Presentation on theme: "Week Twelve Agenda Attendance Announcements December 1, lab classroom in Phillips Hall 222 has been requested, but not official. Mimic Simulator Lab Assignment."— Presentation transcript:
Week Twelve Agenda Attendance Announcements December 1, lab classroom in Phillips Hall 222 has been requested, but not official. Mimic Simulator Lab Assignment 4-1-3 Review Week Eleven Information Current Week Information Upcoming Assignments
Week Eleven Topics Review Week Ten Information 1.Analog to digital signaling 2.PBX and PSTN 3.Definitions 4.Trunk capacity Current Week Information 1.VoIP 2.Codec 3.WLAN
Analog and Digital Signaling The human voice generates sound waves The telephone converts the sound waves into an analog signal. To obtain clear voice connections, the PSTN switches convert analog speech to a digital format and send it over the digital network. At the other end of the connection, the digital signal is converted back to analog and to the normal sound waves that the ear can hear. Digital signals don’t pick up the noise levels as analog signals, and doesn’t induce any additional noise when amplifiing signals. Digital signals hold their original form better than analog signals over greater distances, regeneration, coded, and decoded translations.
Analog and Digital Signaling The human range for speech is approximately 400 to 4000 hertz (hz). Higher frequencies are filtered. Sampling is the method used on analog signals to formalize the digitizing process. A voltage level corresponds to the amplitude of the signal.
Analog and Digital Signaling Pulse Code Modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a numeric (usually binary) code. The standard code word size is 8 bits.
Analog and Digital Signaling There are several steps involved in converting an analog signal into PCM digital format, as shown in the figure
Companding Signal is compressed for more efficient transmission, and less noise Two common methods: The A-law standard is used in Europe, Mu-law is used in North America and Japan The methods are similar—but they are not compatible
Analog and Digital Signaling 1.Filter analog signal – remove frequencies > 4000 hertz 2.Sample – rate at least twice the highest frequency according to Nyquist Theorem. Samples the filtered input signal at a constant frequency using Pulse Amplitude Modulation (PAM). 3.Digitize – occurs prior to transmission over the telephone network (PCM process)
Analog and Digital Signaling 4. Quantization and coding – A process that converts each analog sample value into a discrete value to which a unique digital code word is assigned. 5. Companding – A process in which compression is followed by expansion; often used for noise reduction in equipment, in which case compression is applied before noise exposure and expansion after exposure. A process in which the dynamic range of a signal is reduced for recording purposes and then expanded to its original value for reproduction or playback.
Companding A signal is compressed for more efficient transmission, and less noise Two common methods: The A-law standard is used in Europe, Mu-law is used in North America and Japan The methods are similar—but they are not compatible
Public Switched Telephone Network (PSTN) Telephones connect to a CO (Central Office) through the local loop The local loop is an analog connection All analog signals are converted to digital at the CO Except for the local loop the entire phone system is a modern digital network
What is a Private Branch Exchange (PBX)? PBX is a private telephone network used within a company. The users of the PBX phone system share a number of outside lines for making external phone calls. A PBX connects the internal telephones within a business and also connects them to the public switched telephone network (PSTN).
PBX Features A PBX is a business telephone system that provides business features such as call hold, call transfer, call forward, follow-me, call park, conference calls, music on hold, call history, and voice mail. Most of these features are not available in traditional PSTN switches. A PBX switch often connects to the PSTN through one or more T1 digital circuits. A PBX supports end-to-end digital transmission, employs PCM switching technology, and supports both analog and digital proprietary telephones
PBX Features Analog PBXs were phased out about twenty years ago. Today, most all vendors manufacture digital PBXs.
Trunk Line Capacity In this diagram, 7 telephones connect to the CO in Neighborhood A and 6 connect to the CO in Neighborhood B How many simultaneous conversations should this trunk line carry?
Trunk Line Capacity The science of Traffic Engineering answers this question
What is Traffic Engineering? Voice traffic engineering is the science of selecting the correct number of lines and the proper types of service to accommodate users. Detailed capacity planning of all network resources should be considered to minimize degraded voice service in integrated networks. We can calculate the bandwidth required to support a number of voice calls with a given probability that the call will go through
Terminology Blocking probability Grade of Service (GoS) Erlang Centum Call Second (CCS) Busy hour Busy Hour Traffic (BHT) Call Detail Record (CDR)
Definitions The blocking probability value describes the calls that cannot be completed because insufficient lines have been provided. For example, a blocking probability value of 0.01 means that 1 percent of calls would be blocked. GoS is the probability that a voice gateway will block a call while attempting to allocate circuits during the busiest hour. GoS is written as a blocking factor, Pxx, where xx is the percentage of calls that are blocked for a traffic system. For example, traffic facilities that require P01 GoS define a 1 percent probability of callers being blocked.
Definitions One Erlang equals one full hour, or 3600 seconds, of telephone conversation The busy hour is the 60-minute period in a given 24- hour period during which the maximum total traffic load occurs. The busy hour is sometimes called the peak hour. The BHT, in Erlang’s or CCSs, is the number of hours of traffic transported across a trunk group during the busy hour (the busiest hour of operation). A CDR is a record containing information about recent system usage, such as the identities of sources (points of origin), the identities of destinations (endpoints), the duration of each call, etc
Trunk Capacity Calculation For example, one hour of conversation (one Erlang might be ten 6-minute calls or 15 4-minute calls. Receiving 100 calls, with an average length of 6 minutes, in one hour is equivalent to ten Erlangs For example, if you know from your call logger that 350 calls are made on a trunk group in the busiest hour and that the average call duration is 180 seconds, you can calculate the BHT as follows: BHT = Average call duration (seconds) * calls per hour/3600 BHT = 180 * 350/3600 BHT = 17.5 Erlangs
Capacity Information There are years of data on the number and duration of a phone conversation This historical data can be used to calculate the capacity or number of trunk lines needed in a telephone system Erlang Tables are used for this calculation
What is an Erlang Table? Erlang tables show the amount of traffic potential (the BHT) for specified numbers of circuits for given probabilities of receiving a busy signal (the GoS) The BHT calculation results are stated in Erlangs Erlang tables combine offered traffic (the BHT), number of circuits, and GoS in the following traffic models:
What is an Erlang Table? Erlang B: This is the most common traffic model, which is used to calculate how many lines are required if the traffic (in Erlangs) during the busiest hour is known. The model assumes that all blocked calls are cleared immediately. Extended Erlang B: This model is similar to ErlangB, but it takes into account the additional traffic load caused by blocked callers who immediately try to call again. The retry percentage can be specified. Erlang C: This model assumes that all blocked calls stay in the system until they can be handled. This model can be applied to the design of call center staffing arrangements in which calls that cannot be answered immediately enter a queue
What is an Erlang Table? Erlang C: This model assumes that all blocked calls stay in the system until they can be handled. This model can be applied to the design of call center staffing arrangements in which calls that cannot be answered immediately enter a queue
Trunk Capacity Calculation The network design is based on a star topology that connects each branch office directly to the main office. There are approximately 15 people per branch office. The bidirectional voice and fax call volume totals about 2.5 hours per person per day (in each branch office). Approximately 20 percent of the total call volume is between the headquarters and each branch office. The busy-hour loading factor is 17 percent. In other words, the BHT is 17% of the total traffic. One 64-kbps circuit supports one call. The acceptable GoS is P05
Trunk Capacity Calculation 2.5 hours call volume per user per day * 15 users = 37.5 hours daily call volume per office 37.5 hours * 17 percent (busy-hour load) = 6.375 hours of traffic in the busy hour 6.375 hours * 60 minutes per hour = 382.5 minutes of traffic per busy hour 382.5 minutes per busy hour * 1 Erlang/60 minutes per busy hour = 6.375 Erlangs 6.375 Erlangs* 20 percent of traffic to headquarters = 1.275 Erlangs volume proposed
Final Calculation To determine the appropriate number of trunks required to transport the traffic, the next step is to consult the Erlangtable, given the desired GoS This organization chose a P05 GoS. Using the 1.275 Erlangsand GoS= P05, as well as the ErlangB table: http://www.erlang.com/calculator/erlb/ http://www.erlang.com/calculator/erlb/ four circuits are required for communication between each branch office and the headquarters office
What do the terms FXS and FXO mean? FXS and FXO are the name of ports used by Analog phone lines (also known as POTS -Plain Old Telephone Service) or phones. FXS -Foreign eXchange Subscriber interface is the port that actually delivers the analog line to the subscriber. In other words it is the ‘plug on the wall’ that delivers a dial tone, battery current and ring voltage.
What do the terms FXS and FXO mean? FXO -Foreign eXchange Office interface is the port that receives the analog line. It is the plug on the phone or fax machine, or the plug(s) on your analog phone system. It delivers an on-hook/off-hook indication (loop closure). Since the FXO port is attached to a device, such as a fax or phone, the device is often called the ‘FXO device’. FXO and FXS are always paired, i.e similar to a male / female plug. Without a PBX, a phone is connected directly to the FXS port provided by a telephone company
Connecting a Traditional PBX to the PSTN If you have a PBX, then you connect the lines provided by the telephone company to the PBX and then the phones to the PBX. Therefore, the PBX must have both FXO ports (to connect to the FXS ports provided by the telephone company) and FXS ports (to connect the phone or fax devices to).
Telephone Signaling In a telephony system, a signaling mechanism is required for establishing and disconnecting telephone communications.
Three Types of Signaling Used To Make a Phone Call Supervision signaling: Typically characterized as on-hook, off- hook, and ringing, supervision signaling alerts the CO switch to the state of the telephone on each local loop. Supervision signaling is used, for example, to initiate a telephone call request on a line or trunk and to hold or release an established connection. Address signaling: Used to pass dialed digits (pulse or DTMF) to a PBX or PSTN switch. These dialed digits provide the switch with a connection path to another telephone or customer premises equipment. Informational signaling: Includes dial tone, busy tone, reorder tone, and tones indicating that a receiver is off-hook or that no such number exists, such as those used with call progress indicators
Analog Telephony Signaling Loop start: Loop start is the simplest and least intelligent signaling protocol, and the most common form of local-loop signaling. Only for residential use. Ground start: Also called reverse battery, ground start is a modification of loop start that provides positive recognition of connects and disconnects (off-hook and on-hook)., PBXs typically use this type of signaling. E&M: E&M is a common trunk signaling technique used between PBXs.
Digital Telephone Signaling CAS CCS DPNSS ISDN QSIG Digital Signaling –standards based protocol to allow different vendor’s PBXs to communicate SS7 Digital Signaling -used within the PSTN for signaling between PSTN switches
Why Integrate Voice and Data Networks? Integrating data, voice, and video in a network enables vendors to introduce new features The unified communications network model enables distributed call routing, control, and application functions based on industry standards Enterprises can mix and match equipment from multiple vendors and geographically deploy these systems wherever they are needed Only one network to maintain
VoIP or IP Telephony? Cisco distinguishes between the two Most technical discussions don’t VoIP –analog phones and/or analog PBXs are still used, but the analog signals are converted to IP packets with a Voice Enabled router IP Telephony –IP phones are used; the system is completely IP. Specialized call processing software replaces the PBX –this may be called an IP PBX
VoIP Connection To setup a VoIP communication we need the do the following: The ADC (Analog to Digital Converter) converts analog voice to digital signals (bits) The voice data is compressed to send the fewest number of bits while still retaining the original information (Codec) Voice packets are sent using a real-time protocol (typically RTP over UDP over IP) We need a signaling protocol to call users: ITU-T H323 or SIP At the receiver we have to disassemble packets, extract data, then convert them to analog voice signals and send them to sound card (or phone) All that must be done in a real time fashion cause we cannot waiting for too long for a vocal answer! (QOS)
VoIP Technology VoIP is an “Overlay” technology VoIP is applied on top of an IP Network If the IP network is not working properly VoIP will simply be one more thing that is broken Make sure the IP network is working correctly FIRST--then implement VoIP
H.323 Protocol H.323 is a standard for teleconferencing that was developed by the International Telecommunications Union (ITU). It supports full multimedia audio, video and data transmission between groups of two or more participants, and it is designed to support large networks. H.323 is still a very important protocol, but it has fallen out of use for consumer VoIP products due to the fact that it is difficult to make it work through firewalls that are designed to protect computers running many different applications. It is a system best suited to large organizations that possess the technical skills to overcome these problems. As a solution for a home or small office telephony system it is best avoided
Session Initiation Protocol (SIP) SIP (Session Initiation Protocol) is an Internet Engineering Task Force (IETF) standard signaling protocol for teleconferencing, telephony, presence and event notification and instant messaging. It provides a mechanism for setting up and managing connections, but not for transporting the audio or video data. It is probably now the most widely used protocol for managing Internet telephony
SIP Protocols SIP-Session Initiation Protocol MegacoH.248 -Gateway Control Protocol MGCP-Media Gateway Control Protocol MIMERVP over IP -Remote Voice Protocol Over IP Specification SAPv2-Session Announcement Protocol SDP-Session Description Protocol SGCP-Simple Gateway Control Protocol Skinny-Skinny Client Control Protocol (SCCP
SIP Protocols Sip is the major VoIP protocol in use today Very similar to http Sip uses port 5060 Sip has the same Status Codes as http Instead of a get as in http, Sip issues an INVITE when someone makes a call.The following are SIP responses: 1xx Informational (e.g. 100 Trying, 180 Ringing) 2xx Successful (e.g. 200 OK, 202 Accepted) 3xx Redirection (e.g. 302 Moved Temporarily) 4xx Request Failure (e.g. 404 Not Found, 482 Loop Detected) 5xx Server Failure (e.g. 501 Not Implemented) 6xx Global Failure (e.g. 603 Decline
SIP VoIP System User agents or phones register with a SIP Proxy. To initiate a session, the caller (or User Agent Client) sends a request with the SIP URL of the called party. If the client knows the location of the other party it can send the request directly to their IP address; if not, the client can send it to a locally configured SIP network server. The server will resolve the called user's location and send the request to them. During the course of locating a user, one SIP network server can proxy or redirect the call to additional servers until it arrives at one that definitely knows the IP address where the called user can be found. Once found, the request is sent to the user.
SIP VoIP System If phone A know the location of phone B, it can call phone B directly without going through the proxy server Sip uses email-style addresses to identify users
RTP RTP is the Real-time Transport Protocol RTP is used by H.323 and SIP for the actual transmission of the VoIP packets RTP uses UDP Additionally, RTCP (Real-time Control Protocol) provides this information: Packet Loss Jitter Delay Signal Level Call Quality Metrics Echo Return Loss
OSI Model ISO Model LayerProtocol or Standard PresentationApplications/CODECS SessionH.323 and SIP TransportRTP / UDP / TCP NetworkIP – Non QoS Data LinkATM, FR, PPP, Ethernet
Cisco’s Solution IP Telephony The main component of Cisco’s solution is the Cisco Unified Communications Manager: It is a server used for call control and signaling, It replaces a PBX The IP phone itself performs voice-to-IP conversion, and voice-enabled routers are not required within the enterprise network If connection to the PSTN is required, a voice- enabled router or other gateway must be added where calls are forwarded to the PSTN
Multisite WAN with Centralized Processing Design
Definition of CODEC A codec is a device or computer program capable of encoding and/or decoding a digital data stream or signal. The word codec is a portmanteau of 'compressor-decompressor' or, more commonly, 'coder-decoder‘.
Voice Coding and Compression CODEC A DSP (Digital Signal Processor is a hardware component that converts the analog signal to digital format Codecs are software drivers that are used to encode the speech in a compact enough form that they can be sent in real time across a network using the bandwidth available Codecs are implemented within a DSP VoIP software or hardware may give you the option to specify the codecs you prefer to use This allows you to make a choice between voice quality and network bandwidth usage, which might be necessary if you want to allow multiple simultaneous calls to be held using an ordinary broadband connection
Coding and Compression Algorithm The different codecs provide a certain quality of speech Advances in technology have greatly improved the quality of compressed voice and have resulted in a variety of coding and compression algorithms PCM: The toll quality voice expected from the PSTN. PCM runs at 64 kbps and provides no compression, and therefore no opportunity for bandwidth savings The other algorithms use compression to save bandwidth Voice quality is affected
Which CODEC is most affective? G.729 is the recommended voice codec for most WAN networks (that do not do multiple encodings) because of its relatively low bandwidth requirements and high mean opinion score (MOS) (ITU-T P.800)
Reducing the Amount of Voice Traffic The codecs chosen are a trade-off between bandwidth and voice quality Two techniques used to reduce voice traffic: cRTP
Every IP packet consists of a header and the payload (data, voice) Although the payload of a voice packet is small (20 bytes when G.729 is used), the header is 40 bytes cRTP compresses the header to 2 or 4 bytes Use on slow WAN links, but it is CPU intensive
VAD Voice Activity Detection On average, about 35 percent of calls are silence In traditional voice networks, all voice calls use a fixed bandwidth of 64 kbps regardless of how much of the conversation is speech and how much is silence When VoIP is used, this silence is packetized along with the conversation. VAD suppresses packets of silence, so instead of sending IP packets of silence, only IP packets of conversation are sent Therefore, gateways can interleave data traffic with actual voice conversation traffic, resulting in more effective use of the network bandwidth
QoS for Voice Classify Packets Mark Packets Marked packets can be prioritized in the scheme of queuing LLQ –Cisco’s Low Latency Queuing is the recommended method for VoIP networks
CAC –Call Admission Control CAC protects voice traffic from being negatively affected by other voice traffic by keeping excess voice traffic off the network. If a WAN link is fully utilized with voice traffic then adding more voice calls will degrade all the calls CAC checks if the link is maximized and won’t allow new calls to go through until bandwidth is available Callers will get a busy signal or “all circuits busy message”
LFI Link fragmentation and interleaving ensures that small voice packets don’t get stuck behind a large data packet The data packets are fragmented into smaller packets The voice packets can slip in between them because the are initially small.
Wireless Technologies MMDS = Multichannel multipoint distribution services used for general purpose broadband networking. United States LMDS = Local multipoint distribution service used for wireless cable television (TV), referring to premium wireless subscription TV rather than traditional free broadcast TV or cable TV.
Wireless Technologies GSM = Global system for mobile communication is a cellular phone protocol. Used in many part of the world. GPRS = General packet radio service is a radio technology for GSM networks. Europe and Asia. Not related to GPS CDMA = Code division multiple access is a cellular phone protocol used for digital communication. United States
Cisco’s Acquisitions Cisco acquired the company Aironet-Aironet manufactured enterprise-level wireless solutions Cisco acquired Linksys –home/small office wireless solutions Cisco acquired Airespace–wireless LAN controllers
What is RF? Radio frequency is a term that refers to alternating current (AC) having characteristics such that, if the current is input to an antenna, an electromagnetic (EM) field is generated suitable for wireless broadcasting and/or communications. Frequencies of electromagnetic radiation in the range between 10 kHz and 300 MHz. Many types of wireless devices make use of RF fields. Cordless and cellular phone, radio and television broadcast stations, satellite communications systems, and two-way radio services all operate in the RF spectrum.
Phenomena Affecting RF Reflection: Occurs when the RF signal bounces off objects such as metal or glass surfaces. Refraction: Occurs when the RF signal passes through objects such as glass surfaces and changes direction. Absorption: Occurs when an object, such as a wall or furniture, absorbs the RF signal. Scattering: Occurs when an RF wave strikes an uneven surface and reflects in many directions. Scattering also occurs when an RF wave travels through a medium that consists of objects that are much smaller than the signal’s wavelength, such as heavy dust. Diffraction: Occurs when an RF wave strikes sharp edges, such as external corners of buildings, which bend the signal. Multipath: Occurs when an RF signal has more than one path between the sender and receiver. The multiple signals at the receiver might result in a distorted, low-quality signal.
Power Consumption by WLANs WLANs transmit signals just as radio stations do to reach their listeners The transmit power levels for WLANs are in milliwatts (mW), whereas for radio stations the power levels are in megawatts (MW) The amount of power that can be used in WLANs is set by the FCC Wireless LANs operate in the unlicensed frequency bands, which is why they operate at very low power levels
Wireless LANs 802.11 wireless LANs extend the 802.3 Ethernet LAN infrastructures to provide additional connectivity options. In an 802.3 Ethernet LAN, each client has a cable that connects the client NIC to a switch. The switch is the point where the client gains access to the network. In a wireless LAN, each client uses a wireless adapter to gain access to the network through a wireless device such as a wireless router or access point. The wireless adapter in the client communicates with the wireless router or access point using RF signals. Once connected to the network, wireless clients can access network resources just as if they were wired to the network.
Wireless LAN Standard 802.11 wireless LAN is an IEEE standard that defines how radio frequency (RF) in the unlicensed industrial, scientific, and medical (ISM) frequency bands is used for the physical layer and the MAC sub-layer of wireless links. Data Rate: 802.11: 1 -2 Mb/s data rates 802.11a and g: support up to 54 Mb/s, 802.11b: supports up to a maximum of 11 Mb/s 802.11n: Up to 500 Mb/s.
Wireless LAN Standard Modulation technique: Direct Sequence Spread Spectrum (DSSS) 802.11b, 802.11g Orthogonal Frequency Division Multiplexing (OFDM). 802.11a, 802.11g, 802.11n Band: 2.4 GHz: 802.11b, 802.11g, 802.11n 5 GHz: 802.11a, 802.11n
802.11a OFDM modulation and uses the 5 GHz band. Less likely to experience interference than devices that operate in the 2.4 GHz band Because there are fewer consumer devices that use the 5 GHz band. There are some important disadvantages to using the 5 GHz band. The first is that higher frequency radio waves are more easily absorbed by obstacles such as walls, making 802.11a susceptible to poor performance due to obstructions. The second is that this higher frequency band has slightly poorer range than either 802.11b or g. Also, some countries, including Russia, do not permit the use of the 5 GHz band, which may continue to curtail its deployment.
802.11n The IEEE 802.11n standard is intended to improve WLAN data rates and range without requiring additional power or RF band allocation. 802.11n uses multiple radios and antennas at endpoints, each broadcasting on the same frequency to establish multiple streams. The multiple input/multiple output (MIMO) technology splits a high data-rate stream into multiple lower rate streams and broadcasts them simultaneously over the available radios and antennae. This allows for a theoretical maximum data rate of 248 Mb/s using two streams. The standard is now ratified
802.11n Operates in the 2.4 GHz band or in the 5 GHz band The 2.4GHz band is more crowded with interference from lots of other devices and 802.11g networks The 5GHz band is less crowded but the range is less Terminology: A “dual-frequency” or “dual-band” AP allows you to select which band 2.4GHz or 5 GHz A “dual-radio” AP allows the AP to operate at both frequencies You can allows your old 802.11g clients to connect on the 2.4 GHz band and your new 802.11n clients to connect on the 5GHz band
Wi-Fi Certification The 3 key organizations influencing WLAN standards are: ITU-R ITU-R regulates allocation of RF bands. The ITU-R regulates the allocation of the RF spectrum. IEEE IEEE specifies how RF is modulated to carry information. The IEEE developed and maintains the standards for local and metropolitan area networks. The dominant standards in the IEEE 802 are 802.3 Ethernet, and 802.11 Wireless LAN
Wi-Fi Certification Wi-Fi Alliance (www.wi-fi.org)www.wi-fi.org Wi-Fi ensures that vendors make devices that are interoperable. The Wi-Fi Alliance is to improve the interoperability of products by certifying vendors for conformance to industry norms and adherence to standards. Certification includes all three IEEE 802.11 RF technologies, as well as early adoption of pending IEEE drafts, such as 802.11n, and the WPA and WPA2security standards based on IEEE 802.11i.
802.11g and n (2.4GHz) Although there are 11 channels, these channels overlap each other You can have only use three APs in close proximity without interference. The APS will operate on channels 1, 6 and 11
802.11a and n ( 5GHz) Twelve separate non-overlapping channels As a result, you can have up to twelve access points set to different channels in the same area without them interfering with each other. This makes access point channel assignment much easier and significantly increases the throughput the wireless LAN can deliver within a given area. In addition, RF interference is much less likely because of the less- crowded 5 GHz band.
Wireless NICs The device that makes a client station capable of sending and receiving RF signals is the wireless NIC. Like an Ethernet NIC, the wireless NIC, using the modulation technique it is configured to use, encodes a data stream onto an RF signal. Wireless NICs are most often associated with mobile devices, such as laptop computers. In the 1990s, wireless NICs for laptops were cards that slipped into the PCMCIA slot. PCMCIA wireless NICs are still common, but many manufacturers have begun building the wireless NIC right into the laptop. Unlike 802.3 Ethernet interfaces built into PCs, the wireless NIC is not visible, because there is no requirement to connect a cable to it.
Wireless NICs Other options have emerged over the years as well. Desktops located in an existing, non-wired facility can have a wireless PCI NIC installed. To quickly set up a PC, mobile or desktop, with a wireless NIC, there are many USB options available as well.
Wireless Access Point (AP) An access point connects wireless clients (or stations) to the wired LAN. An access point is a Layer 2 device that functions like an 802.3 Ethernet hub. Client devices do not typically communicate directly with each other; they communicate with the AP. In essence, an access point converts the TCP/IP data packets from their 802.11 frame encapsulation format in the air to the 802.3 Ethernet frame format on the wired Ethernet network.
Autonomous AP Originally in WLANs, all of the configurations and management was done on each access point This type of access point was a stand-alone device The term for this is a fat AP, standalone AP, intelligent AP, or, most commonly, an autonomous AP All encryption and decryption mechanisms and MAC layer mechanisms also operate within the autonomous AP
Upcoming Deadlines Assignment 1-4-3 Data Center Design Project Phase 3: Data Center Network Design is due November 24,2010. Assignment 11-1, Concept Question 8, due November 24, 2010 Assignment 12-1, Concept Questions 9, due December 1, 2010. Assignement 13-1 Concept Questions 10 is due December 8, 2010. Assignment 1-4-4 Final Design Document is due December 15, 2010.