E1 Frame Structuret- A Frame is composed from 256 bits that are divided to 32 Time Slots (TS) x 8 bits per

TS - Each channel rate is 64 Kb/s - The channels are in consecutive time slots numbered 0-31 - Frame rate is 8 KHz- TS 1-15, 17-31 are used for user data and are referred as channels 1-30 - TS0 is used for synchronization, alarms and messages (future use) - TS16 is used for signaling (but can be also used for data)

ป้�ายกำ�ากำ�บ: E1 Frame Structuret

A 2.048 Mbps point-to-point dedicated, digital circuit provided by the telephone companies in Europe. E1 is the European counterpart of the North American T1 line, which transmits at 1.544 Mbps, and E1 and T1 lines can be interconnected for international use. E2 through E5 lines provide multiple E1 channels.

An E1 line uses two wire pairs (one for transmit, one for receive) and time division multiplexing (TDM) to interleave 32 64-Kbps voice or data channels. See DS and T1.

While study on the radio system during my working hour, found that the system here are using normal Ethernet cable to connect the main system  to the Base Station. Normally an Ethernet cable maximum transmission distance are not recommended to be more the  100 Meter.

According to the senior engineer over there, if the base station located far away from the main system, they normally using fiber optic or some other medium such as wireless, antenna in special frequencies or E1 line from ISP.

An E1 carrier is a telecommunications facility designed to carry digital information at a bit rate of 2.048 Mbps. In conventional telecommunications, the most common use for an E1 carrier is to connect central offices within an individual telephone company. Telephone companies also lease E1 carriers to their customers for their own private purposes. Most systems use E1 circuits to transmit digitized voice, management, and control traffic between zones. The Frame Relay and Cell Relay protocols provide the means for exchanging information over the E1 communication facilities that connect remote zones.

Various types of transmission media can be used in implementing a private E1 facility, such as various types of privately installed cabling or point-to-point microwave circuits.

An E1 circuit is divided into 32 time slots, each of which implements a separate communication channel that can support a bit rate of 64,000 bps. Each of these individual channels is referred to as a Digital Signal Level zero (DS0) channel.

The term framing refers to the order in which user bits and other information is transmitted over a physical transmission medium. An E1 frame comprises a total of 256 bits. Each of the 32 inputs is assigned a fixed time slot; the E1 uses a time-division multiplexing technique to divide the capacity of the carrier into 32 channels. The framing bit is used to create a pattern to help synchronize the equipment. Picture above illustrates the format of the E1 transmission frame

E1

An E1 link operates over two separate sets of wires, usually twisted pair cable. A nominal 3 Volt peak signal is encoded with pulses using a method that avoids long periods without polarity changes. The line data rate is 2.048 Mbit/s (full duplex, i.e. 2.048 Mbit/s downstream and 2.048 Mbit/s upstream) which is split into 32 timeslots, each being allocated 8 bits in turn. Thus each timeslot sends and receives an 8-bit sample 8000 times per second (8 x 8000 x 32 = 2,048,000). This is ideal for voice telephone calls where the voice is sampled into an 8 bit number at that data rate and reconstructed at the other end. The timeslots are numbered from 0 to 31.

One timeslot (TS0) is reserved for framing purposes, and alternately transmits a fixed pattern. This allows the receiver to lock onto the start of each frame and match up each channel in turn. The standards allow for a full Cyclic Redundancy Check to be performed across all bits transmitted in each frame, to detect if the circuit is losing bits (information), but this is not always used.

One timeslot (TS16) is often reserved for signalling purposes, to control call setup and teardown according to one of several standard telecommunications protocols. This includes Channel Associated Signaling (CAS) where a set of bits is used to replicate opening and closing the circuit (as if picking up the telephone receiver and pulsing digits on a rotary phone), or using tone signalling which is passed through on the voice circuits themselves. More recent systems used Common Channel Signaling (CCS) such as ISDN or Signalling System 7 (SS7) which send short encoded messages with more information about the call including caller ID, type of

transmission required etc. ISDN is often used between the local telephone exchange and business premises, whilst SS7 is almost exclusively used between exchanges and operators. SS7 can handle up to 4096 circuits per signalling channel[citation needed], thus allowing slightly more efficient use of the overall transmission bandwidth (for example: uses 31 voice channels on an E1).

Unlike the earlier T-carrier systems developed in North America, all 8 bits of each sample are available for each call. This allows the E1 systems to be used equally well for circuit switch data calls, without risking the loss of any information.

While the original CEPT standard G.703 specifies several options for the physical transmission, almost exclusively HDB3 format is used.

[edit] Hierarchy levels

The PDH based on the E0 signal rate is designed so that each higher level can multiplex a set of lower level signals. Framed E1 is designed to carry 30 E0 data channels + 1 signalling channel, all other levels are designed to carry 4 signals from the level below. Because of the necessity for overhead bits, and justification bits to account for rate differences between sections of the network, each subsequent level has a capacity greater than would be expected from simply multiplying the lower level signal rate (so for example E2 is 8.448 Mbit/s and not 8.192 Mbit/s as one might expect when multiplying the E1 rate by 4).

Note, because bit interleaving is used, it is very difficult to demultiplex low level tributaries directly, requiring equipment to individually demultiplex every single level down to the one that is required.

Signal Rate

E0 64 kbit/s

E1 2.048 Mbit/s

E2 8.448 Mbit/s

E3 34.368 Mbit/s

E4 139.264 Mbit/s

Digital Signal 0 (DS0) is a basic digital signalling rate of 64 kbit/s, corresponding to the capacity of one voice-frequency-equivalent channel.[1] The DS0

rate, and its equivalents E0 and J0, form the basis for the digital multiplex transmission hierarchy in telecommunications systems used in North America, Europe, Japan, and the rest of the world, for both the early plesiochronous systems such as T-carrier and for modern synchronous systems such as SDH/SONET.

The DS0 rate was introduced to carry a single digitized voice call. For a typical phone call, the audio sound is digitized at an 8 kHz sample rate using 8-bit pulse-code modulation for each of the 8000 samples per second. This resulted in a data rate of 64 kbit/s.

Because of its fundamental role in carrying a single phone call, the DS0 rate forms the basis for the digital multiplex transmission hierarchy in telecommunications systems used in North America. To limit the number of wires required between two involved in exchanging voice calls, a system was built in which multiple DS0s are multiplexed together on higher capacity circuits. In this system, twenty-four (24) DS0s are multiplexed into a DS1 signal. Twenty-eight (28) DS1s are multiplexed into a DS3. When carried over copper wire, this is the well-known T-carrier system, with T1 and T3 corresponding to DS1 and DS3, respectively.

Besides its use for voice communications, the DS0 rate may support twenty 2.4 kbit/s channels, ten 4.8 kbit/s channels, five 9.67 kbit/s channels, one 56 kbit/s channel, or one 64 kbit/s clear channel.

E0 (standardized as ITU G.703) is the European equivalent of the North American DS0 for carrying a single voice call. However, there are some subtle differences in implementation. Voice signals are encoded for carriage over E0 according to ITU G.711. Note that when a T-carrier system is used as in North America, robbed bit signaling can mean that a DS0 channel carried over that system is not an error-free bit-stream. The out-of-band signaling used in the European E-carrier system avoids this

Digital signal 1 (DS1, also known as T1, sometimes "DS-1") is a T-carrier signaling scheme devised by Bell Labs.[1] DS1 is a widely used standard in

telecommunications in North America and Japan to transmit voice and data between devices. E1 is used in place of T1 outside of North America, Japan, and

South Korea. Technically, DS1 is the logical bit pattern used over a physical T1 line; however, the terms "DS1" and "T1" are often used interchangeably.

Bandwidth

A DS1 circuit is made up of twenty-four 8-bit channels (also known as timeslots or DS0s), each channel being a 64 kbit/s DS0 multiplexed carrier circuit[2]. A DS1 is also a full-duplex circuit, which means the circuit transmits and receives 1.544 Mbit/s concurrently. A total of 1.536 Mbit/s of [2] bandwidth is achieved by sampling each of the twenty-four 8-bit DS0s 8000 times per second. This sampling is referred to as 8-kHz sampling (See Pulse-code modulation). An additional 8 kbit/s of overhead is obtained from the placement of one framing bit, for a total of 1.544 Mbit/s, calculated as follows:

[edit] DS1 frame synchronization

Frame synchronization is necessary to identify the timeslots within each 24-channel frame. Synchronization takes place by allocating a framing, or 193rd, bit. This results in 8 kbit/s of framing data, for each DS1. Because this 8-kbit/s channel is used by the transmitting equipment as overhead, only 1.536 Mbit/s is actually passed on to the user. Two types of framing schemes are Super Frame (SF) and Extended Super Frame (ESF). A Super Frame consists of twelve consecutive 193-bit frames, whereas an Extended Super Frame consists of twenty-four consecutive 193-bit frames of data. Due to the unique bit sequences exchanged, the framing schemes are not compatible with each other. These two types of framing (SF and ESF) use their 8 kbit/s framing channel in different ways.

[edit] SF framing

In SF Framing, aka Super Frame, the framing channel is divided into two channels of 4 kbit/s each. One channel is for terminal frame alignment; the second is used to align the signaling frames. The terminal frame and signaling frame bits are interleaved, rather than consecutive (they are switched in Figure 2). (correction per ANSI T1.403 Section 7.2 "A frame is a set of 192 digit time-slots for the information payload preceded by one digit time-slot containing the framing (F) bit, for a total of 193 digit time-slots." Meaning the first bit of the frame is a framing bit and not the last bit.)

The terminal frame alignment channel is carried in odd-numbered frames inside the super frame and occurs with the DS0 channel synchronization. Since the framing bits occur only once per frame, in the 193rd position, the bit placement of each DS0 can be calculated. After the framing bit is sensed, the first DS0 timeslot is taken as the next 1-8 bits. Timeslot 2 is bits 9-16, timeslot 3 is 17-24, through to timeslot 24. See Figure 1. The Terminal frame alignment pattern is carried in odd-numbered frames, inside the super frame, and consists of alternating 1s and 0s: 1–0–1–0–1–0.

Figure 1. SF Framing bit. A single bit is added to the end of the DS1 frame to signal the start of a new frame.

Signaling frame alignment channel is carried in even-numbered frames inside the super frame and is used for signaling frame alignment. The signaling frame alignment

pattern consists of a 0–0–1–1–1–0. Signaling frames are identified by the framing signal's transition from 1 to 0 and from 0 to 1; thereby frames six and twelve carry signaling information. See Figure 2.

The SF format uses bit robbing to pass signaling information. Bit robbing modifies the least significant bit in each user data timeslot twice per Super Frame. (See also A&B). The two modified frames are the sixth (A) and the twelfth (B). Using two bits, four possible signaling states can be passed in each direction (0–0, 0–1, 1–0, 1–1). In order for A/B signaling to work, the exact placement of the bits must be known by both sides. Information on the frame sequence is necessary to "pick out" the A and B bits. Channel information must also be known in order to pick out the last bit of each channel. If the proper alignment (timing) did not occur, the wrong bit could be modified or read as the robbed bit. This method of signaling is also commonly referred to as Channel Associated Signaling or CAS. See Figure 2.

Figure 2. SF Framing Format (Note: Signal and Terminal frame alignment bits are interleaved)

The SF format is also known as D4 framing and D3/D4 framing format.

[edit] ESF framing

In ESF, aka Extended Super Frame, twenty-four frames make up the (extended) super frame. ESF divides the 8 kbit/s framing channel into three segments. The frame pattern uses 2 kbit/s, and a Cyclic redundancy check (CRC) uses 2 kbit/s. The remaining 4 kbit/s make up an administrative data link (DL) channel. The framing

pattern occupies the 4th, 8th, 12th, 16th, 20th and 24th frames. The pattern consists of a 0–0–1–0–1–1 sequence. This is the only pattern that is repeated in the ESF format. See Figure 3.

Figure 3. ESF Framing Format.

The CRC algorithm checks a known segment of data and adds the computed value to it. The combined data and CRC blocks are both transmitted. The receive circuitry will run the same CRC algorithm against the data portion and compare the calculation to the transmitter's CRC value. In this manner, corrupted data can be flagged as "CRC errors". The CRC checksum is passed in the 2nd, 6th, 10th, 14th, 18th, and 22nd frames. (See also Error-correcting code).

The administrative channel provides a means to communicate within the DS1 stream (sub-channel). Statistics on CRC errors can be requested and sent from one end to

another. The data channel occupies the twelve odd numbered frames. Signaling and other information passes over this channel. Provisions in the ESF standard would allow the normal A/B bit robbed signal to be enhanced. The A/B bits can be extended to four bits (ABCD). This provides 16 distinct states. An improvement from A/B, which provides 4. To overcome incompatibility with A/B signaling, equipment repeats the A&B bits (e.g. C = A and D = B). These additional signaling bits will offer new features as equipment is built to support it.

CRC errors can be detected and counted in at least one of four different registers. The registers are for transmit (in and out) and receive (in and out). Using recovered CRC data, it is possible to segment and isolate the direction of problems

Alarms

Alarms are normally produced by the receiving terminal equipment when the framing is compromised. There are three defined alarm indication signal states, identified by a legacy color scheme: red, yellow and blue.

Red alarm indicates the alarming equipment is unable to recover the framing reliably. Corruption or loss of the signal will produce “red alarm.” Connectivity has been lost toward the alarming equipment. There is no knowledge of connectivity toward the far end.

Yellow alarm indicates reception from the far end of a data or framing pattern that reports the far end is in “red alarm.” Red alarm and yellow alarm states cannot exist simultaneously on a single piece of equipment because the “yellow alarm” pattern must be received within a framed signal. For ESF framed signals, all bits of the Data Link channel within the framing are set to data “0”; the customer data is undisturbed. For D4 framed signals, the pattern sent to indicate to the far end that inbound framing has been lost is a coercion of the framed data so that bit 2 of each timeslot is set to data “0” for three consecutive frames. Although this works well for voice circuits, the data pattern can occur frequently when carrying digital data and will produce transient “yellow alarm” states, making ESF a better alternative for data circuits.

Blue alarm indicates a disruption in the communication path between the terminal equipment. Communication devices, such as repeaters and multiplexers must see and produce line activity at the DS1 rate. If no signal is received that fills those requirements, the communications device produces a series of pulses on its output side to maintain the required activity. Those pulses represent data “1” in all data and all framing time slots. This signal maintains communication integrity while providing no framing to the terminal equipment. The receiving equipment displays a “red alarm” and sends the signal for “yellow alarm” to the far end because it has no framing, but at maintenance interfaces the equipment will report “AIS” or Alarm Indication Signal. AIS is also called “all ones” because of the data and framing pattern.

These alarm states are also lumped under the term Carrier Group Alarm (CGA). The meaning of CGA is that connectivity on the digital carrier has failed. The result of the CGA condition varies depending on the equipment function. Voice equipment typically coerces the robbed bits for signaling to a state that will result in the far end properly handling the condition, while applying an often different state to the customer equipment connected to the alarmed equipment. Simultaneously, the customer data is often coerced to a 0x7F pattern, signifying a zero-voltage condition on voice equipment. Data equipment usually passes whatever data may be present, if any, leaving it to the customer equipment to deal with the conditi

Time-division multiplexing (TDM) is a type of digital or (rarely) analog multiplexing in which two or more signals or bit streams are transferred apparently

simultaneously as sub-channels in one communication channel, but are physically taking turns on the channel. The time domain is divided into several recurrent

timeslots of fixed length, one for each sub-channel. A sample byte or data block of sub-channel 1 is transmitted during timeslot 1, sub-channel 2 during timeslot 2, etc.

One TDM frame consists of one timeslot per sub-channel. After the last sub-channel the cycle starts all over again with a new frame, starting with the second sample, byte

or data block from sub-channel 1, etc.

Time-division multiplexing (TDM) is a type of digital or (rarely) analog multiplexing in which two or more signals or bit streams are transferred apparently simultaneously as sub-channels in one communication channel, but are physically taking turns on the channel. The time domain is divided into several recurrent timeslots of fixed length, one for each sub-channel. A sample byte or data block of sub-channel 1 is transmitted during timeslot 1, sub-channel 2 during timeslot 2, etc. One TDM frame consists of one timeslot per sub-channel. After the last sub-channel the cycle starts all over again with a new frame, starting with the second sample, byte or data block from sub-channel 1, etc.

Contents[hide]

1 Application examples 2 TDM versus packet mode communication 3 History

o 3.1 Transmission using Time Division Multiplexing (TDM)

4 Synchronous time division multiplexing (Sync TDM) 5 Synchronous digital hierarchy (SDH) 6 Statistical time-division multiplexing (Stat TDM) 7 Notes 8 References

9 See also

[edit] Application examples The plesiochronous digital hierarchy (PDH) system, also known as the

PCM system, for digital transmission of several telephone calls over the same

four-wire copper cable (T-carrier or E-carrier) or fiber cable in the circuit switched digital telephone network

The SDH and synchronous optical networking (SONET) network transmission standards, that have surpassed PDH.

The RIFF (WAV) audio standard interleaves left and right stereo signals on a per-sample basis

The left-right channel splitting in use for stereoscopic liquid crystal shutter glasses

TDM can be further extended into the time division multiple access (TDMA) scheme, where several stations connected to the same physical medium, for example sharing the same frequency channel, can communicate. Application examples include:

The GSM telephone system

[edit] TDM versus packet mode communication

In its primary form, TDM is used for circuit mode communication with a fixed number of channels and constant bandwidth per channel.

Bandwidth Reservation distinguishes time-division multiplexing from statistical multiplexing such as packet mode communication (also known as statistical time-domain multiplexing, see below) i.e. the time-slots are recurrent in a fixed order and pre-allocated to the channels, rather than scheduled on a packet-by-packet basis. Statistical time-domain multiplexing resembles, but should not be considered the same as time-division multiplexing.

In dynamic TDMA, a scheduling algorithm dynamically reserves a variable number of timeslots in each frame to variable bit-rate data streams, based on the traffic demand of each data stream. Dynamic TDMA is used in

HIPERLAN/2 ; IEEE 802.16a .

[edit] History

Time-division multiplexing was first developed in telegraphy; see multiplexing in telegraphy: Émile Baudot developed a time-multiplexing system of multiple Hughes machines in the 1870s.

For the SIGSALY encryptor of 1943, see PCM.

In 1962, engineers from Bell Labs developed the first D1 Channel Banks, which combined 24 digitised voice calls over a 4-wire copper trunk between Bell central office analogue switches. A channel bank sliced a 1.544 Mbit/s digital signal into 8,000 separate frames, each composed of 24 contiguous bytes. Each byte represented a single telephone call encoded into a constant bit rate signal of 64 Kbit/s. Channel banks used a byte's fixed position (temporal alignment) in the frame to determine which call it belonged to.[1]

[edit] Transmission using Time Division Multiplexing (TDM)

In circuit switched networks such as the Public Switched Telephone Network (PSTN) there exists the need to transmit multiple subscribers’ calls along the same transmission medium.[2] To accomplish this, network designers make use of TDM. TDM allows switches to create channels, also known as tributaries, within a transmission stream.[2] A standard DS0 voice signal has a data bit rate of 64 kbit/s, determined using Nyquist’s sampling criterion.[2][3] TDM takes frames of the voice signals and multiplexes them into a TDM frame which runs at a higher bandwidth. So if the TDM frame consists of n voice frames, the bandwidth will be n*64 kbit/s.[2]

Each voice sample timeslot in the TDM frame is called a channel .[2] In European systems, TDM frames contain 30 digital voice channels, and in American systems, they contain 24 channels.[2] Both standards also contain extra bits (or bit timeslots) for signalling (see Signaling System 7) and synchronisation bits.[2]

Multiplexing more than 24 or 30 digital voice channels is called higher order multiplexing.[2] Higher order multiplexing is accomplished by multiplexing the standard TDM frames.[2] For example, a European 120 channel TDM frame is formed by multiplexing four standard 30 channel TDM frames.[2] At each higher order multiplex, four TDM frames from the immediate lower order are combined, creating multiplexes with a bandwidth of n x 64 kbit/s, where n = 120, 480, 1920, etc.[2]

[edit] Synchronous time division multiplexing (Sync TDM)

There are three types of (Sync TDM): T1, SONET/SDH (see below), and ISDN[4].

[edit] Synchronous digital hierarchy (SDH)

Plesiochronous digital hierarchy (PDH) was developed as a standard for multiplexing higher order frames.[2][3] PDH created larger numbers of channels by multiplexing the standard Europeans 30 channel TDM frames.[2] This solution worked for a while; however PDH suffered from several inherent drawbacks which ultimately resulted in the development of the Synchronous Digital Hierarchy (SDH). The requirements which drove the development of SDH were these:[2][3]

Be synchronous – All clocks in the system must align with a reference clock. Be service-oriented – SDH must route traffic from End Exchange to End

Exchange without worrying about exchanges in between, where the bandwidth can be reserved at a fixed level for a fixed period of time.

Allow frames of any size to be removed or inserted into an SDH frame of any size.

Easily manageable with the capability of transferring management data across links.

Provide high levels of recovery from faults. Provide high data rates by multiplexing any size frame, limited only by

technology. Give reduced bit rate errors.

SDH has become the primary transmission protocol in most PSTN networks.[2][3] It was developed to allow streams 1.544 Mbit/s and above to be multiplexed, in order to create larger SDH frames known as Synchronous Transport Modules (STM).[2] The STM-1 frame consists of smaller streams that are multiplexed to create a 155.52 Mbit/s frame.[2][3] SDH can also multiplex packet based frames e.g. Ethernet, PPP and ATM.[2]

While SDH is considered to be a transmission protocol (Layer 1 in the OSI Reference Model), it also performs some switching functions, as stated in the third bullet point requirement listed above.[2] The most common SDH Networking functions are these:

SDH Crossconnect – The SDH Crossconnect is the SDH version of a Time-Space-Time crosspoint switch. It connects any channel on any of its inputs to any channel on any of its outputs. The SDH Crossconnect is used in Transit Exchanges, where all inputs and outputs are connected to other exchanges.[2]

SDH Add-Drop Multiplexer – The SDH Add-Drop Multiplexer (ADM) can add or remove any multiplexed frame down to 1.544Mb. Below this level, standard TDM can be performed. SDH ADMs can also perform the task of an SDH Crossconnect and are used in End Exchanges where the channels from subscribers are connected to the core PSTN network.[2]

SDH network functions are connected using high-speed optic fibre. Optic fibre uses light pulses to transmit data and is therefore extremely fast.[2] Modern optic fibre transmission makes use of Wavelength Division Multiplexing (WDM) where signals transmitted across the fibre are transmitted at different wavelengths, creating additional channels for transmission.[2][3] This increases the speed and capacity of the link, which in turn reduces both unit and total costs.[2]

[edit] Statistical time-division multiplexing (Stat TDM)

STDM is an advanced version of TDM in which both the address of the terminal and the data itself are transmitted together for better routing. Using STDM allows bandwidth to be split over 1 line. Many college and corporate campuses use this type of TDM to logically distribute bandwidth.

If there is one 10MBit line coming into the building, STDM can be used to provide 178 terminals with a dedicated 56k connection (178 * 56k = 9.96Mb). A more common use however is to only grant the bandwidth when that much is needed. STDM does not reserve a time slot for each terminal, rather it assigns a slot when the terminal is requiring data to be sent or received.

This is also called asynchronous time-division multiplexing[4](ATDM), in an alternative nomenclature in which "STDM" or "synchronous time division multiplexing" designates the older method that uses fixed time slots.

Orthogonal Frequency-Division Multiple Access (OFDMA) is a multi-user version of the popular Orthogonal frequency-division multiplexing (OFDM) digital

modulation scheme. Multiple access is achieved in OFDMA by assigning subsets of subcarriers to individual users as shown in the illustration below. This allows simultaneous low data rate transmission from several users.

Contents[hide]

1 Key features o 1.1 Claimed advantages over CDMA o 1.2 Claimed advantages over OFDM with time-domain statistical

multiplexingo 1.3 Claimed OFDMA Advantages o 1.4 Recognised disadvantages of OFDMA

2 Characteristics and principles of operation 3 Usage 4 References 5 Literature 6 See also

7 External links

[edit] Key features

The advantages and disadvantages sumarized below are further discussed in the Characteristics and principles of operation section. See also the list of OFDM Key features.

[edit] Claimed advantages over CDMA

OFDM can combat multipath interference with more robustness and less complexity.[1]

OFDMA can achieve a higher MIMO spectral efficiency due to providing flatter frequency channels than a CDMA RAKE receiver can.[1]

No Cell size breathing as more users connect

[edit] Claimed advantages over OFDM with time-domain statistical multiplexing

Allows simultaneous low-data-rate transmission from several users. Pulsed carrier can be avoided. Lower maximum transmission power for low data rate users. Shorter delay, and constant delay. Contention-based multiple access (collision avoidance) is simplified. Further improves OFDM robustness to fading and interference.

[edit] Claimed OFDMA Advantages

Flexibility of deployment across various frequency bands with little needed modification to the air interface.[1]

Averaging interferences from neighboring cells, by using different basic carrier permutations between users in different cells.

Interferences within the cell are averaged by using allocation with cyclic permutations.

Enables orthogonality in the uplink by synchronizing users in time and frequency.[1]

Enables Single Frequency Network coverage, where coverage problem exists and gives excellent coverage.

Enables adaptive carrier allocation in multiplication of 23 carriers = nX23 carriers up to 1587 carriers (all data carriers).

Offers Frequency diversity by spreading the carriers all over the used spectrum.

Offers Time diversity by optional interleaving of carrier groups in time. Using the cell capacity to the utmost by adaptively using the highest

modulation a user can use, this is allowed by the gain added when less carriers are allocated (up to 18dB gain for 23 carrier allocation instead of 1587 carriers), therefore gaining in overall cell capacity.

[edit] Recognised disadvantages of OFDMA

Higher sensitivity to frequency offsets and phase noise.[1]

Asynchronous data communication services such as web access are characterized by short communication bursts at high data rate. Few users in a base station cell are transferring data simultaneously at low constant data rate.

The complex OFDM electronics, including the FFT algorithm and forward error correction, is constantly active independent of the data rate, which is inefficient from power consumption point of view, while OFDM combined with data packet scheduling may allow that the FFT algorithm hibernates during certain time intervals.

The OFDM diversity gain, and resistance to frequency-selective fading, may partly be lost if very few sub-carriers are assigned to each user, and if the same carrier is used in every OFDM symbol. Adaptive sub-carrier assignment based on fast feedback information about the channel, or sub-carrier frequency hopping, is therefore desirable.

Dealing with co-channel interference from nearby cells is more complex in OFDM than in CDMA. It would require dynamic channel allocation with advanced coordination among adjacent base stations.

The fast channel feedback information and adaptive sub-carrier assignment is more complex than CDMA fast power control.

[edit] Characteristics and principles of operation

Based on feedback information about the channel conditions, adaptive user-to-subcarrier assignment can be achieved. If the assignment is done sufficiently fast, this further improves the OFDM robustness to fast fading and narrow-band cochannel interference, and makes it possible to achieve even better system spectral efficiency.

Different number of sub-carriers can be assigned to different users, in view to support differentiated Quality of Service (QoS), i.e. to control the data rate and error probability individually for each user.

OFDMA resembles code division multiple access (CDMA) spread spectrum, where users can achieve different data rates by assigning a different code spreading factor or a different number of spreading codes to each user.

OFDMA can be seen as an alternative to combining OFDM with time division multiple access (TDMA) or time-domain statistical multiplexing, i.e. packet mode communication. Low-data-rate users can send continuously with low transmission power instead of using a "pulsed" high-power carrier. Constant delay, and shorter delay, can be achieved.

OFDMA can also be described as a combination of frequency domain and time domain multiple access, where the resources are partitioned in the time-frequency space, and slots are assigned along the OFDM symbol index as well as OFDM sub-carrier index.

OFDMA is considered as highly suitable for broadband wireless networks, due to advantages including scalability and MIMO-friendliness, and ability to take advantage of channel frequency selectivity.[1]

In spectrum sensing cognitive radio, OFDMA is a possible approach to filling free radio frequency bands adaptively. Timo A. Weiss and Friedrich K. Jondral of the University of Karlsruhe proposed a spectrum Pooling system in which free bands sensed by nodes were immediately filled by OFDMA subbands.

[edit] Usage

OFDMA is used in:

the mobility mode of the IEEE 802.16 Wireless MAN standard, commonly referred to as WiMAX,

the IEEE 802.20 mobile Wireless MAN standard, commonly referred to as MBWA,

the downlink of the 3GPP Long Term Evolution (LTE) fourth generation mobile broadband standard. The radio interface was formerly named High Speed OFDM Packet Access (HSOPA), now named Evolved UMTS Terrestrial Radio Access (E-UTRA).

the Qualcomm Flarion Technologies Mobile Flash-OFDM the now defunct Qualcomm/3GPP2 Ultra Mobile Broadband (UMB)

project, intended as a successor of CDMA2000, but replaced by LTE.

OFDMA is also a candidate access method for the IEEE 802.22 Wireless Regional Area Networks (WRAN). The project aims at designing the first cognitive radio based standard operating in the VHF-low UHF spectrum (TV spectrum).

The term "OFDMA" is claimed to be a registered trademark by Runcom Technologies Ltd.[1], with various other claimants to the underlying technologies through patents.