Understanding Wireless Voice-over-IP (VoIP)
Voice over Wireless LAN (VoWLAN)
Connect802 can help you assure wireless Voice-over-IP VoIP support with your 802.11 Wi-Fi wireless LAN. We understand the signal strength, signal-to-noise and carrier-to-interference issues that must be considered to assure wireless VoIP success.
If you have questions, please don't hesitate to call us today! We'll be happy to provide you with any technical explanation that you need to help you assure a successful wireless networking deployment.
A VoIP-enabled telephone uses standard TCP/IP protocols to send digitized voice calls across a conventional, IP-routed infrastructure (like the Internet). A special signaling protocol is used to set up the end-to-end call. VoWLAN is the implementation of VoIP over a wireless network.
What is VoIP Telephony ?
VoIP (also called "IP Telephony") is transmission of a telephone call using digitized voice data carried by Internet Protocol (IP). This can be accomplished using the services of either a wired Ethernet or a Wi-Fi wireless LAN. Alternatively, one could make a phone call using the Public Switched Telephone Network (PSTN) with a conventional telephone.
In the PSTN, intelligence and control belong to the central office and the network itself. With VoIP, intelligence is in the phone set or other edge device. This allows the VoIP user access to much more sophisticated and productive features, beyond those that can be offered in the centralized PSTN environment. Some of these capabilities include the following:
Location Awareness: Since the VoIP user goes through a login and registration process when they attach their phone to the network they can move to different locations and still be reached through the same telephone number.
Presence Management: In the PSTN a called party is either "on hook" or "off hook". With VoIP the availability of a called party takes on a variety of additional possible characteristics. For example, if someone is in a meeting they may be able to receive text messages, but not talk on the phone. The VoIP features that relate to various called party status responses are implemented is called Presence Management.
Call Forking: A user can register at more than one location so an incoming call will ring, for example, at their desk, at home, and on their cell phone.
Unified Messaging: Access to all messaging formats from a single mobile terminal including: voice mail, fax, pager, and text messaging. The user has the ability to manage their data and information services under a single umbrella.
User Profile Management: Each user has a unique profile file that identifies their membership in groups, their availability times, the types of media they can receive, and (in a billing environment) how their account should be billed.
Why Move From PSTN Telephony to VoIP? - The VoIP Value Proposition
Voice-over-IP telephony as an alternative to standard PSTN PBX systems provides a number of advantages and is evolving to be the implementation-of-choice for all new telephone systems. Some areas of benefit to consider include the following:
Application Advantage: The natural integration between VoIP telephony and Ethernet/Wi-Fi networking allows for the implementation of software applications that increase user productivity and improve the customer experience. The most basic VoIP applications integrate database functions with telephone functions to present customer records on-screen at the time an incoming call is received. Easy voice conferencing with accompanying multimedia (slide shows, video, shared virtual whiteboarding) improve in-house productivity. Telecommuting and the virtual team enjoys a seamless mechanism for collaboration and communication. A cohesive reporting and management structure lets the system manager have the information they need for capacity planning and troubleshooting.
Return-on-Investment (ROI) Advantage: Because the infrastructure for VoIP telephony is the same as that used for computer data networking there is the need to install and manage only one communications system (the "network") instead of wiring for phones and also creating a computer data system.
The "Converged Network" Advantage: The historical separation of voice networks (the phone company), video networks (cable and satellite systems), and data (the Internet) is gradually disappearing. This "convergence" of voice, video, and data systems can be seen in 3G cellular where VoIP combines with web browsing on PDA's and phones, and camera phones provide picture and movie capabilities. Today's investment in VoIP technology becomes a foundation for tomorrow's converged network services.
Understanding Voice-over-IP: "How Does It Work?"
To the user of a VoIP telephone the experience of placing a call can be identical to the PSTN alternative: You pick up the phone, hear a dial tone, dial the number, the called party's phone rings, and you talk. Of course, the experience may be enhanced by picking a desired called party from an on-screen list, a sales lead database, or other integrated function.
Behind the scenes a number of things must happen to make a call possible. This is true for both PSTN telephony as well as VoIP systems.
How does the network know where each called party is located?
Each station registers its location with a server when it attaches to the network
Stations are hard-wired with circuits going back to the telephone company central office
How does a calling party discover the location of a called party?
The called party's location is resolved (using DNS) into an Internet domain name
Central office switch equipment connects to the numeric identity of the called party's phone number
What communication standards are used for the registration and location services?
Signaling messages may be sent in accordance with the IETF Session Initiation Protocol (SIP) standards or the ITU-T H.323 standards
Signaling and inter-switch communication is performed in accordance with the ITU-T Signaling System Number 7 (SS7) communication standards used by telephone companies worldwide
VoIP Signaling Messages
In both VoIP and PSTN telephony there are signaling messages used to communicate information concerning the connection between a calling party, the switched network system (either packet-switched IP or telephone switches between telephone company central offices), and the called party. These signaling messages include requests and replies that perform the tasks necessary to establish, manage, and terminate a call:
"ESTABLISH CALL FORWARDING" or "END CALL FORWARDING"
"CONNECT TO THE LAST CALLING NUMBER"
"SETUP A CALL" to which the response should be "CALL SETUP IS PROCEEDING"
"CALL ACCEPTED" (now the parties can talk) or "CALLED PARTY IS NOT AVAILABLE" (perhaps they're on vacation) or "CALLED PARTY BUSY" (they're using their phone - busy signal) or "CALL REJECTED" (the calling party's number has been blocked)
"INITIATE 3-WAY CALLING", "INDICATE THAT A CALL IS WAITING", "TRANSMIT CALLER ID INFORMATION"
The PSTN uses a set of rules (protocols) referred to as Signaling System Number 7 or "SS7" which defines, among other things, the signaling messages. VoIP initially used a set of rules called H.323 and now also uses an alternative set of signaling rules called Session Initiation Protocol or "SIP". Special servers can translate between the protocols used in SS7, H.323, and SIP allowing calls to be placed across systems utilizing different signaling methods.
Identifying the Calling and Called Parties
Connect802, in San Ramon, California, is in Area Code 925. Our local central office exchange is 552, and that makes our phone number 925-552-0802. This structure for a PSTN end-station identifier is defined in an ITU-T standard called E.164. E.164 telephone numbers are physically mapped to individual telephones because the circuits are wired from the telephone company central office, to the customer’s premises, and to the telephones themselves. When you call 1-925-552-0802 you reach Connect802’s main switchboard.
With VoIP each calling set must register its identity with the network. The identity can be an E.164 numeric value, an email address, or a URL. H.323 records this information in a server called a Registration, Admission, and Status (RAS) server. As a side note, don’t confuse the acronym RAS as used here in the VoIP environment with the same acronym used in the Microsoft Windows’ networking world where it stands for “Remote Access Server” (the two are completely unrelated). SIP calls the server a SIP Proxy or sometimes the term Registration Server is used. The terminology differs from H.323 to SIP, but the functionality is the same.
H.323 represents information using a binary code called ASN.1 (Abstract Syntax Notation.1) whereas SIP uses plain ASCII text. The difference is transparent to the user of the VoIP system but, as a decision maker choosing a VoIP product you should know that H.323 is a more closed system, requiring special software development for the addition of new features. SIP, on the other hand, was designed to work easily with web-enabled devices and is the signaling protocol used in 3G cellular phone systems. All other things being equal, we recommend going with SIP as opposed to H.323.
The Telco network (PSTN) is designed to manage analog voice circuits
The Internet is designed to manage packet-switched connections
VoIP Products and Services are available through Connect802 Technology Partners
utilizing the Connect EZ Predictive RF CAD Design for Accurate Wireless VoIP Design
Definitions and Descriptions of VoIP that you might find on the Web:
VoIP technology allows people to make phones calls that travel over the Internet rather than solely across wires owned by long-distance phone companies. Such calls can be made from telephone systems that tap into the Internet, and from PCs.
U.S. government reports estimate that the United States, Canada, Guam, Bermuda and Trinidad will run out of 10-digit numbers by the year 2025, driven by demand for cell phones, faxes and other devices. The coming crunch has led at least one industry organization to draw up a plan for a 12-digit future that could add some 640 billion new numbers to the pool.
VoIP's efficiencies come through its use of packet-switching technology, which breaks up communications into small bits that are dispersed to find the fastest path across the network and recombined at the end point. Traditional telephony, by contrast, is "circuit-switched," creating a dedicated channel for the duration of the call.
VoIP technology offers an alternative to POTS. In order to understand the changes that VoIP technology makes to POTS, one should have a good grasp of POTS. POTS have traditionally been highly regulated by both state public utility commissions and the FCC. With the passage of the Telecommunications Act of 1996, the FCC relaxed many of its POTS regulations which were based on specifically defined geographic calling areas. By using a mobile technology such as VoIP, users do not have a defined calling area, nor are they required to identify the source of the call. These differences have sparked the current regulatory debate. While POTS and VoIP exist as separate technologies, in the future VoIP may dominate the telephone industry.
Academic technologists are concerning themselves with wireless delivery of VoIP, and a new Institute for Electrical and Electronics Engineers (IEEE) standard designed for longer wireless links, colloquially known as 802.16 or WiMAX. The first, wireless VoIP, works much like traditional VoIP, only transmits voice packets over the Internet Protocol on a wireless network. This technology generally requires more bandwidth than a wireless network is capable of providing, but still seems to be drawing considerable interest.
Because current 802.11a and 802.11g wireless standards can only send signals up to 300 feet, some experts hail the WiMAX (802.16) standard as the "next big thing" in wireless telecommunications, and note that the technology's ability to send wireless signals at 70 Mbps for up to 30 or 40 miles could henceforth revolutionize wireless network design.
Most users implementing VOIP these days are primarily concerned about voice quality, latency and interoperability. All are fundamental quality-of-service considerations that companies need to deal with before they can even begin justifying the move to VOIP. But some security organizations are cautioning users about the dangers of unsecured VOIP services. For instance, in an August 2001 paper on its Web site, the Bethesda, Md.-based SANS Institute warned of privacy- and authentication-related issues stemming from VOIP services and urged users to apply the same precautions they've used to protect their data services.
Voice over Internet Protocol (VoIP) is a rapidly emerging technology for voice communication that uses the ubiquity of IP-based networks to deploy VoIP client devices—such as desktop IP phones, mobile VoIP-enabled handheld devices, and VoIP gateways—in an increasing number of businesses and homes around the world. (Microsoft Windows Embedded Developer Center)