Polycom SoundStation IP 7000 FAQ

What is the SoundStation IP 7000?
The SoundStation IP 7000 is a high-performance IP conference phone ideally suited for offices, conference rooms, and board rooms. It features phenomenal audio quality, with Polycom HD Voice technology at 22 kHz, along with advanced IP functionality and a number of different flexibility and expandability options.

What is Polycom HD Voice technology?
Polycom HD Voice delivers much clearer, more vibrant and life-like conversations than traditional phone technology. It combines wideband codecs with our patented Acoustic Clarity Technology 2 into a complete, integrated design to maximize the audio performance of your phone. You can learn more at www.polycom.com/hdvoice.

What does “22 kHz” mean on the SoundStation IP 7000?
22 kHz refers to the high end of the frequency response that the SoundStation IP 7000 is capable of delivering. The 22 kHz frequency response is nearly CD quality audio, and is even greater than the capabilities of the human voice. The SoundStation IP 7000 is the first conference phone ever developed that has been capable of audio performance at up to 22 kHz.

Will every call on the Polycom SoundStation IP 7000 be in HD?
There are a number of factors in addition to the conference phone that determine the quality of the call. For example, calls to a traditional “narrow band” phone will be limited to the lower capabilities of that traditional phone. In addition, the IP PBX or hosted IP telephony service that the phone is connected to will also determine the audio quality of the call. Check with your IP PBX or hosted service provider for more information on what types of wideband, or HD Voice, services are supported.

My IP PBX only supports wideband audio codecs that offer 7 kHz voice quality, so what are the benefits of the 22 kHz capabilities of the SoundStation IP 7000?
Even though your IP PBX supports 7 kHz wideband codecs today, it may support higher quality codecs in the future. Purchasing the SoundStation IP 7000 provides investment protection and security in knowing you have purchased a platform that can support these higher quality codecs. The phone also matches the 22 kHz capabilities of Polycom HDX for future integration.

Is HD Voice on the SoundStation IP 7000 compatible with the IP 550 and IP 650?
Yes, HD Voice calls can be established between those products. Check with your IP PBX or hosted service provider to verify support for HD Voice.

What if I call another phone that does not support HD Voice?
The call will be in narrowband, traditional phone quality if the other phone or audio conferencing service does not support wideband capabilities. Even in narrowband calls, the SoundStation IP 7000 offers the industry’s best narrowband audio quality on a conference phone.

Can multiple SoundStation IP 7000 units be connected together?
Yes, up to two of these conference phones can be connected, with all microphones and speakers active, and the call controlled via either phone.

Can additional MICS be added to two IP 7000s that are connected together?
Yes, up to two expansion microphones can be added to this configuration. The microphones on both the phones and expansion microphones will be active during a call.

Can the expansion microphones also be added to a single SoundStation IP 7000?
Yes, up to two expansion microphones can be added to a single phone.

Will the IP 4000 Mics work on the SoundStation IP 7000?
No, the expansion microphones for the SoundStation IP 7000 were specifically designed for the high-fidelity capabilities of the phone, and expansion microphones from other models cannot be used with it.

Can the SoundStation IP 7000 be connected to any video conferencing systems?
Later in 2008, the conference phone will be able to integrate with the Polycom HDX line of high-definition video conferencing systems. More details will be available at that time.

Wideband VoIP Codecs

L256: The simplest of all wideband codecs, the 7 kHz L256 directly sends all the bits of digital audio sampled into 16-bit words at 16 kilo samples per second (ksps), using no compression whatever, hence the name (“Linear 256” ksps). L256 is a basic requirement in all VoIP phones, but is seldom used because of its high bit rate.

G.719: Perhaps the best match among requirements for communication systems at 20 kHz, G.719 is a recent ITU-approved arrival that combines excellent quality for music and voice with low latency, modest processor load, and network-friendly bit rates.

G.722: This is the grandfather of 7 kHz wideband VoIP codecs, and the most widely deployed so far. G.722 applies adaptive differential pulse code modulation (ADPCM) to high and low frequencies separately, yielding an algorithm that works equally well with music or voice.

G.722.1: Also known as “Siren 7,” this modern 7 kHz audio codec is in almost every videoconferencing system today and is gaining traction in VoIP because of its higher efficiency and lower bit rate. G.722.1 is a “transform” (as in “Fourier transform”) codec and works by removing frequency redundancies in any kind of audio.

G.722.2: This codec, “AMR-WB,” is a 7 kHz wideband extension of the popular adaptive multi-rate (AMR) cellphone algorithm, and excels in delivering wideband high-quality voice at the lowest bit rates. G.722.2’s algebraic code excited linear prediction (ACELP) algorithm is optimized for speech, and works by sending constant descriptions of how to shape and stimulate a human speech tract to reproduce the sound you feed into it.

G.722.1: Annex C. Also known as “Siren14,” this is a 14 kHz extension of G.722.1 and is popular because of its wider bandwidth, its efficiency, and its availability (under license) for zero royalty.

Speex: Speex is an open-source CELP codec. MPEG. There are more than 25 versions of the moving pictures expert group (MPEG) transform codecs, each delivering a set of performance levels optimized for various parameters. The variant best suited to telecommunications is MPEG4 AAC-LD, a lower-delay version of the intended MP3 successor, MPEG4 AAC.

MP3: The popular MP3 format uses a form of transform coding, and is optimized for media distribution.

FLAC: The Free Lossless Audio Codec (FLAC) produces much higher bit rates than most other codecs, but compensates by preserving complete audio quality.

SIP Trunking for the Switchvox

Did you know .e4 offers SIP Trunks for your Switchvox System?

There are many VoIP products in the market today- most of which offer the promise of reduced cost, better call control, and of course actual compatibility with Asterisk based products and services.

Recently, .e4 launched a new product that meets the need of residential customers as well as the demanding need of office environments. This SIP Trunking product has been tested extensively with Switchvox. Our base package gives Switchvox SOHO, and SMB users an unlimited number of talk paths at one penny per minute in 6 second billing increments.

SIP Trunk Setup in Switchvox is simple. Quite literally we can have your outbound and inbound calls running through our SIP service and in to your happy ear in less than 2 minutes.

The VoIP Users Conference

If you’ve been hanging around twitter on a Friday you’ve probably seen the Asterisk/VoIP tweeters buzzing about the VoIP Users Conference. The VUC as it is often referred to by regulars is a unique way for Fans of Asterisk and other VoIP/IP Telephony technologies to get their geek on.

About the VUC:
The VoIP Users Conference is a weekly live discussion about VoIP, SIP, Asterisk and all kinds of telephony-related topics. The conference has been running for over two years.

Randy Resnick aka Randulo/Zeeek @randulo
Michael Graves – @mjgraves

SIP Trunking

The continuing advance of networking technology is enabling new and better forms of communication, but also adding complexity to the process. Companies must maintain flexibility in a changing market, while also providing opportunities for growth. Older technologies need to be maintained, while new and improved capabilities are implemented. And of course, costs must be controlled, as employees come up to speed on the latest communication options.

SIP Trunking Defined:
Protocols are conceptual models that let applications exchange data through a communications network. SIP, or Session Initiation Protocol, is an application-layer standard for creating, modifying and terminating Internet Protocol (IP) sessions with one or more participants. SIP has its roots as a protocol for bridging traditional analog telephone networks and IP networks, but it can also support a number of emerging technologies, such as VoIP, presence management and Instant Messaging. These technologies allow information to be converted into a common format, so many kinds of devices can exchange information. Essentially, SIP lets different kinds of IP traffic share the same network connections, which opens new possibilities for integrating voice with other communication options.

Trunking is a communications concept that lets multiple users share network assets by defining access rules for lines, frequencies or bandwidth. A SIP Trunk can support multiple users with voice calls, conference calls, multimedia distribution and other features. SIP Trunking actually offers a number of inherent advantages over traditional telephony. SIP connections can support very high audio quality, and with compression algorithms, can fit more calls within a given amount of bandwidth. Transmitting call-related information (such as caller ID) is easy, and calls can even be enhanced with new features, such as pictures of the caller. With a SIP connection, a telephone number isn’t limited to ten digits, so Direct Inward Dialing could be implemented for every extension.

Why SIP Trunking Matters:
The trend toward converged networks means that SIP-supported communications are the way of the future. Companies can use their IP-enabled networks for both voice and data, making the most of their high-speed Internet connections. They can converge local, long distance, toll-free, private voice and data traffic onto an MPLS IP-based network. Years of investment in data networks can therefore deliver an additional return, when voice and data traffic are moved to a single network platform.

SIP also allows for dynamic bandwidth allocation, ensuring that voice traffic is given the highest priority – calls always go through. There’s no need to pre-allocate space to fixed voice channels, as long as enough total bandwidth is available. That means SIP supports highly efficient use of network resources, allocating bandwidth based on the needs of the moment. Companies can also increase the performance of their call centers by using SIP to enable new routing, tracking and call transfer functionality. SIP supports a number of refer-and-redirect capabilities, which can be used to define communications-enabled business processes. Intelligent routing options can improve process efficiency and increase customer satisfaction. SIP-enabled VoIP systems can also support “Next Generation” contact centers, in which voice communication is integrated with digital communication, supporting features like Web-based “click to call” options. SIP is based on an open standard, and can be the foundation for building new and innovative services. SIP standards add value and functionality, with extensions that cover data compression, messaging, security and call control. SIP allows for the potential to define new services suggested by end users and companies alike, and simplifies interoperability in a multi-vendor environment. SIP can support a number of advanced communications capabilities. For example, hotels could deliver ads for local businesses or restaurants to the screens of SIP-enabled telephones, and also track room availability. Lawyers could track time automatically, with call-logging systems feeding information directly to billing and account management.

A key advantage is that SIP Trunking lets companies implement VoIP technology at their own pace. SIP Trunks can connect existing key systems and TDM PBXs to VoIP networks, allowing for a phased migration. The SIP standard also makes for easier interoperability. Companies can manage their infrastructure investments while migrating to business class VoIP.

Finally, SIP trunking can reduce the Total Cost of Ownership (TCO). By converging voice and data networks, companies can reduce access costs, improve bandwidth utilization and bring down operational expenses. Routine tasks (such as Moves, Adds and Changes) are quicker and easier in a VoIP environment. VoIP networks can be easier to manage than legacy systems, and can often be controlled remotely, using Web-based tools.

Business Implications:
Implementing a SIP Trunk as a road to VoIP offers definite business value, but the specific value delivered depends on the size of the enterprise. Large companies can save a significant amount of money by converging their voice and data networks, particularly if they’re covering multiple locations. Smaller companies can chart a phased migration path toward next-generation technologies without a large up-front investment. Companies of any size will benefit from features like dynamic bandwidth allocation.

The VoIP marketplace today is divided between equipment vendors and service providers, and many companies feel pressured to act sooner rather than later. SIP Trunking services are a good way for companies to experiment with VoIP capabilities while using their existing equipment, eliminating the need for significant capital investment and extensive user training programs. This ability to implement the change gradually is a way to reduce TCO, something that’s often overlooked in purchasing decisions.

Looking Forward:
In the future, SIP will dominate telecommunications, as carriers establish peer relationships to expand on-net calling. SIP applications will become more intelligent, learning how to prioritize messages, adapt to changing user preferences and ensure privacy.

SIP Trunking delivers tangible benefits in cost reduction, productivity and stronger business interactions, in both a B2B and a B2C context. Although still a new concept for many companies it already offers a broad range of features and significant business benefits.

.e4 offers a comprehensive suite of SIP trunking products through our .e4 SIP Portal. Signup today and receive a free credit and install support for you PBX.

VoIP Security Audit

VoIP is about convergence, saving money and resources.  However, these types of systems also create unique inroads for attack.  As VoIP has become more accessible and popular, security threats to VoIP networks have grown .e4 has worked with countless companies to address and consolidate the VoIP security threat.

Contact .e4 for a comprehensive VoIP security audit today.

Introducing the Aastra 6739i Color Touch Screen VoIP Phone

Aastra, a leading company at the forefront of the enterprise communication market, resets the standard for premium IP phones with the launch of the Aastra 6739i, its most advanced desk phone to date. The company unveiled the new color touch screen phone today to hundreds of telecom resellers and service providers at the BroadSoft Connections executive conference.

In an increasingly competitive business climate, executives need access to the highest quality communication technology available to facilitate personal communications and collaboration. The feature-packed Aastra 6739i is a market-leading touch screen business desk phone, delivering high quality enterprise communications with advanced features, such as Bluetooth and dual Gigabit Ethernet.

The Aastra 6739i offers an exceptional user experience with its large, high quality color touch screen combined with an intuitive interface and navigation menus. For unrivalled voice clarity, the Aastra 6739i brings high definition sound with Aastra Hi-Q™ audio technology combined with full wideband handset and speakerphone hardware.

Extending Aastra’s commitment to interoperability, the 6739i is part of Aastra’s 67xi SIP desk phone series. These are designed to integrate and deploy easily with Aastra’s own IP systems as well as all leading SIP compatible IP call managers, making them appeal to a wide range of SMB and enterprise customers.

“Business communications needs have evolved, and executives are demanding the best features and usability without having to make a huge up-front investment. The Aastra 6739i offers high definition audio quality and a full range of functions to a market that is continuing to demand best-in-class technology at a competitive price.” said Simon Beebe, Vice President of Product Management, responsible for SIP phones at Aastra.

Aastra 6739i Features:

  • Large 5.7inch full VGA (640×480) color touch screen display
  • Intuitive graphical user interface and navigation menus
  • Integrated Gigabit Ethernet ports
  • Bluetooth headset support
  • USB port and expansion modules support
  • Aastra Hi-Q™ audio technology
  • Full wideband response hardware: handset, headset port and speakerphone
  • Additional headset connection options: modular RJ jack, built-in EHS/DHSG port
  • Up to nine lines call appearances with multi-proxy support
  • Up to 55 programmable softkeys
  • XML support for productivity-enhancing applications

The Aastra 6739i can be pre-ordered now for shipping before year end. For more information on this new product please contact 877.434.8647

Power Over Ethernet Myths- The Facts about PoE

Power-over-Ethernet (PoE) technology integrates power and data across standard Cat5/5e/6 network cabling and provides more flexibility in today’s workplace. PoE enables power to be supplied to network devices, such as IP phones, network cameras, and wireless access points through a single, most often existing, network cable. When combined with an uninterruptable power supply (UPS) a PoE network delivers continuous operation and minimizes business downtime by eliminating most power interruptions. With the ability to install endpoints in any location PoE technology provides a scalable and flexible networking infrastructure geared for growth and efficiency.

Myth 1: PoE Switches can provide all the power I need or will need.

Today most switches are merely PoE-enabled. This means the majority rely on power management to share available power across the switch ports. The switches are designed with a smaller power supply that is typically capable of powering the switch itself and providing the required 15.4 watts of power over a limited number of ports.

For example: A 24-port PoE Switch with power management typically has a 195-watt power supply. After the 40 watts needed to power the switch, you have approximately 155 watts remaining. If 12 of the 24 ports are used to connect end devices using 11.5 watts each, you would only have 17 watts remaining to provide power on the last 12 ports. The math doesn’t match the ports: 195W – 40W (switch) – 138 (12 devices @ 11.5W/ea) = 17W left for power on 12 ports

Myth Busted: A PoE Switch is often not the best and most cost effective solution.

Myth 2: A midspan and a PoE switch are the same.

A PoE Midspan is not a switch. A Midspan is an additional PoE power source that can be used to offer full power to all endpoint devices. PoE Midspans (Power Hub or Power Injector) pass data from a switch and ‘inject’ safe power acting as a patch panel of sorts. Midspans are commonly used with either a non-PoE switch, an existing PoE switch, or a new PoE switch in a network. In addition to offering full power across all available ports, midspans costs substantially less per port and overall than a new PoE enabled switch.

Myth Busted: Midspans do not switch – they make use of existing best-in-class switches. They inject safe power across all ports and cost less than PoE switches.

Myth 3: Only a switch that has PoE built in should be used to power devices like IP Phones, Access Points, and IP Security Cameras.

Switches were designed to, well, switch. PoE Switches are designed with power management and have to distribute different power as required to ports but there is often not enough power for all devices plus the power required to complete the primary task – switching. Networks that have multiple devices like IP phones, IP cameras, wireless access points quickly go beyond the limited capacity of managed power PoE switches. As more PoE devices continue to grow in capabilities and market share this managed power limitation will become more and more evident. Midspans, in contrast to switches, were designed to provide full power on every port and deliver safe and reliable power based on the industry standards (IEEE802.3af/at).

Myth Busted: Rather than relying on power management in a switch use a midspan that can deliver full power (15.4W) to every port for all PoE-enabled devices now and in the future.

Myth 4: Ethernet devices not PoE-enabled (non 802.3af/at compliant) cannot be powered using PoE technology.

Many devices do not directly accept Power-over-Ethernet but can still use PoE technology. If the device uses less than 12.5 watts (802.3af) or less than 50 watts (802.3at+) and connects to an IP Ethernet network you can use a PoE splitter. PoE splitters enable you to accept PoE power from any IEEE 802.3af/at compliant switch or midspan then separates the data and power on to two seprate cables. The data is connected to the end device through a standard RJ45 plug while the power is connected using a standard 5.5 x 2.1 x 12mm Adapter Plug. Splitters can also convert the input voltage to the required voltage for a non-PoE device. Splitters are traditionally used with older network products which only accept power through their (DC) jack and data through their RJ-45 jack.

Myth Busted: PoE splitters can be used in conjunction with PoE midspans and switches to provide both the data connectivity and power required by most endpoint devices.

Myth 5: I need/will need additional PoE switch ports to power my IP cameras and high-power pan, tilt, and zoom (PTZ) cameras.

Today, many devices have evolved into more advanced solutions with higher power requirements. The traditional approach was to endure a “forklift upgrade”. This meant buying new PoE switches at considerable cost and physically swapping out the existing switches to meet higher power requirements or add more powered ports. There is an easy and more cost-effective way – separate the data and power in the wiring closet (IBF). It is more efficient and costs less to separate your data and power allowing you to keep your best-in-class business switch for your IP needs and supplement it where required with best-in-class midspan technology to power the endpoints.

Myth Busted: A PoE Switch is often not the best and most cost effective solution.

Myth 6: All midspans are created equal . . . they are all the same.

Always select a best-in-class midspan. If you wanted to enhance your switched network wouldn’t use a best-in-class network switch? Of course you would. A midspan designed and manufactured by a leading power supply company that understands power, power requirements, and one that delivers enterprise-level solutions.

Select a midspan manufacturer that has multiple members on the IEEE (PoE) committee helping to define safe, new PoE standards. This ensures that every midspan is designed to meet current and future IEEE specifications for Power-over-Ethernet.

Select a midspan manufacturer that designs, manufactures, and tests its own product rather than outsourcing these tasks across the globe to cut costs.

Select a midspan that has a high-speed, common interface to access the management console. A USB port is not as cheap as a serial port (RS-232) but is faster, more user-friendly, and more common on high quality midspans.

Myth Busted: Although there are many midspan manufacturers out there, few have the power supply experience, quality controls, and manufacturing capability to produce best-in-class midspans. All midspans are NOT created equal.

Contact .e4 Today- Our trained team of PoE Specialists are standing by to help you make an educated decision:

Setup Guide: Switchvox Voicemail System with Polycom SoundPoint IP Phones

When setting up your new Polycom IP Phone for the first time you should start by logging in to the phone by selecting the Messages Key on the right hand side of your phone or by entering 899 on the number pad. This will log you in to the Switchvox message center.

Setting up your Unavailable Message:

1. Press the Messages key or dial 899
2. Enter your Password (By Default this will be your extension number)
3. Press 0 (Zero) for “Mailbox Options”
4. Press 1 (One) to record your “Unavailable Message”
5. Record Your Message After the Tone Press # (Pound Key)
6. Follow Instructions – 1= Save, 2 = Listen, 3 = Re-Record

Setting up your Busy Message:

1. Press the Messages key or dial 899
2. Enter your Password (By Default this will be your extension number)
3. Press 0 (Zero) for “Mailbox Options”
4. Press 2 (Two) to record your “Busy Message”
5. Record Your Message After the Tone Press # (Pound Key)
6. Follow Instructions – 1= Save, 2 = Listen, 3 = Re-Record

Setting up your Recorded Directory Name:

1. Press the Messages key or dial 899
2. Enter your Password (By Default this will be your extension number)
3. Press 0 (Zero) for “Mailbox Options”
4. Press 3 (Three) to record your “Name”
5. Record Your Message after the Tone Press # (Pound Key)
6. Follow Instructions – 1= Save, 2 = Listen, 3 = Re-Record

Please visit our website to learn more about Digium products, Switchvox 4.5, Polycom VoIP Phones, and what .e4 Technologies can bring to the table for your business. On the web @ http://e4strategies.com

Digium news – Digium Positioned in the Visionaries Quadrant of Leading Analyst Firm’s Corporate Telephony Magic Quadrant

Wednesday, August 13, 2008

HUNTSVILLE, Ala. — Digium(R), Inc., the Asterisk(R) Company, today announced it has been positioned in the Visionaries quadrant of Gartner’s Magic Quadrant for Corporate Telephony(1) report. The evaluation is recognition by one of the world’s best-known research firms and is further proof of Digium’s undisputed leading role in the open source telecom industry, according to the Huntsville, Ala.-based Digium.”Digium’s founder Mark Spencer had the vision to create Asterisk, which today is the world’s most widely used open source telephony software, so we believe the Visionaries quadrant is a perfect fit for Digium,” said Danny Windham, chief executive officer of Digium. “We think the Gartner report highlights Digium’s unique role as the innovative force behind Asterisk and the company more businesses are turning to as an alternative to the high-cost, low-flexibility telecom giants of the past.”Digium was founded in 1999 and has rapidly grown to challenge larger, long-established competitors. Asterisk allows companies of all sizes in any industry to adopt corporate phone systems that are easier to customize and cost a fraction of traditional proprietary systems. The market for products and services based upon open source telephony and specifically Asterisk-based solutions is gaining momentum worldwide. Digium created, owns and leads the ongoing development of Asterisk, the world’s most successful open source communications project.

About the Magic Quadrant

The Magic Quadrant is copyrighted 2008 by Gartner, Inc. and is reused with permission. The Magic Quadrant is a graphical representation of a marketplace at and for a specific time period. It depicts Gartner’s analysis of how certain vendors measure against criteria for that marketplace, as defined by Gartner. Gartner does not endorse any vendor, product or service depicted in the Magic Quadrant, and does not advise technology users to select only those vendors placed in the “Leaders” quadrant. The Magic Quadrant is intended solely as a research tool, and is not meant to be a specific guide to action. Gartner disclaims all warranties, express or implied, with respect to this research, including any warranties of merchantability or fitness for a particular purpose.

About Digium

Digium(R), Inc., the Asterisk(R) Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company’s product line includes a wide range of hardware to enable resellers and customers to implement turnkey solutions or to design their own voice over IP (VoIP) systems. More information is available at www.digium.com.

The Digium logo, Digium, Asterisk, Asterisk Business Edition, AsteriskNOW, Asterisk Appliance and the Asterisk logo are trademarks of Digium, Inc. All other trademarks are property of their respective owners.

(1) Magic Quadrant for Corporate Telephony; Steve Blood, Geoff Johnson, Jay Lassman and Rich Costello; Aug. 8, 2008.

Schwartz Communications John McElhenny, 781-684-0770 digium@schwartz-pr.com or Digium Jane Brooks, 256-428-6075 jbrooks@digium.com