Quick Guide to PoE (Power Over Ethernet)

May 5, 2006 by Garrett Smith

The RJ45 (CAT5) cabling that connects your office or home networking can be utilized for more than transporting voice or data across your local area network (LAN). The IEEE 802.3af Power over Ethernet standard was created as the standard method of delivering power to hardware devices such as IP Phones, IP Video Cameras and other network equipment. With PoE, there is no need to power some devices locally with an external AC power supply, they can simply receive their required voltage over the network cabling.

A POE system is comprised of three elements, the power sourcing equipment (PSE), the device which needs to be powered (PD) and the cable.

The Power Sourcing Equipment (PSE) is connected to a device, and determines if the device is IEEE 802.3af compliant or non-compliant. If determined to be a non-compliant device, no electrical current is passed through the network cable to the device. If the device is determined by the PSE to be a PoE compliant device, the required voltage is supplied to the device to power it.

There are several varieties of Power Sourcing Equipment to fit your specific requirements.

Power Sourcing Equipment (PSE)

Single Port (RJ45) PoE Injector – designed to provide PoE to a single device, a single PoE Injector has (2) RJ45 network connections. The network cable is plugged into the PoE Injector, where the necessary voltage is added. A second ethernet cable runs out of the PoE Injector and is connected to the device to be powered. Some examples of Single Port PoE Injectors include thePW130, and the Linksys WAPPOE”6

Mid-Span PoE Hub

Midspans come is port spans typically ranging from 4-24 ports. A Midspan PoE Hub does not have ethernet switching capability, it is designed to be paired in tandem with an existing ethernet switch, adding voltage for IEEE 802.3af compliant devices as needed. Midspans are a good choice for businesses that have existing 10/100 Ethernet switches already installed, and are less expensive than replacing legacy switches with newer, PoE enabled switches. The Midspan hub is typically stacked on the ethernet switch and each network cable is fed into the hub, where power is added, and then cabling run out to each specific network device. Some examples of Midspan hubs include the SEI Juicebox and a variety of products from PowerDsine.

PoE Switch

Many manufacturers offer ethernet switches with IEEE 802.3af PoE Midspan functionality built in. PoE Switches typically come in port span increments from 12-48 ports, and offer both 10/100/1000 Ethernet managed switching capabilities and PoE injection. Some examples of PoE Switches include the Linksys SRW224P and the Edgewater 2402

Benefits of Power Over Ethernet?

  • Easy, fast and convenient to install
  • Reduces overall power consumption
  • Reduces installation costs
  • Reduces cable runs
  • IEEE 802.3af is a unified, worldwide standard
  • It will save you money

Polycom Communicator C100S – Another Cool New Product for Skype

Recent news from Ebay executives peg the current number of Skype users in the US at 6 million, and growing fast. The company claims more than 100 million users worldwide, but until recently, the majority of Skype users were overseas in Asia and Europe. With rapidly growing usership in the US market, many major electronics vendors are rushing to bring “Skype Certified” products to market.

Polycom, well known for audio and video conferencing, is targeting business customers who utilize Skype service, with its latest product, the Polycom Communicator C100S.

The Communicator is a lightweight, portable speakerphone offering a standard USB connection to a PC or laptop. The Polycom Communicator is a “Skype Certified” product, meaning it has undergone standard interoperability testing in cooperation with Skype, and is officially sanctioned for use with Skype service.

With a retail price point under $150, the Communicator promises a superior user experience versus a standard microphone or headset. In addition to two microphones, the Communicator incorporates an echo cancellation function in the USB driver; in use, it disables Skype’s own echo cancellation. In addition to its primary function, Polycom says it can be used to play music and audio from the connected PC.

Polycom Communicator promises the ultimate hands-free Skype experience. Based on the same technology used in Polycom’s legendary line of triangular SoundStation conference phones, the Skype-certified Polycom Communicator enables crystal-clear, natural conversations when using Skype. Enjoy the freedom of not wearing your headset for hands-free Skype calls, or plug into the built-in stereo headphone port for private conversations.

The Polycom Communicator delivers high-fidelity wideband voice quality that sounds like you are in the same room with the people you’re calling. Two high-quality microphones provide excellent range for group conversations with up to four participants. Polycom’s Acoustic Clarity Technology eliminates echoes and feedback, maximizing your Skype experience.

RedFone Communications Introduces Dual T1 Version of their popular foneBridge T1 Interface for Asterisk

RedFone Communications has released a new, dual T1 version of their popular foneBridge solid-state T1 interface for use with Asterisk Open Source PBX.

foneBRIDGE is a T1 or E1/PRI-to-Ethernet Bridge. It is an integrated black-box “appliance” designed to streamline installation and enable redundant design of Asterisk based VoIP systems. Designing, building, and installing an Asterisk system has never been this easy!

foneBRIDGE eliminates the need to install proprietary TDM (PRI/T1) hardware cards in approved/compatible server configurations. Instead, foneBRIDGE terminates T1/E1 and/or PRI lines on the trunk side and provides direct Ethernet communication to a network of Asterisk servers using native Asterisk TDMoE formats and utilities.

foneBRIDGE provides an economical means of buliding and maintaining a redundant Asterisk solution. Instead of purchasing a TDM board for each redundant server, multiple redundant servers can share a single foneBRIDGE. If the foneBRIDGE itself fails, replacing it is far simpler that replacing a TDM card in an active server.

More from: Asterisk Garrett Smith

Linksys SPA-3102 Analog VoIP Adaptor Now Available at VoIPSupply.com

May 2, 2006 by Garrett Smith

VoIPSupply.com is now carrying the Linksys SPA-3102, the next generation of Linksys 3000 series telephone adaptors.

The SPA-3102 NA from Linksys is slated as the replacement for the popular SPA-3000, and offers both 1 FXS station side port and 1 FXO PSTN jack, and features integrated call routing for local and emergency calls, excellent for service providers and remote offices!

Phone Adapter + PSTN Gateway

The SPA-3102 NA features VoIP adapter functionality found in the SPA-2002 and SPA-1001 with the additional benefit of an integral connection for legacy telephone network “hop-on, hop-off” applications. SPA-3102 NA users will be able to leverage their broadband phone service connections more than ever by automatically routing local calls from cell phones and land lines to a VoIP service provider and vice versa.

A typical user calling from a land line or mobile phone will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA-3000 via a local phone number or by using a telephone connected directly to the unit. The advanced authentication and call routing intelligence programmed into the SPA-3102 NA will connect the caller via the Internet to the far end destination with security and ease. Using the SPA-3102 NA at the far end, calls can be answered immediately or further processed as a local call to any legacy land line or mobile phone allowed by the SPA-3102 NA dial plan.

If power is lost to the unit or the VoIP service is down, calls can be sent to a traditional carrier via the FXO interface.

VoIP Supply, LLC Launches Online Rebate Center

May 1, 2006 by Garrett Smith

VoIP Supply, LLC Launches Online Rebate Center
Rebate Center Features Exclusive Offers on Products from Industry Leading Vendors

BUFFALO, NY (May 1st, 2006) VoIPSupply.com a leading provider of Voice over IP hardware, software and services, today announced the launch of a rebate center for www.voipsupply.com. The rebate center features exclusive instant and mail-in-rebates on VoIP products from industry leading vendors. The rebate center is the result of increased vendor participation in VoIP Supply promotional initiatives.

Benjamin P. Sayers, President and CEO of VoIP Supply, LLC stated “The rebate center offers both consumers and vendors the opportunity to benefit from VoIP Supply’s market presence. Consumers realize greater cost savings, while vendors realize increase sales revenues through VoIP Supply rebate center promotion initiatives.”

The rebate center, which is located at www.voipsupply.com/rebate_center , will be updated weekly with both instant and mail-in-rebate offers on a large variety of products. The rebate center currently features offerings from industry leading vendors such as Polycom, Vegastream, and Zoom Technologies, with savings of up $500 per product.

New MP-11X Analog VoIP Gateways from AudioCodes Shipping

April 19, 2006 by Garrett Smith

AudioCodes has released a new series of flexible analog gateways, the MediaPack MP-11X series, in FXS/FXO spans ranging from 2-8 ports.

The MediaPack ™ Series Analog VoIP Gateways off affordable, feature-rich solutions for connecting legacy telephones, fax machines and PBX systems with IP-based telephony networks, and integrate seamlessly with new IP-based PBX architecture. MediaPack MP11X Series gateways are designed for interoperability with leading Softswitches, H.323 Gatekeepers and SIP servers.

New Models Include:

MediaPack Series (11x) Features Include:

  • Spans ranging from 2 to 8 analog ports
  • Selectable, multiple LBR coders per channel
  • T.38 compliant
  • Echo canceller, Jitter Buffer, VAD and CNG
  • Complies with MGCP, H.323 (v4) and SIP control protocols
  • Comprehensive support for supplementary services
  • Enhanced capabilities including MWI, long-haul, metering, CID and outdoor protection
  • Web management for easy configuration and installation
  • EMS for comprehensive management operations (FCAPS)


Linksys Ships PAP2T-NA Dual FXS Analog VoIP Adapter

April 13, 2006 by Garrett Smith

The Linksys PAP2T Internet Phone Adapter enables high-quality feature-rich VoIP (voice over IP) service through your broadband Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone ports to connect analogue phones or use one of the ports for a fax machine.

Each phone port operates independently, with separate phone service and phone numbers — like having two telephone lines. You’ll get clear reception and a reliable fax connection, even while using the Internet at the same time.

With Internet telephony, along with low domestic and international phone rates, an impressive array of special telephone features are available. Choose your preferred free local dialing US area code, regardless of where you live. Or add a virtual telephone number in any area code, forwarded to your Internet phone. You can even add a toll-free number.

The Linksys Internet VoIP Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephony service provider, such as Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and much more.

Features on the new Linksys PAP2T-NA Includes:

  • Two voice ports (RJ11) for analog phones or Fax machines with two independent telephone numbers
  • One RJ-45 port for 10/100 Mbps Ethernet connection
  • Supports Dynamic Host Configuration Protocol (DHCP)
  • Supports Session Initiation Protocol (SIP)
  • Supports multiple voice compression standards: G.711, G.726, G.729, and G.723.1
  • Supports Simultaneous calls with G.729 codec
  • Web-based configuration through a built-in web server
  • Supports DTMF tone detection and generation
  • Supports FSK Caller ID, DTMF Caller ID and FSK VMWI
  • Supports Echo Cancellation and Voice Activity Detection (VAD)
  • Password protected access and configuration
  • Supports auto-provisioning with remote firmware upgrade

Digium Announces New Hardware Products at VON

Digium Announces New Hardware Products at VON
New Transcoder and Echo Cancellations Cards Improve Call Quality and Communication between PSTN and VoIP Gateways

HUNTSVILLE, AL and SAN JOSE, CA — (March 14, 2006) – Source: Digium Press Release — Digium Inc., the creator of Asterisk™, and pioneer of open source telephony, today announced the availability of new hardware solutions to enhance Asterisk transcoding and echo cancellation performance for VoIP and PSTN gateways. These new products include the TC400P VoIP transcoding card and the TE420P and TE415P four-port T1/E1/J1/PRI cards with onboard hardware echo cancellation.

“Our product team is always working to develop solutions like these that ultimately further the open source movement in VoIP,” said Mark Spencer, president of Digium. “Not only are we constantly striving to improve Asterisk’s performance, but we also want to contribute to the overall VoIP experience, while keeping costs low.”

The TC400P provides hardware transcoding of VoIP codecs; decreasing Asterisk’s work load and providing improved CPU efficiency and an increase in channel density over a software-only solution. The TC400P provides Asterisk with full transcoding support and hardware acceleration for the G.723.1 and G.729A codecs.

The TE420P and TE415P improve upon Digium’s existing TE411P and TE406P. These premium interface cards provide carrier-grade echo cancellation, Voice Quality Enhancement (VQE), DTMF decoding, and tone recognition. Both cards minimize loads on the processor and PCI bus, and are designed to perform in the most difficult environments. The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels.

Digium designed these solutions to be fully compatible with existing software applications and is completely integrated with the Asterisk platform. Additionally, the open source driver supports an API interface for custom application development. All new solutions support industry standard telephony and data protocols, including Primary Rate ISDN (both North American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC and Frame Relay data modes.

About Digium

Digium is the original creator and primary developer of Asterisk, the industry’s first open source PBX and Asterisk Business Edition, the professional-grade version of Asterisk. Used in combination with Digium’s PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures.

Digium provides quality hardware and software products that enable telephony applications including legacy PBX, IVR, auto attendant, next generation gateways, media servers and application servers. Digium also offers a full range of professional services including consulting, technical support and customer software development services.

About Asterisk

Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed to from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, and VoIP packet protocols such as IAX, SIP and H.323. It supports US and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respected owners.

More from: Asterisk Garrett Smith

Cisco Introduces New Unified Communications System to Streamline Business Processes, Drive Productivity

Cisco Introduces New Unified Communications System to Streamline Business Processes, Drive Productivity

ORLANDO, Florida (VoiceCon Spring 2006) – March 6, 2006 – Source: Cisco Systems Press Release – Cisco Systems, Inc. today announced the Cisco® Unified Communications system, a new suite of voice, data and video products and applications specifically designed to help organizations of all sizes to communicate more effectively. It will allow customers to integrate their communications system with their IT infrastructure, streamlining business processes for the way effective businesses need to work today.

Based on the Cisco Service-Oriented Network Architecture (SONA) announced in December 2005, the Cisco Unified Communications system is an open and extensible platform for real-time communications based on presence, mobility and the intelligent information network. By using the IT data network as the service delivery platform, the system helps workers to reach the right resource the first time by delivering presence and preference information to an organization’s employees.

“The Cisco Unified Communications system is the first true second-generation Internet Protocol (IP) Communications system providing not just telephone services, but rather a rich communications environment that seamlessly integrates voice, video and data collaboration in one system. It is also the first new Cisco system to fully support Cisco SONA, announced in December 2005,” said Charles Giancarlo, chief development officer, Cisco Systems, Inc. “Cisco SONA extends the power of the network to optimize applications, processes and resources to deliver greater business benefits to enterprises. By building on Cisco SONA, Cisco Unified Communications leverages network intelligence to greatly simplify the day-to-day challenges of collaboration with colleagues.”

The Cisco Unified Communications system is based on Cisco’s industry-leading IP Communications portfolio including Cisco CallManager, Cisco Unity, Cisco MeetingPlace and Cisco IP Contact Center and now includes additional innovative products, applications, features and capabilities. New to the Cisco Unified Communications system are Cisco Unified Personal Communicator, Cisco Unified Presence Server and Customer Interaction Analyzer. Current customers will be able to upgrade their existing systems to take advantage of the new capabilities.

Cisco Unified Personal Communicator simplifies the way workers share information by helping them to communicate in real time. Its user-friendly GUI (Graphical User Interface) makes it easy to move through multiple communications applications. The Unified Personal Communicator bridges the gap between the stand-alone applications on the desktop, telephone and network. Using dynamic presence information, employees can search existing directories to locate contacts and simply “click to call” using voice and video, allowing them to exchange ideas face-to-face. The virtual nature of IP networks allows remote or traveling employees to securely access these tools from wherever they are.

The Cisco Unified Presence Server collects information about a user’s status, such as whether or not they are using a device such as a telephone, personal computer or video terminal at a particular time. Using this information, applications such as Cisco Unified Personal Communicator and Cisco Unified CallManager can help users connect with colleagues more efficiently by determining the most effective method of communication. The Cisco Unified Presence Server aggregates presence information from the network as well as Cisco Unified CallManager and third-party devices using SIP and SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) and then publishes that information to Cisco Unified IP Phones, Cisco Unified Personal Communicator and third-party services and applications such as IBM Lotus Sametime and Microsoft Live Communications Server (LCS) 2005.

The Customer Interaction Analyzer is being introduced to maximize effective communications with customers, a new approach to analytics in the contact center. It uses information from customer interactions, including self service and agent assisted interactions, to determine things like customer distress, agent distress, silence and word patterns. The data helps to give the conversations business context and can help a business to coach and train agents, make changes to processes and self service scripts based upon findings – ultimately creating better customer relationships and growth for the business.

Additional new features of the Cisco Unified Communications system include the following:

Cisco Unified CallManager 5.0 and Cisco Unified CallManager Express 3.4 and Survivable Remote Site Telephony (SRST) 3.4 now natively support SIP, effectively opening up the system to an emerging standards-based developer community while retaining the current security and resiliency features. A new program, SIP Verified, provides third-party verification for voice, data and video SIP endpoints. An initial set of vendors who have completed this testing is also announced.

Cisco Unified CallManager 5.0 is now available in a choice of operating models based on customer and channel partner preference. A new appliance model version based on Linux is available now and a version based on the existing open operating system model is scheduled to be available within 12 months.

“Miercom has exercised and reviewed key components of the entire Cisco Unified Communications system and after seeing it in action, we believe that Cisco has leapfrogged their competition in a number of areas,” said Ed Mier, principal, Miercom. “Cisco’s native implementation of SIP, which is interoperable with Skinny Client Control Protocol (SCCP) helps give customers investment protection for their system so that it can adapt as quickly as the standard does.”

Cisco continues to bridge the communications islands with innovative solutions building on the enterprise Wi-Fi (802.11) networks and the GSM public networks. In conjunction with leading wireless handset suppliers such as Nokia and RIM, Cisco will soon bring to market single and dual mode Smartphone solutions which drive enhanced productivity of mobile enterprise employees both inside and outside the office. These single device products allow users to reduce the communications complexity and help companies manage costs without losing the productivity.

“Because of their expertise in network infrastructure, Cisco was really the only vendor we considered when we decided to implement an end-to-end IP Communications solution,” said Mike DeDecker, network administrator at Warner Pacific Insurance Services. “As we move forward with our implementation and look for new ways to reduce costs and streamline processes, Cisco Unified Communications is at the top of the list. Cisco’s long-term networking background gives us the assurance we need when we’re looking to put applications in place.”

Cisco and its partners provide a lifecycle services approach to deploy and manage the Cisco Unified Communications system. New Cisco Operate Services for Unified Communications combine technical support services capabilities such as server replacement, application software updates and hardware and software problem resolution into one service that covers the entire system. To ensure proper deployment, Cisco is also offering Planning and Design Service Bundles and Optimization Services that are packaged for easy ordering.

Cisco also today introduced a number of new phones and updates to existing applications, as well as announcing Cisco Unified CallManager and Cisco Unified IP Phone are localized for China, Korea and Japan. For more information on the Cisco Unified Communications system visit: www.cisco.com/go/unified.

About Cisco Systems
Cisco Systems, Inc. (NASDAQ: CSCO) is the worldwide leader in networking for the Internet. Information about Cisco can be found at http://www.cisco.com. For ongoing news, please go to: http://newsroom.cisco.com

Linksys Shipping WRTP54G-NA Wireless G Broadband Router

The Linksys WRTP54G-NA Wireless-G Broadband Router is really four devices in one box. First, there’s the Wireless Access Point, which lets you connect both screaming fast Wireless-G (802.11g at 54Mbps) and Wireless-B (802.11b at 11Mbps) devices to the network. There’s also a built-in 4-port full-duplex 10/100 Switch to connect your wired-Ethernet devices together. Connect four PCs directly, or attach more hubs and switches to create as big a network as you need. The Router function lets your whole network share a high-speed cable or DSL Internet connection.


  • All-in-one Internet-sharing Router, 4-port Switch, and 54Mbps Wireless-G (802.11g) Access Point
  • Shares a single Internet connection and other resources with Ethernet wired and Wireless-G and B devices
  • Two standard phone jacks enable feature-rich telephone service over your cable or DSL Internet connection
  • High security: Wi-Fi Protected Access? (WPA), wireless MAC address filtering, powerful SPI firewall

The fourth function is the phone adapter which enables high-quality feature-rich telephone service through your high-speed connection even while you’re surfing the Internet. There are two standard telephone jacks, each operating independently — like having two phone lines. With Vonage Service, you’ll get low domestic and international phone rates, Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and lots of other available special phone features. Choose any free local dialing US area code, regardless of where you live. Add a virtual phone number in any area code, or even a US-wide toll-free number.

To protect your data and privacy, the Wireless-G Broadband Router can encode all wireless transmissions with up to 128-bit encryption, and supports both Wired Equivalent Privacy (WEP) and the industrial-strength wireless security of Wi-Fi Protected Access? (WPA). The Router can serve as a DHCP Server, has a powerful SPI firewall to protect your PCs against intruders and most known Internet attacks, supports VPN pass-through, and can be configured to filter internal users’ access to the Internet. Configuration is a snap with the web browser-based configuration utility.

With the Linksys Wireless-G Broadband Router at the center of your home or office network, you can share a high-speed Internet connection, files, printers, and multi-player games, and turn that Internet connection into a high-quality, high-value telephone service!

The Linksys WRTP54G-NA is available at a retail price of $154.95 at VoIPSupply.com

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