honest and open.

Here are the facts of the new pricing structure:

New connection charge (3.9 eurocents per call)
Skype Pro calling plan for some countries (Flat Rate)
Select regions receive 1.7 eurocent per minute global calling

From the looks of this, and other moves, the “land-grab” is over. Skype is now looking to monetize of all of the subscribers that have been using the service for free, now that they are either hooked-on it or have seen the value of Skype. Great for Skype, not so great for infrequent users. It had to come sometime….

Polycom IP430 IP Phone Review

by Garrett Smith

Introduction to the Polycom IP430

The Polycom IP 430 is an enterprise-grade IP phone, The SoundPoint® IP 430 is designed to meet the telephony needs of general business users – cubicle workers that conduct a low-to-medium volume of calls – by delivering a robust feature set encompassing traditional telephony features such as call, park, pick-up, hold and transfer, as well as more advanced capabilities such as shared call / bridged line appearance, multiple call appearances, and presence.

Polycom IP430 Features

The Polycom IP430 is a two-line desktop IP phone that delivers the outstanding voice quality and smoothness of natural two-way conversations with its full-duplex speakerphone featuring the Polycom Acoustic Clarity Technology.

• 2 lines
• 132 x 46 pixel graphical LCD with line LED indicators
• Full-duplex speakerphone with Acoustic Clarify Technology
• Intuitive user interface
• Robust feature set and security
• Integrated Power over Ethernet Circuitry (IEEE 802.3AF)
• Dual switched 10/100 Mbps Ethernet ports
• Built-in auto-sensing IEEE 802.3af PoE support
• SIP

Polycom IP430 Product Packaging and Documentation

The Polycom IP430 was packaged well and came with all accessories. Documentation consisted of PDF files I downloaded from Polycom’s website.

One thing I noticed about the IP 430 itself is that there is no special notched Ethernet jack like Polycom phones of the past. This type of jack could only be used with their special included notched Ethernet cable. This special cable would be used for Power over Ethernet or as a point where a wall power supply would plug into. The Ethernet jacks on the phone are ordinary, which means that readily available ordinary Ethernet cables are used.

Polycom IP430 Administration, Configuration, and Use

I used the voice over ip phone’s internal web pages to initially configure the phone to get it up and running quickly. The five sections of this page are titled Home, General, Network, SIP, and Lines. The settings on these pages are enough to register the phone and make calls, but more settings can be changed by using configuration files with an FTP server instead. The phone can also be configured through its LCD screen.

Polycom IP430 Asterisk Integration

The version of Asterisk used for this review is 1.4 beta 3. I used vsftpd as an FTP server for configuring the phone via FTP. The phone came with SIP firmware version 1.6.7 and bootrom version 3.2.1.0012. I found out that Polycom allows for the downloading of older versions of SIP firmware from its web site. The most current versions of SIP firmware and bootrom files are only available to Polycom channel partners.

I downloaded a zip file for SIP firmware version 1.6.7. This zip file contains sample configuration files, the SIP firmware, and locale specific files. The home directory I used on my FTP server is /home/PlcmSpIp. I edited the configuration files as needed by referring to a SIP admin guide downloadable from the Polycom web site. After rebooting the phone, the new settings in the configuration files were not being used on the phone. I found out after some more reading that any settings entered via the phone’s internal web pages would override the FTP configuration files. I found a way of erasing these internal phone settings through a menu entry. After rebooting the phone again, the phone took the settings from the FTP configuration files.

The FTP configuration files have many changeable settings. Changes I made included SNTP time sync, turning an audio MWI off, assigning an extension for retrieving voicemail, and making both of the two line appearances active on the assigned extension.General features such as call transfer, call forwarding, conference, and do not disturb all worked fine.

Polycom IP430 VoIP Service Provider Integration

I don’t use VoIP service providers myself. I had no problems getting the phone to be used as an extension on my Asterisk PBX and dialing calls via Ma Bell.

Polycom IP430: The Good

The best thing about the Polycom IP 430 is it’s mass deployment and configuration functions. These features are found on other phones too now.

Polycom IP430: The Bad

Dialing a SIP URL is possible only by using the twelve key number pad. Many key presses are needed to get an output of one letter, which is inconvenient. Entering a SIP URL would be better accomplished by using an internal web page instead.

One of the phone’s PDF files I was reading said the phone has instant message functions. Composing or replying to an incoming instant message is possible only by using the twelve key number pad. This text entry method has the same problem that entering a SIP URL does as a result.

Overall Impressions

The phone sounds as good as other similar IP phones I have used. I recommend this product.

About the Reviewer

Dave Roper is an IT professional with experience in VoIP, security, networking, Linux, and Windows. He has worked with VoIP companies at the carrier and office level and in deployments of PBXs and IP phones. He has been working with Asterisk and VoIP since 2004 and Linux since 1998.

Linksys WIP300 Wi-Fi IP Phone Review

January 17, 2007 by Garrett Smith

Introduction to the Linksys WIP 300

The Linksys WIP 300-NA Wi-Fi IP Phone enables high-quality voice over IP (VoIP) service through a Wireless-G network and high-speed Internet connection. Connect at home, your office, or at a public hotspot, and make low-cost phone calls through your Internet Telephony Service Provider.

Linksys WIP 300 Features

The WIP300 Wireless VoIP Phone operates in the 2.4GHz band, supports 802.11g and the latest VoIP SIP protocols. The large, full-color high resolution display features an intuitive user interface enabling users to easily and quickly configure the handset using Secure Easy Setup (SES).

• Pixel-based display—Provides intuitive access to calling features
• Nine speed dials configurable in the set
• Comfort noise generation (CNG), voice activity detection (VAD), adaptive jitter buffer, and echo cancellation
• RF and battery level indication
• Local phone book
• Embedded 2.4GHz antenna
• ABS+PC plastic housing
• 1.8” COLOR TFT LCD with backlight
• Simple keypad with backlight
• Remote Firmware upgrading via Wi-Fi
• SIP v2 signaling protocol, RFC-3261
• POP3/SMTP E-mail access (optional)
• SMS (optional, by system default)
• USB charger interface

Linksys WIP300 Product Packaging and Documentation

The Linksys WIP 300 is packaged well in a small box, with plastic packaging to prevent the phone and battery from moving around. Documentation consisted of a quick start guide on paper and a full user guide in the form of a PDF file on a CD. I didn’t read either one since the onscreen menus were intuitive enough to get the phone up and running. An electrical charger is included too.

Linksys WIP 300 Administration, Configuration, and Use

The administrative/configuration interface for the WIP 300 is similar to that of a typical cell phone, in that every menu function is squeezed into the small space of two soft keys, directional keys, a small LCD screen, etc.

You have to get used to the menu structure of the phone and how the buttons operate. There is an easier to use internal web page too that can be used to view or change settings.

Linksys WIP300 Asterisk Integration

I was able to connect the Linksys WIP 300 to my wireless equipment with no problems. For the review, I have a Buffalo WBR-G54 802.11g wireless access point/router. Since my WBR-G54 is not my primary firewall, I had to set it up as a quasi access point to ensure that the WIP 300 and my Asterisk server were on the same subnet. I setup the WIP 300 with manual IP address settings since my own network is generally an exception to the rule. Most often, a wireless VoIP phone would get its IP address from an up-steam DHCP server. This would be the case if a wireless access only product with a separate DHCP server is in use or a wireless access point/router that has its own built in DHCP server is used instead.

Wireless settings were programmed as 802.11g only, a hard to guess preshared key and SSID, WPA encryption, TKIP, and no SSID broadcast. These settings were programmed into one of the default profiles on the WIP300.

Settings for SIP registration were programmed into MySQL for realtime integration. I programmed additional statements in extensions.conf related to other phones I have.

For the review, I programmed my primary desktop IP phone to always forward incoming calls to the Linksys WIP 300. I was able to receive calls with no problems during this test.

Linksys WIP300 VoIP Service Provider Integration

I don’t personally use a VoIP service provider myself for any long distance calls. I have a traditional Ma Bell phone line instead for any local or long distance phone calls. The phone had no problem acting as an extension on my Asterisk server, so I was able to make local and long distance calls as I normally would on a desktop IP phone.

Linksys WIP 300: The Good

The phone is small, lightweight, and would function well as a secondary IP phone. Sound quality was good on the call flow tests I made. A test of sending and receiving email messages via POP3 and SMTP worked well.

Linksys WIP 300: The Bad

The battery cover easily slid off when I handled the phone in my hand.
The cover reminded me of one found on a TV remote control. The latching mechanism needs to be improved.

A blind transfer function test didn’t work properly with Asterisk. I made a call from the WIP 300 to one of my two desktop IP phones. I put the call on hold from the WIP 300 and dialed the other desk phone, then pressed the transfer soft key. I answered the second desk phone. Audio only went only one way between the two desk phones. A similar blind call transfer initiated from the desk phones went fine. There may be a problem in the SIP or RTP implementation of the WIP300. An upgrade of the firmware (1.00.07) and bootloader (1.00.02) didn’t solve this problem. The version of Asterisk for this review is 1.4beta2.

The Linksys WIP 300 does numerical dialing only. This will probably be the case for most if not all wireless IP phones of this type. I tried to dial a SIP URL from the keypad or the speed dial list, but was unable to.

I am not sure how well the battery will perform over time, so further tests of this type would need to be done. I was unable to perform a wireless roaming test since I only own one wireless access point/router.

Overall Impressions

The phone functioned well overall and I recommend it.

Cordless VoIP Phones The New VoIP Trend at CES

January 11, 2007 by Garrett Smith

Cordless VoIP Phones Are All the Rage
Fueled by the need to push VoIP servicse such as Skype away from the desktop and closer to the traditional calling experience, VoIP hardware vendors proved at this year’s CES that 2007 may very well be the year of the cordless VoIP phone. Now when I say cordless VoIP phone, I mean just that – a phone with no cord to it’s base station – not wireless phones. While I have no doubt that wireless phones will continue to soar in popularity as wireless accessibility expands, I believe cordless phones of the DECT and GHz variety show the most promise for residential users.

The Benefits of Cordless VoIP Phones
Cordless VoIP phones could be considered the most ideal VoIP CPE option for the residential and small office VoIP users because of the following:

Flexibility – Most cordless VoIP phones offer a range of over 100ft
Scalability – Most cordless VoIP phones offer multi-handset capability
Familiarity – Who hasn’t used a cordless phone?
Streamlined – No need for multiple ethernet drops or cords
Accessibility – Never miss a call when your away from your desk

Top Cordless VoIP Phone Solutions

Netgear SPH200D Cordless VoIP Phone for Skype $199.99 – One of the hottest new products for Skype that came out of CES was the Netgear SPH200D. The Netgear SPH200D is Skype Certified SPH200D incorporates DECT (Digital Enhanced Cordless Telephony) technology which utilizes the 1.9GHz band enabling the included cordless phone to avoid interference from WiFi networks. The SPH200D’s dual-mode capability allows users to leverage Skype and the PSTN for calling. The dual-mode capability enables “life-line” capability Skype users, something that Skype lacks as a stand-alone service.

Netgear SPH200D Features and Functionality – The Cordless Phone Base Station incorporates an Ethernet (RJ-45) connector which plugs into the home network router and a PSTN (RJ11) connector which plugs into a traditional telephone wall jack. The SPH200D comes with one handset included and can support up to four cordless handsets per household, with additional handsets available for purchase separately.

Manage your contact list and see who’s available to talk to
Use premium services like SkypeIn, SkypeOut, Skype Voicemail (additional fees apply)
Supports up to 4 handsets per household (additional handsets sold separately)
DECT Cordless–ideal for long range and clear voice quality
Multiple Languages Supported

Uniden UIP 1868 $239.99 – The UIP1868 is a fully integrated VoIP home phone system, deploying VoIP capability to a maximum of ten handsets and offering consumers the benefits of VoIP communication in any room of their homes. Featuring 5.8GHz technology, the UIP1868 is compatible with Uniden’s TCX800, TCX400, TCX440 color screen phone and the new CLX500 flip phone accessory cordless handsets and is expandable up to a maximum of ten handsets without the need for additional broadband connections. As such, Uniden’s UIP1868 affords consumers the flexibility of utilizing VoIP communication at up to ten locations.

Uniden UIP1868 Features and Functionality – Uniden’s UIP1868 supports one telephone line and one fax line. Other key features include call waiting, caller ID, handset speakerphone, four-way conferencing, DirectLink™ two-way radio capability and a do not disturb setting.

Supports up to 10 Digital 5.8GHz Accessory Handsets
Supports 1-VoIP Voice Line and 1-VoIP Fax Line
Full Fax Support, including Fax over IP/Modem Detection & Fax Relay* – External Fax Machine Required.
WAN/LAN Bridging, Routing and Firewall Functions

Purchase the Uniden UIP1868 Now at VoIP Supply.com For Only $239.99

Cordless VoIP Phones – The Conclusion

There is little doubt that in order for VoIP subscriptions to continue to grow in the residential and small office sectors, VoIP service providers are going to need to support and embrace cordless VoIP phones. Although, only a handful of providers currently support them, consumer demand and product advancements are forcing more and more service providers to support and offer this solution. Cordless VoIP phones may not be the end all be all of customer premise hardware, but the benefits are un-deniable, and their popluarity soaring. 2007 could very well be the year cordless VoIP phones become as mainstream as analog cordless phones. Prior to make any decisions on a cordless VoIP phone make sure to contact your VoIP Service provider to ensure that they support the cordless VoIP phone that you are interested in using.

Linksys WIP320 – The Lastest Wi-FI VoIP Phone For Skype

January 3, 2007 by Garrett Smith

Replicating the Traditional Phone

One of the majors factor hindering the adoption of Skype by the North American market has been it’s reliance on being “at the computer” in order to use the service. Over the past year, numerous manufacturers have released Wi-Fi phones for use with Skype, bringing the promise of an old world calling experience to the new world of Skype calling. While it is still yet to be determined what impact the mass release of Wi-Fi phones for Skype has had on the adoption rates in North America, one thing is for certain: the Linksys WIP320 is the most advanced Wi-Fi phone for Skype to date.

The Linksys WIP320 – Makes Skype Calling Easy

The Linksys WIP320 combines sleek styling with robust functionality to deliver one of the most convient Skype user experiences. With it’s 1.8in TFT, color lcd display, the Linksys WIP320 makes it easy to find, select, and call all of your favorite Skype users. Because it is 802.11b/g compliant, the Linksys WIP320 is easy to register to just about any wireless access point or hot spot, giving your true mobility, and a calling experience closer to that of a traditional cordless phone. Unlike most Wi-Fi phones for Skype, the Linksys WIP320 is designed with only the features necessary to be successful in making calls – eliminating complex menu’s, configurations, and features which often lead to less then pleasureable experiences.

Aggressively Priced, Great Value

The Linksys WIP320 is aggressively priced at $169.99. When compared with other Wi-Fi phones for Skype, the WIP320 is well within competitive price ranges, and when compared with other Wi-Fi VoIP phones, it is one of the best VoIP deals on the market today. When you take into consideration that you can now get unlimited SkypeOut for $14.95 a year, SkypeIn for an additional $40, you can have a complete, unlimited home calling solution for less then $19 per month including the cost of hardware. What an incredible deal!

For more information about the Linksys WIP320, please vist the Linksys WIP320 product page at VoIP Supply.

Has VoIP Gone Mainstream?

December 29, 2006 by Garrett Smith

I Don’t Think So

Ken Camp points to a CIO today article on VoIP in the Enterprise as validation of his claim that VoIP has gone mainstream. While I agree that VoIP has emerged from relative obscurity this year, I do not think VoIP has gone mainstream as a form of business or residential communication. Maybe this has something to do with a difference of opinion when it comes to what mainstream means. Let me explain…

So What Does Mainstream Mean?

According to Wikipedia, Mainstream is, generally, the common current of thought of the majority. It is a term most often applied in the arts (i.e., music, literature, and performance). This includes:

* something that is ordinary or usual;
* something that is familiar to the masses;
* something that is available to the general public.

Using this definition, it is obvious that there is a lot of grey area surrounding what mainstream means, but with grey area comes debate and a need for further examination!

Here is Why VoIP is Not Mainstream Yet

To start, VoIP is still not ordinary or usual as a means of business or residential communication. With only 7.9% of US households using VoIP, and less then 25% of small medium business using VoIP, it is obvious that VoIP is gaining momentum, but it is certainly not an ordinary or usual means of communication for the majority of individuals. In fact these percentages suggest that right now VoIP is unusual!

I do not have up to the minute statistics on how familiar VoIP is to the masses, but earlier this year, there were reports that 50% of Americans surveyed knew of or heard about making telephone calls over the Internet. Knowing of and hearing of, I suppose, is a form of familiarity, but is 50% really enough to characterize true mainstream familiarity? I mean how many people know what an iPod is?

Yes, VoIP is available to the general public, but only the general public that has broadband Internet access. Currently only 52% of US households have broadband Internet access. Is 52% of all households enough to consider VoIP as available to the general public? Yes and No. I believe VoIP truly is available to the general public, as there are no regulated restrictions on the service, but from a purist point of view, if only half the people can actually use the service, well then, quite possibly, some would say it is not.

2007: The Year VoIP Goes Mainstream?

There is no doubt that VoIP and business VoIP has come a long way in 2006. I believe the advances that have been made this year and the momentum of the industry will make 2007 the true banner year for VoIP. The biggest challenge facing VoIP and its mainstream ambitions are the fact that yes VoIP scales, yes VoIP is reliable, but not all of the time. The “all of the time” factor is what is keeping VoIP from truly being a mainstream service. Think about other true mainstream devices, gadgets, services, etc. Are they “all the time” or just “most of the time”? Until VoIP can prove to majority that it is an “all the time” service, it will continue to be an emerging niche holding the promise of mainstream viability.

What is VoIP?

December 6, 2006 by Garrett Smith

Let’s go back to the basic to tell you what VoIP is. A simple explanation is: VoIP is just like your traditional analog phones but it’s making phone calls over the Internet.

What is VoIP?

VoIP stands for Voice over Internet Protocol. The “Voice” part of Voice over Internet Protocol is self-explanatory. The Internet Protocol is where some start to get confused. An Internet Protocol (IP) is a data-oriented protocol used for communicating data across a packet-switched network (home, office, the internet). Internet Protocols (IP) allow you to transmit data any interconnected networks. Utilizing a combination of hardware, software, and voice protocols, your conversations are carried over the IP network to its intended destination.

Internet Protocols (IP) allow you to transmit data any interconnected networks. Utilizing a combination of hardware, software, and voice protocols, your conversations are carried over the IP network to its intended destination.

In Layman’s terms: VoIP allows you to make phone calls over any interconnected network, mostly commonly thought of as the Internet, instead of using traditional analog PSTN (public Switched Telephone Network) lines.

Watch the 3-minute video below to learn what it is all about:

Now you know what VoIP is, but why do people need VoIP instead of just using traditional analog phones? Read our blog to learn: Why Switch to VoIP?

VoIP Supply at Internet Telephony Expo

September 29, 2006 by Garrett Smith

VoIP Supply will be hosting a partner pavilion at the Fall Internet Telephony Conference and Expo, October 11th through the 13th in Classy San Diego, California. In speaking with the folks at TMC, I am confident that this will be the best VoIP Expo of the year.

VoIP Supply’s Partner pavilion will feature products and services from the following companies:

In addition to these partners, VoIP Supply will also have the products from the following manufacturers on display:

You will also have the opportunity to purchase numerous different products at the show. Make sure to stop by the VoIP Supply Partner Pavilion at the IT EXPO show!

New Sangoma Technologies Pricing

September 19, 2006 by Garrett Smith

We have recently adjusted our pricing on the Sangoma Technologies line of Digital PCI Cards for use with Asterisk! Sangoma Technologies is one to follow for performance, quality and innovation. Their years of experience as a supplier of connectivity hardware and software products for Wide Area Network (WAN) and Internet infrastructure have provided them with a head start in hardware technologies that has enabled us to maintain leadership and keep the competition wallowing in our wake. Here is a snapshot of the Sangoma Technologies product line and our new prices:

Sangoma A 101 – $499.99 – The Sangoma Technologies A101 Single T1/E1 interface card is next generation hardware for use with Asterisk. The Sangoma A101 supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI, block mode raw bit-stream interface for integration with the Asterisk Open Source PBX/IVR platform, and Channelized mode supporting individual DMA into voice timeslots plus onboard HDLC support of PRI channel for soft PBX implementations that can use these features.
Sangoma A 102 – $849.99 – The Sangoma Technologies A102 Dual T1/E1 interface card is designed for optimal support of over voice and data over T1/E1. Like the Sangoma A101, the Sangoma A102 supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI, block mode raw bit-stream interface for integration with the Asterisk Open Source PBX/IVR platform, and Channelized mode supporting individual DMA into voice timeslots plus onboard HDLC support of PRI channel for soft PBX implementations that can use these features.
Sangoma A 104 – $1,449.99 – The Sangoma Technologies A104 Quad T1/E1/J1 interface card is an updated, quad port version of Sangoma’s range of advanced, flexible telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. Based on bus mastering PCI technology supported by a ring-buffer DMA architecture, the A104U provides full speed 132 Mbps PCI bus transfer with minimal real-time processor load and fully supported interrupt sharing. This provides optimal performance in demanding environments such as soft PBX/IVR voice applications.
Sangoma A 104D – $2,249.99 – The Sangoma Technologies A104D is the, quad port version of Sangoma’s range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The Sangoma A104D provides full speed 132 Mbps PCI bus transfer with FPGA and optional DSP based processing to unload the host CPU in demanding environments such as soft PBX/IVR voice applications. Compatible with both the 3.3v and 5v PCI bus, A104 cards operate in all commercially available motherboards sharing IRQs properly with themselves and all other PCI compatible devices, so you never have to worry about hardware compatibility issues.
Sangoma A 108 – $2,599.99 – The Sangoma Technologies A108 is the eight port version of Sangoma’s range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The Sangoma A108 supports voice enhancement capabilities including G.168-2002 echo cancellation with 1024 tap/128ms tail per channel on all channel densities, DMF encoding/decoding and tone recognition, voice quality enhancement and adaptive noise reduction.
Sangoma A 108D – $4,699.99 – The Sangoma Technologies A108D is especially built for the soft telephony industry. The Sangoma A108D supports voice enhancement capabilities including G.168-2002 echo cancellation with 1024 tap/128ms tail per channel on all channel densities, DMF encoding/decoding and tone recognition, voice quality enhancement and adaptive noise reduction.

Sangoma’s PCI architecture (of course with autosense 3.3/5v support) has superior performance and compatibility simply because the family approach means that we only ever have to solve an interface problem once. If you don’t enjoy experimenting with different motherboards, then this family is for you.

When it comes to quality, there are no compromises. Sangoma may not always have the lowest price, but they will always have the highest performance/cost ratio. Sangoma always chooses the best of breed for integration into their products. As an example, Sangoma Octasic-driven optional hardware echo cancellation is acknowledged even by the competition as being superior to anything else on the market.

Meet Our Newest Addition to The VoIP Insider!

Skype Announces New Calling Plans

Polycom IP430 IP Phone Review

Linksys WIP300 Wi-Fi IP Phone Review

Cordless VoIP Phones The New VoIP Trend at CES

Linksys WIP320 – The Lastest Wi-FI VoIP Phone For Skype

Has VoIP Gone Mainstream?

What is VoIP?

What is VoIP?

VoIP Supply at Internet Telephony Expo

New Sangoma Technologies Pricing

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