azimuth

1. the direction of a celestial object from the observer, expressed as the angular distance from
the north or south point of the horizon to the point at which a vertical circle passing through
the object intersects the horizon.
2. (पपपपपपपपपप पप पप पपपपपप पपपपप पप पपपप, पपपपपपप पप पपपपप
पप पपपपपप पपपपप पप पपपपप पपपप पप पपप पपप पपपपपप पप पप पप,
पपप पप पपपपप पप पपपपपप पप पपपपपप पपपप पप पपपपपप पपपपप
पपपपपपप पप पपप पपपप पपप)

OR

The azimuth is the angle between a celestial body (sun, moon) and the North, measured clockwise
around the observer's horizon. It determines the direction of the celestial body. For example, a
celestial body due North has an azimuth of 0º, one due East 90º, one due South 180º and one due
West 270º.

Understanding Azimuth and

Elevation
Why are azimuth and elevation so important for us... the PhotoPillers? The answer is
simple: because we will need to calculate and set them to take advantage of the Find
option of the Planner to plan our ideal sun and moon shots in just seconds.

Azimuth and elevation are the two coordinates that define the position of a celestial
body (sun, moon) in the sky as viewed from a particular location at a particular time.

Let’s say that you wish the sun or moon to be in a determined position in the sky...
because you love the composition. That position in the sky is defined by the azimuth
and the elevation.
A representation of the azimuth and the elevation of the sun.

What’s the azimuth?

The azimuth is the angle between a celestial body (sun, moon) and the North,
measured clockwise around the observer's horizon.

It determines the direction of the celestial body. For example, a celestial body due North
has an azimuth of 0º, one due East 90º, one due South 180º and one due West 270º.
Map view representing the azimuths: 0º, 90º, 180º and 270º.
On the Planner, you’ll find the azimuth and elevation of the sun/moon for the selected
date and time on one of the top panels. The azimuth is also represented on the map by
the azimuth lines, and the elevation on the time bar.

To help you better understand how the azimuth is represented on PhotoPills, I've
drawn North and azimuth angle of the sun for two different moments on the following
screenshots.

The first one is telling you that on February 11th 2014 at 10:17am the sun was at
azimuth 136.5º; and the second one that on February 11th 2014 at 3:01pm the sun was
at azimuth 214.6º.
Sun at azimuth 136.5º.
 Sun at azimuth 214.6º.
Learn more about how sun and moon information is displayed on the Planner watching
the following video tutorial:

 Planner: sun and moon info.

TIPS
 Notice that the map view of the Planner is always oriented towards North.

What’s the elevation?

The elevation is the vertical angular distance between a celestial body (sun, moon) and
the observer's local horizon or, also called, the observer’s local plane.
For us, the elevation of the sun is the angle between the direction of the geometric
center of sun's apparent disk and the observer's local horizon.

We'll say that the sun/moon is at elevation of 12º when its geometric center is situated
at 12º above the observer’s local horizon or local plane.

The following two pictures show the elevation of the sun in two different positions of the
observer.

Observer at the sea level, local plane and elevation of the sun.
 Observer at the top of a mountain, local plane and elevation of the sun.
Once you've decided the position of the sun/moon you'd like to have in your image,
calculating the elevation is sometimes the hardest part when planning the shot.

The best way to learn how to calculate and set the azimuth and the elevation you need
is by having a look at a few real examples. The following video tutorials will teach you
how to do it for different situations, step-by-step:

 Planner: find sunrises and sunsets.

 Planner: find moonrises and moonsets.
 Planner: find a sun or a moon at a determined position.
 Planner: find a sun or a moon over a mountain.
In the case you prefer reading, I am sure you'll enjoy these How-to articles:

 How to find sunrises and sunsets.

 How to find moonrises and moonsets.
In conclusion, mastering azimuth and elevation will give you the power to plan any
photo you imagine with sun and moon, including: a full moon setting under a secret
stone arch, a sunrise between two giant rocks located on a magic beach, a sunset over
the main street in your hometown or a dramatic full moon appearing from behind a
nearby hill...

You're in control... just let your imagination run wild!

co-channel
denoting or relating to a radio transmission that is on the
same frequency channel as another
or

co-channel
1.
denoting or relating to a radio transmission that is on the same frequency channel as
another: co-channel interference

(pu A public land mobile network (PLMN) is any wireless communications

system intended for use by terrestrial subscribers in vehicles or on foot. Such
a system can stand alone, but often it is interconnected with a fixed system
such as the public switched telephone network (PSTN).

Radio frequency
This article is about the generic oscillation. For the radiation, see Radio wave. For the electronics, see Radio frequency
engineering.

"RF" redirects here. For other uses, see RF (disambiguation).

Radio frequency (RF) is any of the electromagnetic wave frequencies that lie in the range
extending from around 3 kHz to 300 GHz, which include those frequencies used in radio
communication or radar.[1] RF usually refers to electrical rather than mechanical oscillations.
However, mechanical RF systems do exist (see mechanical filter and RF MEMS).
Although radio frequency is a rate of oscillation, the term "radio frequency" or its abbreviation
"RF" are used as a synonym for radio – i.e., to describe the use of wireless communication, as
opposed to communication via electric wires. Examples include:
 Radio-frequency identification
 ISO/IEC 14443-2 Radio frequency power and signal interface[2]

Special properties of RF currentEdit

Electric currents that oscillate at radio frequencies have special properties not shared by direct
current or alternating current of lower frequencies.
 Energy from RF currents in conductors can radiate into space as electromagnetic waves (radio waves).
This is the basis of radiotechnology.
 RF current does not penetrate deeply into electrical conductors but tends to flow along their surfaces; this
is known as the skin effect.
 RF currents applied to the body are harmful, but often do not cause the painful sensation of electric
shock that lower frequency currents produce.[3][4] This is because the current changes direction too
quickly to trigger depolarization of nerve membranes. However they can cause serious superficial burns
called RF burns.
 RF current can easily ionize air, creating a conductive path through it. This property is exploited by "high
frequency" units used in electric arc welding, which use currents at higher frequencies than power
distribution uses.
 Another property is the ability to appear to flow through paths that contain insulating material, like
the dielectric insulator of a capacitor. This is because capacitive reactance in a circuit decreases with
frequency.
 In contrast, RF current can be blocked by a coil of wire, or even a single turn or bend in a wire. This is
because the inductive reactance of a circuit increases with frequency.
 When conducted by an ordinary electric cable, RF current has a tendency to reflect from discontinuities in
the cable such as connectors and travel back down the cable toward the source, causing a condition
called standing waves. Therefore, RF current must be carried by specialized types of cable
called transmission line.

Radio communicationEdit
To receive radio signals an antenna must be used. However, since the antenna will pick up
thousands of radio signals at a time, a radio tuner is necessary to tune into a particular frequency
(or frequency range).[5] This is typically done via a resonator – in its simplest form, a circuit with
a capacitor and an inductor form a tuned circuit. The resonator amplifies oscillations within a
particular frequency band, while reducing oscillations at other frequencies outside the band.
Another method to isolate a particular radio frequency is by oversampling (which gets a wide
range of frequencies) and picking out the frequencies of interest, as done in software defined
radio.
The distance over which radio communications is useful depends significantly on things other
than wavelength, such as transmitter power, receiver quality, type, size, and height of antenna,
mode of transmission, noise, and interfering signals. Ground waves, tropospheric
scatterand skywaves can all achieve greater ranges than line-of-sight propagation. The study
of radio propagation allows estimates of useful range to be made.
Frequency bandsEdit
Main article: Radio spectrum

The radio spectrum of frequencies is divided into bands with conventional names designated by
the International Telecommunications Union(ITU):
 Frequency Wavelength Designation Abbreviation[6] IEEE bands[7]

3–30 Hz 105–104 km Extremely low frequency ELF -

30–300 Hz 104–103 km Super low frequency SLF -

300–3000 Hz 103–100 km Ultra low frequency ULF -

3–30 kHz 100–10 km Very low frequency VLF -

30–300 kHz 10–1 km Low frequency LF -

300 kHz – 3 MHz 1 km – 100 m Medium frequency MF -

3–30 MHz 100–10 m High frequency HF HF

30–300 MHz 10–1 m Very high frequency VHF VHF

300 MHz – 3 GHz 1 m – 10 cm Ultra high frequency UHF UHF, L, S

3–30 GHz 10–1 cm Super high frequency SHF S, C, X, Ku, K, Ka

30–300 GHz 1 cm – 1 mm Extremely high frequency EHF Ka, V, W, mm

300 GHz – 3 THz 1 mm – 0.1 mm Tremendously high frequency THF -

Frequencies of 1 GHz and above are conventionally called microwave,[8] while frequencies of
300 GHz and above are designated millimeter wave. More detailed band designations are given
by the standard IEEE letter- band frequency designations[7] and the EU/NATO frequency
designations.[9]
In medicineEdit
Radio frequency (RF) energy, in the form of radiating waves or electrical currents, has been used
in medical treatments for over 75 years,[10]generally for minimally invasive surgeries
using radiofrequency ablation including the treatment of sleep apnea.[11] Magnetic resonance
imaging (MRI) uses radio frequency waves to generate images of the human body.[12]
Radio frequencies at non-ablation energy levels are sometimes used as a form of cosmetic
treatment that can tighten skin, reduce fat (lipolysis), or promote healing.[13]
RF diathermy is a medical treatment that uses RF induced heat as a form
of physical or occupational therapy and in surgical procedures. It is commonly used for muscle
relaxation. It is also a method of heating tissue electromagnetically for therapeutic purposes in
medicine. Diathermy is used in physical therapy and occupational therapy to deliver moderate
heat directly to pathologic lesions in the deeper tissues of the body. Surgically, the extreme heat
that can be produced by diathermy may be used to destroy neoplasms, warts, and infected
tissues, and to cauterize blood vessels to prevent excessive bleeding. The technique is
particularly valuable in neurosurgery and surgery of the eye. Diathermy equipment typically
operates in the short-wave radio frequency (range 1–100 MHz) or microwave energy (range
434–915 MHz).
Pulsed electromagnetic field therapy (PEMF) is a medical treatment that purportedly helps to
heal bone tissue reported in a recent NASA study. This method usually employs electromagnetic
radiation of different frequencies - ranging from static magnetic fields, through extremely low
frequencies (ELF) to higher radio frequencies (RF) administered in pulses.
Effects on the human bodyEdit
Extremely low frequency RFEdit
High-power extremely low-frequency RF with electric field levels in the low kV/m range is
known to induce perceivable currents within the human body that create an annoying tingling
sensation. These currents will typically flow to ground through a body contact surface such as
the feet, or arc to ground where the body is well insulated.[14][15]
MicrowavesEdit
Main article: Microwave burn

Microwave exposure at low-power levels below the Specific absorption rate set by government
regulatory bodies are considered harmless non-ionizing radiation and have no effect on the
human body. However, levels above the Specific absorption rate set by the U.S. Federal
Communications Commission are considered potentially harmful (see Mobile phone radiation
and health).
Long-term human exposure to high-levels of microwaves is recognized to
cause cataracts according to experimental animal studies and epidemiological studies. The
mechanism is unclear but may include changes in heat sensitive enzymes that normally protect
cell proteins in the lens. Another mechanism that has been advanced is direct damage to the lens
from pressure waves induced in the aqueous humor.
High-power exposure to microwave RF is known to create a range of effects from lower to
higher power levels, ranging from unpleasant burning sensation on the skin and microwave
auditory effect, to extreme pain at the mid-range, to physical burning and blistering of skin and
internals at high power levels (see microwave burn).
General RF exposureEdit
The 1999 revision of Canadian Safety Code 6 recommended electric field limits of 100 kV/m for
pulsed EMF to prevent air breakdown and spark discharges, mentioning rationale related
to auditory effect and energy-induced unconsciousness in rats.[16] The pulsed EMF limit was
removed in later revisions, however.[17]
For health effects see electromagnetic radiation and health.
For high-power RF (electromagnetic, not electrical) exposure see radiation burn.
For low-power RF exposure see radiation-induced cancer.
As a weaponEdit
See also: Directed energy weapons § Microwave weapons

A heat ray is an RF harassment device that makes use of microwave radio frequencies to create
an unpleasant heating effect in the upper layer of the skin. A publicly known heat ray weapon
called the Active Denial System was developed by the US military as an experimental weapon to
deny the enemy access to an area. A death ray is a weapon that delivers heat ray electromagnetic
energy at levels that injure human tissue. The inventor of the death ray, Harry Grindell
Matthews, claims to have lost sight in his left eye while developing his death ray weapon based
on a primitive microwave magnetron from the 1920s (note that a typical microwave
oven induces a tissue damaging cooking effect inside the oven at about 2 kV/m).
MeasurementEdit
Since radio frequency radiation has both an electric and a magnetic component, it is often
convenient to express intensity of radiation field in terms of units specific to each component.
The unit volts per meter (V/m) is used for the electric component, and the unit amperes per
meter(A/m) is used for the magnetic component. One can speak of an electromagnetic field, and
these units are used to provide information about the levels of electric and magnetic field
strength at a measurement location.
Another commonly used unit for characterizing an RF electromagnetic field is power density.
Power density is most accurately used when the point of measurement is far enough away from
the RF emitter to be located in what is referred to as the far field zone of the radiation
pattern.[18] In closer proximity to the transmitter, i.e., in the "near field" zone, the physical
relationships between the electric and magnetic components of the field can be complex, and it is
best to use the field strength units discussed above. Power density is measured in terms of power
per unit area, for example, milliwatts per square centimeter (mW/cm²). When speaking of
frequencies in the microwave range and higher, power density is usually used to express
intensity since exposures that might occur would likely be in the far field zone.

throughput
1. the amount of material or items passing through a system or process.

Throughput
This article is about the amount of processed data in communication networks. For hard disk data in
particular, see Throughput (disk drive). For business management, see Throughput (business).
In general terms, throughput is the maximum rate of production or the maximum rate
at which something can be processed.
When used in the context of communication networks, such as Ethernet or packet radio,
throughput or network throughput is the rate of successful message delivery over a
communication channel. The data these messages belong to may be delivered over a
physical or logical link, or it can pass through a certain network node. Throughput is
usually measured in bits per second (bit/s or bps), and sometimes in data packets per
second (p/s or pps) or data packets per time slot.
The system throughput or aggregate throughput is the sum of the data rates that are
delivered to all terminals in a network.[1] Throughput is essentially synonymous to digital
bandwidth consumption; it can be analyzed mathematically by applying the queueing
theory, where the load in packets per time unit is denoted as the arrival rate (λ), and the
throughput, in packets per time unit, is denoted as the departure rate (μ).
The throughput of a communication system may be affected by various factors,
including the limitations of underlying analog physical medium, available processing
power of the system components, and end-user behavior. When various protocol
overheads are taken into account, useful rate of the transferred data can be significantly
lower than the maximum achievable throughput; the useful part is usually referred to
as goodput.
Maximum throughput
See also: Peak information rate
Users of telecommunications devices, systems designers, and researchers into
communication theory are often interested in knowing the expected performance of a
system. From a user perspective, this is often phrased as either "which device will get
my data there most effectively for my needs?", or "which device will deliver the most
data per unit cost?". Systems designers are often interested in selecting the most
effective architecture or design constraints for a system, which drive its final
performance. In most cases, the benchmark of what a system is capable of, or its
"maximum performance" is what the user or designer is interested in. When examining
throughput, the term maximum throughput is frequently used where end-user maximum
throughput tests are discussed in detail.
Maximum throughput is essentially synonymous to digital bandwidth capacity.
Four different values have meaning in the context of "maximum throughput", used in
comparing the 'upper limit' conceptual performance of multiple systems. They are
'maximum theoretical throughput', 'maximum achievable throughput', and 'peak
measured throughput' and 'maximum sustained throughput'. These represent different
quantities and care must be taken that the same definitions are used when comparing
different 'maximum throughput' values. Comparing throughput values is also dependent
on each bit carrying the same amount of information. Data compression can
significantly skew throughput calculations, including generating values greater than
100%. If the communication is mediated by several links in series with different bit rates,
the maximum throughput of the overall link is lower than or equal to the lowest bit rate.
The lowest value link in the series is referred to as the bottleneck.
Maximum theoretical throughput
This number is closely related to the channel capacity of the system,[2] and is the
maximum possible quantity of data that can be transmitted under ideal circumstances.
In some cases this number is reported as equal to the channel capacity, though this can
be deceptive, as only non-packetized systems (asynchronous) technologies can
achieve this without data compression. Maximum theoretical throughput is more
accurately reported to take into account format and specification overhead with best
case assumptions. This number, like the closely related term 'maximum achievable
throughput' below, is primarily used as a rough calculated value, such as for
determining bounds on possible performance early in a system design phase
Asymptotic throughput
The asymptotic throughput (less formal asymptotic bandwidth) for a packet-
mode communication network is the value of the maximum throughput function, when
the incoming network load approaches infinity, either due to a message size as it
approaches infinity,[3] or the number of data sources is very large. As other bit
rates and data bandwidths, the asymptotic throughput is measured in bits per
second (bit/s), very seldom bytes per second (B/s), where 1 B/s is 8 bit/s. Decimal
prefixes are used, meaning that 1 Mbit/s is 1000000 bit/s.
Asymptotic throughput is usually estimated by sending or simulating a very large
message (sequence of data packets) through the network, using a greedy source and
no flow controlmechanism (i.e. UDP rather than TCP), and measuring the network path
throughput in the destination node. Traffic load between other sources may reduce this
maximum network path throughput. Alternatively, a large number of sources and sinks
may be modeled, with or without flow control, and the aggregate maximum network
throughput measured (the sum of traffic reaching its destinations). In a network
simulation model with infinite packet queues, the asymptotic throughput occurs when
the latency (the packet queuing time) goes to infinity, while if the packet queues are
limited, or the network is a multi-drop network with many sources, and collisions may
occur, the packet-dropping rate approaches 100%.
A well known application of asymptotic throughput is in modeling point-to-point
communication where (following Hockney) message latency T(N) is modeled as a
function of message length N as T(N) = (M + N)/A where A is the asymptotic bandwdith
and M is the half-peak length.[4]
As well as its use in general network modeling, asymptotic throughput is used in
modeling performance on massively parallel computer systems, where system
operation is highly dependent on communication overhead, as well as processor
performance.[5] In these applications, asymptotic throughput is used in Xu and Hwang
model (more general than Hockney's approach) which includes the number of
processors, so that both the latency and the asymptotic throughput are functions of the
number of processors.[6]
Peak measured throughput
The above values are theoretical or calculated. Peak measured throughput is
throughput measured by a real, implemented system, or a simulated system. The value
is the throughput measured over a short period of time; mathematically, this is the limit
taken with respect to throughput as time approaches zero. This term is synonymous
with instantaneous throughput. This number is useful for systems that rely on burst data
transmission; however, for systems with a high duty cycle this is less likely to be a
useful measure of system performance.
Maximum sustained throughput
This value is the throughput averaged or integrated over a long time (sometimes
considered infinity). For high duty cycle networks this is likely to be the most accurate
indicator of system performance. The maximum throughput is defined as the asymptotic
throughput when the load (the amount of incoming data) is very large. In packet
switched systems where the load and the throughput always are equal (where packet
loss does not occur), the maximum throughput may be defined as the minimum load in
bit/s that causes the delivery time (the latency) to become unstable and increase
towards infinity. This value can also be used deceptively in relation to peak measured
throughput to conceal packet shaping.
Channel utilization and efficiency
Throughput is sometimes normalized and measured in percentage, but normalization
may cause confusion regarding what the percentage is related to. Channel
utilization, channel efficiency and packet drop rate in percentage are less ambiguous
terms.
The channel efficiency, also known as bandwidth utilization efficiency, is the percentage
of the net bitrate (in bit/s) of a digital communication channel that goes to the actually
achieved throughput. For example, if the throughput is 70 Mbit/s in a 100 Mbit/s
Ethernet connection, the channel efficiency is 70%. In this example, effective 70 Mbit of
data are transmitted every second.
Channel utilization is instead a term related to the use of the channel disregarding the
throughput. It counts not only with the data bits but also with the overhead that makes
use of the channel. The transmission overhead consists of preamble sequences, frame
headers and acknowledge packets. The definitions assume a noiseless channel.
Otherwise, the throughput would not be only associated to the nature (efficiency) of the
protocol but also to retransmissions resultant from quality of the channel. In a simplistic
approach, channel efficiency can be equal to channel utilization assuming that
acknowledge packets are zero-length and that the communications provider will not see
any bandwidth relative to retransmissions or headers. Therefore, certain texts mark a
difference between channel utilization and protocol efficiency.
In a point-to-point or point-to-multipoint communication link, where only one terminal is
transmitting, the maximum throughput is often equivalent to or very near the physical
data rate (the channel capacity), since the channel utilization can be almost 100% in
such a network, except for a small inter-frame gap.
For example, the maximum frame size in Ethernet is 1526 bytes: up to 1500 bytes for
the payload, eight bytes for the preamble, 14 bytes for the header, and four bytes for the
trailer. An additional minimum interframe gap corresponding to 12 bytes is inserted after
each frame. This corresponds to a maximum channel utilization of 1526 / (1526 + 12) ×
100% = 99.22%, or a maximum channel use of 99.22 Mbit/s inclusive of Ethernet
datalink layer protocol overhead in a 100 Mbit/s Ethernet connection. The maximum
throughput or channel efficiency is then 1500 / (1526 + 12) = 97.5 Mbit/s, exclusive of
the Ethernet protocol overhead.
Factors affecting throughput
The throughput of a communication system will be limited by a huge number of factors.
Some of these are described below:
Analog limitations
The maximum achievable throughput (the channel capacity) is affected by the
bandwidth in hertz and signal-to-noise ratio of the analog physical medium.
Despite the conceptual simplicity of digital information, all electrical signals traveling
over wires are analog. The analog limitations of wires or wireless systems inevitably
provide an upper bound on the amount of information that can be sent. The dominant
equation here is the Shannon-Hartley theorem, and analog limitations of this type can
be understood as factors that affect either the analog bandwidth of a signal or as factors
that affect the signal to noise ratio. The bandwidth of wired systems can be in fact
surprisingly narrow, with the bandwidth of Ethernet wire limited to approximately 1 GHz,
and PCB traces limited by a similar amount.
Digital systems refer to the 'knee frequency',[7] the amount of time for the digital voltage
to rise from 10% of a nominal digital '0' to a nominal digital '1' or vice versa. The knee
frequency is related to the required bandwidth of a channel, and can be related to the 3
db bandwidth of a system by the equation:[8]
{\displaystyle \ F_{3dB}\approx K/T_{r}}
Where Tr is the 10% to 90% rise time, and K is a constant of proportionality related to
the pulse shape, equal to 0.35 for exponential rise, and 0.338 for Gaussian rise.

 RC losses: wires have an inherent resistance, and an inherent capacitance when

measured with respect to ground. This leads to effects called parasitic
capacitance, causing all wires and cables to act as RC lowpass filters.
 Skin effect: As frequency increases, electric charges migrate to the edges of
wires or cable. This reduces the effective cross sectional area available for
carrying current, increasing resistance and reducing the signal to noise ratio.
For AWG 24 wire (of the type commonly found in Cat 5e cable), the skin effect
frequency becomes dominant over the inherent resistivity of the wire at 100 kHz.
At 1 GHz the resistivity has increased to 0.1 ohms/inch.[9]
 Termination and ringing: For long wires (wires longer than 1/6 wavelengths can
be considered long) must be modeled as transmission lines and take termination
into account. Unless this is done, reflected signals will travel back and forth
across the wire, positively or negatively interfering with the information-carrying
signal.[10]
 Wireless Channel Effects: For wireless systems, all of the effects associated with
wireless transmission limit the SNR and bandwidth of the received signal, and
therefore the maximum number of bits that can be sent.

IC hardware considerations
Computational systems have finite processing power, and can drive finite current.
Limited current drive capability can limit the effective signal to noise ratio for
high capacitance links.
Large data loads that require processing impose data processing requirements on
hardware (such as routers). For example, a gateway router supporting a
populated class B subnet, handling 10 x 100 Mbit/s Ethernet channels, must examine
16 bits of address to determine the destination port for each packet. This translates into
81913 packets per second (assuming maximum data payload per packet) with a table of
2^16 addresses this requires the router to be able to perform 5.368 billion lookup
operations per second. In a worse case scenario, where the payloads of each Ethernet
packet are reduced to 100 bytes, this number of operations per second jumps to 520
billion. This router would require a multi-teraflop processing core to be able to handle
such a load.

 CSMA/CD and CSMA/CA "backoff" waiting time and frame retransmissions after
detected collisions. This may occur in Ethernet bus networks and hub networks,
as well as in wireless networks.
 flow control, for example in the Transmission Control Protocol (TCP) protocol,
affects the throughput if the bandwidth-delay product is larger than the TCP
window, i.e. the buffer size. In that case the sending computer must wait for
acknowledgement of the data packets before it can send more packets.
  TCP congestion avoidance controls the data rate. So called "slow start" occurs in
the beginning of a file transfer, and after packet drops caused by router
congestion or bit errors in for example wireless links.

Multi-user considerations
Ensuring that multiple users can harmoniously share a single communications link
requires some kind of equitable sharing of the link. If a bottle neck communication link
offering data rate R is shared by "N" active users (with at least one data packet in
queue), every user typically achieves a throughput of approximately R/N, if fair
queuing best-effort communication is assumed.

 Packet loss due to Network congestion. Packets may be dropped in switches and
routers when the packet queues are full due to congestion.
 Packet loss due to bit errors.
 Scheduling algorithms in routers and switches. If fair queuing is not provided,
users that send large packets will get higher bandwidth. Some users may be
prioritized in a weighted fair queuing (WFQ) algorithm if differentiated or
guaranteed quality of service (QoS) is provided.
 In some communications systems, such as satellite networks, only a finite
number of channels may be available to a given user at a given time. Channels
are assigned either through preassignment or through Demand Assigned
Multiple Access (DAMA).[11] In these cases, throughput is quantized per channel,
and unused capacity on partially utilized channels is lost..

Goodput and overhead

Main article: Goodput
The maximum throughput is often an unreliable measurement of perceived bandwidth,
for example the file transmission data rate in bits per seconds. As pointed out above,
the achieved throughput is often lower than the maximum throughput. Also, the protocol
overhead affects the perceived bandwidth. The throughput is not a well-defined metric
when it comes to how to deal with protocol overhead. It is typically measured at a
reference point below the network layer and above the physical layer. The most simple
definition is the number of bits per second that are physically delivered. A typical
example where this definition is practiced is an Ethernet network. In this case the
maximum throughput is the gross bitrate or raw bitrate.
However, in schemes that include forward error correction codes (channel coding), the
redundant error code is normally excluded from the throughput. An example
in modemcommunication, where the throughput typically is measured in the interface
between the Point-to-Point Protocol (PPP) and the circuit switched modem connection.
In this case the maximum throughput is often called net bitrate or useful bitrate.
To determine the actual data rate of a network or connection, the "goodput"
measurement definition may be used. For example, in file transmission, the "goodput"
corresponds to the file size (in bits) divided by the file transmission time. The "goodput"
is the amount of useful information that is delivered per second to the application
layer protocol. Dropped packets or packet retransmissions as well as protocol overhead
are excluded. Because of that, the "goodput" is lower than the throughput. Technical
factors that affect the difference are presented in the "goodput" article.
Other uses of throughput for data
Integrated Circuits
Often, a block in a data flow diagram has a single input and a single output, and operate
on discrete packets of information. Examples of such blocks are FFT modules or binary
multipliers. Because the units of throughput are the reciprocal of the unit for propagation
delay, which is 'seconds per message' or 'seconds per output', throughput can be used
to relate a computational device performing a dedicated function such as
an ASIC or embedded processor to a communications channel, simplifying system
analysis.
Wireless and cellular networks
In wireless networks or cellular systems, the system spectral efficiency in bit/s/Hz/area
unit, bit/s/Hz/site or bit/s/Hz/cell, is the maximum system throughput (aggregate
throughput) divided by the analog bandwidth and some measure of the system
coverage area.
Over analog channels
Throughput over analog channels is defined entirely by the modulation scheme, the
signal to noise ratio, and the available bandwidth. Since throughput is normally defined
in terms of quantified digital data, the term 'throughput' is not normally used; the term
'bandwidth' is more often used instead.

Throughput is a measure of how many units of information a system can process in a given
amount of time. It is applied broadly to systems ranging from various aspects of computer
and network systems to organizations. Related measures of system productivity include ,
the speed with which some specific workload can be completed, and response time, the
amount of time between a single interactive user request and receipt of the response.

What is CSFB and SRVCC in LTE?

Regardless of the pace of LTE networks deployment around the world (faster in some areas,
slower in others), the number of users with 4G devices is intensively growing.

Thanks to factors such as lower costs - due to the gain of production scale, and also by
encouraging migration to 4G plans - offered by operators who already have an available
network, more and more people have access to new services and benefits that this
technology offers.

However, as much as the current data services are improved, and that progress in the area
lead to the adoption of new services, a basic necessity should still continue to exist at least
for a while: voice calls!
While making a voice call may seem simple, largely depends on the scenario where the user
is, and alternatives available for its completion. So it is necessary to understand well what
are the possibilities and the most important concepts of these key scenarios.

In the first generation of cellular networks, the communication through voice calls was the
main goal, and was based on a circuit switched topology or 'channels' (CS Circuited
Switched).

Over time, the need for other services (data!) has emerged. Voice calls have come into
existence with these new services. As demand increasead, these new services were
supported by a new domain, the IP-based packet-switched (PS Packet Switched). The
following figure shows how these two domains work.
And in LTE (4G) system we had another great change: the CS domain has been
extinguished! LTE networks are based exclusively on the PS domain, and voice services
should be carried out in other ways (as we shall see).

But as we mentioned, regardless of network topologies, voice services are still needed. (Of
course, they slightly decreased compared to a few years ago, but are still significant, ie their
demand continues).

With the continue growth of LTE networks, let's try to understand a little more the concepts,
alternatives and solutions for any user to make a voice call on an LTE network?

Note: All telecomHall articles are originally written in Portuguese. Following we translate to
English and Spanish. As our time is short, maybe you find some typos (sometimes we just
use the automatic translator, with only a final and 'quick' review). We apologize and we
have an understanding of our effort. If you want to contribute translating / correcting of
these languages, or even creating and publishing your tutorials, please contact us: contact.

How, when and where?

First of all, we need to understand how, when and where voice calls can occur.

In the 2G legacy networks, voice calls are made practically only on circuits - for each call
(CS domain).

In 3G legacy networks, voice services can use the CS domain, but can also be made
through OTT (Over The Top) solutions, applications that encapsulate the voice and transport
via an IP domain (PS), but who lack the QoS requirements needed to ensure good
communication - with the Non GBR type services (no bit rate guarantee). Example: Skype.
Note: It is very unusual, but it is also possible to make OTT voice calls on 2G networks. In
fact, there may be OTT solutions in any technology - it can be used in legacy networks, and
also in others such as WiFi - which are already commonly used for VoIP.

And in LTE networks, voice calls can be fully IP-based, can use OTT solutions via 4G, or be
transferred to the legacy 2G/3G.
As we begin to see, there are many alternatives. As usual, we will easily see each one.

Note: In this tutorial, we will always refer to voice calls (originating and/or terminating);
However, SMS services are also included.

Alternative to voice calls in a generic 2G-3G-4G Network Topology

And the best way to understand the alternatives or possibilities of making voice calls in LTE
network (4G), it is to start from a 2G-3G-4G network topology very simplified - considering
only the main elements involved.

As we can see in the following figure, the LTE (EPC) has no direct 'link' to the CS network -
as we have seen, it is designed to take care of purely IP (PS) calls. It has no Media Gateway
directly connected, so no CS call is supported by the MME.
In other words, if the user or UE (User Equipment) is on a LTE network, as shown in the
topology above, we can not make a voice call.

Note: As mentioned before and according to the topology above, the only way to have voice
services in LTE would be through OTT services such as Skype. However, this solution is not
discussed today.

If we understand this, it is also easy to realize that in order for we to have voice services in
LTE, changes need to be made. There are some alternatives, and below we have the main
ones:

 VoLGA (Voice over LTE via Generic Access): Use legacy 2G/3G as a generic access, 'packaging'
voice services, and delivering via LTE.
 CSFB (CS Fall Back): whenever the UE have the need to place a call, make it revert (fallback) for
legacy networks.
 VoLTE (Voice over LTE): make voice over LTE itself. In this case, the voice is pure IP - VoIP LTE.
o SRVCC (Single Radio Voice Call Continuity): ensure that purely LTE (VoLTE) calls are
transferred (via handover) to the legacy networks in a transparent manner.

Note: notice that the SRVCC is an option when the voice call has been established in LTE. Ie
it is a conditional alternative - considering that VoLTE option has been used.

Even without knowing very well the options presented, it is easy to imagine that the 'best'
solution would carry voice over their own LTE network. But like everything in life, it also
have the other side, the pros and cons.
To deliver voice services in LTE network is necessary to have an infrastructure that support
it. In other words, there needs to exist an IMS (IP Multimedia Subsystem or IP Multimedia
Core Network Subsystem). If an IMS is available, then the voice over LTE may be provided
as long as a minimum set of IMS functionality and entities also are present.

Note: IMS is much more complete, and have more other purposes than the voice. The voice
is just another 'application' of IMS, as we'll see soon.

This minimum set of features and entities of the IMS (called VoLTE or One Voice) was
standardized to enable LTE operators to provide voice services without having to make very
radical changes in the network (without having to invest in a complete IMS, with all entities
and functionality).

In any case, it requires investment.

And therefore the first two alternatives become attractive: based on legacy network CS
infrastructure. But if on the one hand such alternatives require less investment in LTE
network, these alternatives depend on the existing 2G/3G networks.

Let's talk a little more about each of these possibilities, but always trying to maintain the
overview, in the simplest possible way to understand. Remember that our goal is to learn
the concept, in order to enable a deepening on the subject, if desired, more easily.

VoLGA

The first implementation alternative that emerged was the VoLGA (Voice over LTE via
Generic Access), or: try to use what are already available, with minimal changes required.

To use the infrastructure of legacy 2G/3G networks, VoLGA introduces a new network
entity, the VNC (VoLGA Network Controller), which basically functions as a 2G BSC,
communicating with a GSM MSC (Mobile Switching Center) and as one 3G RNC
communicating with a UMTS MSC (Mobile Switching Center).
When we have a new call (be it originated or terminated), it is managed by the MSC of
legacy network. VNC is who mediates the voice signal and its related messages between the
MSC and the LTE network.

Although it is possible to carry out the delivery of voice and SMS services to users LTE, the
Volga was unsuccessful. This is because, as we have seen, exclusive investment are needed
for this purpose. At the same time however, global efforts to VoLTE increased (eg
investments in IMS), and thus this alternative eventually falled into disuse.

CSFB

But if in one hand operators follow seeking a complete LTE infrastructure (with full IMS) to
meet multimedia services and also purely LTE voice, this is not a topology that is available
in the short and even medium term.

While that reality doesn't come, we must use the legacy network when there is the need of
voice and SMS delivery to LTE users.

And the most common alternative to this is the CSFB (CS Fall Back), an interim solution
until we have full support for voice over LTE.
At CSFB scheme, whenever there is a demand for a new voice call, the LTE user is 'backed'
for a CS legacy network, assuming that this provides an overlapping coverage. In other
words, with CSFB, a voice call is never active in LTE, but in legacy networks.

At the end of the call in the legacy network, the UE can re-register the LTE network.

It goes something like this: the UE is registered (also) in the legacy network. When it got a
call, the legacy network tells to LTE network: 'I have a call to the UE, can you ask it to come
here and make the call?'

To CSFB be possible, users must be using dual mode devices, ie able to operate both in LTE
network and in the legacy network.

To support CSFB, a new interface is introduced: the SGs, connecting the MME to the legacy
MSC.

As the CSFB is currently the most widely used option by several operators, let's see some
basic scenarios of it (CSFB).

CSFB - Registration and Location

When the CSFB UE is turned on, it registers itself in the two networks: LTE and legacy
network (CS).
And to allow quick transfer to the legacy network (either 2G or 3G) when necessary, the LTE
network needs to know the location of the UE.

For this, the MME, which tracks the location of the UE in the LTE network, continuously
provides location information to the legacy MSC, using the new SGs interface.

The set of SGs messages then supports management of mobility, paging and SMS.

CSFB - Originated Call

We will continue, and assume that the UE is initially covered by the LTE network, and that
there is an active IP connection.

When the UE decides to originate a voice call, it sends an SRM (Service Request Message)
to the MME (more specifically the ESR - Extended Service Request).

The MME checks whether the UE is CSFB capable, and notifies the eNodeB to transfer the
UE to the legacy network.

Before performing the UE transfer, the eNodeB can ask it to make RF measures on
neighboring 2G/3G network. The eNodeB then decides the best network for the UE and
performs the transfer.

Once the UE camp in 2G/3G network, it starts the call procedure as usual - the UE starts the
call control procedures in legacy network.

CSFB – Call + Data Connection in LTE

And what happens if I have an active data connection in the IP LTE network, and decide to
make a voice call?

There are two options:

 The data are also transferred to the legacy network, or

 The data are temporarily suspended, until I return to the LTE network.

Although the first option seems the best, we must take into account that the transmission of
IP data is also transferred: it can operate at much lower speeds (legacy systems). In
addition, it may be that the legacy networks deny the IP session due to lack of resources or
for not being able to process it.

The S3 interface is used to carry out the PS session handover for 3G (in this case, the DTM -
Dual Transfer Mode must exist, but this details escapes form our theme today).

There are no 4G data handover supported to 2G - in this case, the data is suspended.
The eRABs 4G are released when the UE performs the CSFB.

An important information is that the S3 is a 'new' interface between MME and SGSN on
GTPCv2. And to support it, the SGSN needs to be updated (most carriers do not want to do
this without a strong justification).

And Gn interface is already on GTPCv1, which is the native GTP version for 3G networks. So
in this case only the MME needs to be updated, and as it is a relatively new node, it is
probably easier to do. Not to mention that the new SGSN may have native support for S3.

CSFB - Terminated Call

Finally, we have the case of a terminated call for LTE user.

The call request arrives first to the MSC where the UE was previously registered.

When the MSC the receives call request, it sends paging messages to the related MME via
SGs interface.

This message is forwarded to the UE, which is still connected to the LTE network.

If the user accepts the call, it sends an SRM (Service Request Message) to the MME.

The then MME notifies the eNodeB to transfer the UE for the legacy network, and the
eNodeB then decide the best network for the UE to make the call.

CSFB – What happens after the end of the CS call?

We have seen that the 4G eRABs are released when the UE performs the CSFB. But what
happens when the UE ends the CS call?

About what should follow next (if the UE should return or not to LTE as soon end the call
CS), there is no specific rule.

Anyway, the main possibilities are:

 The upper layers forcing the 'reselection' to LTE so that the UE enters idle mode in legacy network.
 The operator send LTE 'redirection' information in RRC connection release message of legacy 3G
network after the call is finished. This will result again in reselection to LTE.
 The lower layers (AS - Access Stratum in this case URRC or GRR) reselect to LTE if the reselection
criterion is satisfied. In most cases, operators have their parameters set such that the reselection to
LTE happen if there is a good LTE coverage area overlapping the legacy network.

VoLTE
Everything we have seen so far is based on the making of voice call in the legacy network.
But as we have seen these are 'temporary' solutions until the 'final' solution - VoLTE - is
available.

And the final LTE voice solution (Voice over IP, or more specifically VoLTE) uses the IMS
backbone. An example of network topology supporting VoLTE is shown in the following
figure.

To make voice calls, LTE networks need to have an IMS. When the first LTE networks
appeared, they had no IMS, and without IMS, it was not possible to make any calls to any
PSTN or CS.

We have spoken of the IMS before, but let's remember.

IMS

IMS is a backbone (network) at the application level, which works on top of other wireless
networks and not just the LTE (as 3G, 2G, WiFi and others).

Its concept is quite broad, and to understand it with all its entities, possibilities, interfaces,
protocols, and possibilities is an extremely difficult task, even for the most experienced in
the subject.
The IMS is not new: it already existed before the LTE (as well as other entities, such as the
EPC PRCF, which also is not new!).

Its complete specification consists of thousands and thousands of 3GPP standards. But let's
try to understand in a simpler way than that found there.

As its name suggests (IP Multimedia Services), IMS offers several multimedia IP services,
including VoIP (Voice over IP). In IMS, voice is just 'another' service!

IMS brings together voice features such as authentication, service authorization, call
control, routing, interoperability with PSTN, billing, additional services and VAS. None of
these exist in the EPC: this is the reason why the pure EPC without IMS can not process a
voice call.

In other words, for VoLTE, access is made by the SAE (eUTRAN + EPC), while voice service
lies in the IMS.

An analogy we can do is to consider the IMS being a car. And the LTE voice, as our shuttle
service (to go from one place to another).

 We can buy a very basic car - Basic 1.0 engine, wheels, steering wheel and other minimum parts:
yes, we can go from one place to another.

 Or we can buy a 'connected' car - ultra modern, powerful, tetra-fuel, with all the safety features,
ABS, Air bag, connected to the Internet, etc: we also go from one place to another ... but we can
make several other things as well!

That's more or less what happens with the IMS. It is used in conjunction with the LTE
network to support voice: both full IMS implementation and also the minimum IMS
suggested implementation for Voice over LTE.

But the telecommunications industry would rather not invest in a full IMS, or at least did not
have sufficient reason to do it immediately. And for the adoption of the simpler IMS voice
solution, appear the VoLTE initiative, which specifies a minimum set of features, and selects
a simple choice when multiple options exist for certain features.

However, not all of these features are required for delivery of basic voice services by the
LTE network.

So let's illustrate with a diagram (extremely simple) the implementation of a voice in IMS
(VoLTE).

 Let's assume that we will make a VoLTE call with a CS network whatsoever, for example the PSTN
(Public Switch Telephony Network).
 And consider in the IMS only two simple elements, one for the control plane (with signaling) and
one for the user plane (with voice).
 And the entry being the SAE, or LTE network.
 In IMS, the control element would be a SIP server (soon we will talk about SIP - for now just
understand that when we have a call request to this server, it sets up the call.); and the user
element would be a Media Gateway.

In comparison with the legacy networks, the SIP Server is equivalent to the MSC in the
mobile network topology and the media gateway is equivalent to a typical Media Gateway
on any network topology, which is common in virtually any voice network to handle calls.

The above concept is valid, but in practice the IMS consists of much more entities, as seen
below. Note: Not all possible/existing entities and interfaces are shown in the figure.
Let's (quickly) see a little about these key elements.

Note: Do not worry or try to understand everything now about these elements. Remember
that our goal here today is not that. Anyway, it's worth a read.

The MGCF (Media Gateway Controller Function) is the control element that communicates
with other PSTN networks. It is significant because it has to inter-networking function: can
speak SIP, can speak ISUP, can speak other signaling protocols.

The IM-MGW (IM Media Gateway) is the element that takes care of voice functions for
example making protocol translation required to support the call. More specifically between
the Real Time Transport Protocol (RTP) to analog format or basic PCM in the CS network;
and vice versa.

The HSS (Home Subscriber Server) is an element that also exists in the LTE EPC (although
appeared first in IMS), and its functions are similar.

The MRF (Media Resource Function) provides many services related to voice, such as
conferences, announcements, voice recognition and so on. It is always divided into two
parts, the MRFP (Media Resource Function Processor), for media streams, and the MRFC
(Media Resource Function Controller) that functions basically as an RTP 'mixer'.

An important concept, and that's worth stand out here is the Proxy, for example to make
filters, identify where the users come from, the cases of roaming, etc. Remember that we
are talking about an IP network. Instead of the users to speak directly with the SIP server,
they use the proxy.

The CSCF (Call Session Control Function) has some variations.

 O P-CSCF (Proxy CSCF) among other tasks, provides QoS information related to the LTE network.
Acccess an AF to voice service, and provides the control functions 'policy' and 'charging' to the
PCRF.
 O I-CSCF (Proxy CSCF) is an interrogator.
 And the S-CSCF (Serving CSCF): the CSCF server acts as a central node.

The BGCF (Border Gateway Control Function) functions as a routing table (or table B) and
acts to help the S-CSCF. It has basically routing decisions.

As we speak, the IMS voice is a 'service' - the IMS is a services 'facilitator'. The IMS services
are provided through AS (Application Servers).

One such application is the voice. And there are also video services, conference, etc.

In fact, sometimes the AS are not considered as part of IMS (when we understand the IMS
as a CORE).

And in IMS, the standard AS for voice is the MMTel (Multimedia Telephony Service),
sometimes called MTAS (Multimedia Telephony Application Server).

The SBC (Session Border Controller) is an element of the edges of the IMS to control
signaling and often the media streams involved in calls.

The S-CSCF will be responsible for call routing depending on where the other user (the
other party) are:

 A SBG (Session Border Gateway) if the the other party is in IP domain;

 A MGC/MGW if the other party is in the CS domain (PSTN/PLMN).

IBCF and TrGW are not shown in our figure, but are respectively the control and user plane
for other IMS networks, other SIP networks in general. They are similar to the MGCF/IM-
MGW - the requirements for reaching one or another type of network are different, so also
have separate parts for performing the same functions but with different networks.

SIP

To support telephone signaling between the LTE network and telephone networks, the IMS
uses SIP (Session Initiation Protocol). SIP is a standard protocol for establishing voice calls
over IP networks.
The code is open, and uses the 'request-response' model to allow communication sessions.

There is a set of standard commands that can be used to initiate, manage and terminate
calls between two SIP devices.

The SIP has been adopted by IMS standardization as the protocol to allow signaling between
telephone networks and VoIP networks.

SIP is text-based and was developed - in the 90s - in order to be simple and efficient, just
like the HTTP protocol (in fact, was inspired by HTTP and other protocols such as SMTP).

A good analogy is to compare the SIP with HTTP.

You probably can understand well the HTTP interaction principle, which allows audio
connection, text, video and other elements on a web page. With SIP is pretty much the
same thing: it allows the establishment, management and calls endings (or sessions) for IP
multi-users without knowing the content of the call. A session can be a simple telephone call
between two users, or a multi-user multimedia conference.

Both (SIP and HTTP) take the control of the application to the end user, regardless of the
transport protocol (SIP is a control protocol in the application layer), so there is no need for
switching centers/servers.

The SIP however is not a resource reservation protocol, and has nothing to do with QoS.

A short break: our tutorial today is already quite extensive, but we'll keep a little more with
this issue because these concepts are very important, and you'll be hearing a lot of them.

To try to understand it better, let's see a simplified example for a voice call establishment
process using IMS platform and SIP signaling.
  Initially, the UE sends a SIP message like 'Invite', containing the description of one or more
measures for the voice session (Initial SDP - Session Description Protocol - Offer).
 Then the P-CSCF forwards this same message to the S-CSCF (which has been identified during the
registration process).
 All going well, the termination network will have sent a message of type 'offer response' to the S-
CSCF, and this sends this message to the P-CSCF, authorizing the allocation of the resources
necessary for this session.
 Finally, the P-CSCF forwards the 'offer response' message back to the UE, which confirms the
receipt of the 'offer response' message and the resource reservation is started.

This is a very simplified example of how you can be getting (origination) of a voice service
by the UE, via IMS.

Several other diagrams exist, with far more complex scenarios, but the basic idea can be
seen above, and extended if necessary.
Let's complete the tutorial today, now seeing the case where an initially established call on
IMS has to be 'transferred'.

SRVCC

Finally we come to our last alternative listed at the beginning of this tutorial: SRVCC (Single
Radio Voice Call Continuity).

The SRVCC however is not an alternative for delivery, but a rather a handover process of a
voice call previously started in the LTE (whether One Voice - VoLTE LTE or IMS Full Voice).

It is a call transfer method (handover), in a simplified and reliably way, when an LTE user
has an active voice session in IMS and is moving to areas without LTE coverage, but with
legacy 2G/3G coverage.

The main advantage is that the call will not drop - will only be transferred to the CS domain
of the legacy networks.

If in the above case the UE moves out of LTE coverage area with an active call (but goes to
a legacy 2G/3G coverage), we must maintain the continuity of this active voice call. In this
case, the SRVCC is used: the procedure where the context of an active voice call on the IMS
is transferred to the CS legacy network (e.g. IMS node context transfer to the MSC).

The challenge with SRVCC is to perform the handover while the UE is connected to only a
single radio at any given moment.
There are two versions of SRVCC:

 SRVCC handover to GSM or UMTS, defined by 3GPP;

 SRVCC Handover to 1xRTT networks defined by the 3GPP2.

To allow SRVCC both the UE and LTE networks, as also the legacy, must support SRVCC.
For this, a new special SV interface is introduced between the MME and the MSC, which runs
on GTPv2 protocol.

To support SRVCC, the IMS network should also include an application server, called SCC AS
(Server Centralization and Continuity Application Server).

This application server is who manages the signaling required for the process.

Let's see a simplified example of some SRVCC procedures from LTE to GSM.

 When an UE that supports VoLTE is in an LTE coverage area, it starts voice sessions via the IMS
network, which will host the session and provide applications and session control based on SIP.
 When the UE moves from an LTE coverage area for a CS 2G/3G coverage area, with the active IMS
session, the IMS switches the session to the CS domain, maintaining both parts aware of the
handover session.

Example of SRVCC Handover

Realizing that its LTE signal level begins to decrease, the UE with an active IMS voice
session signals it to the eNodeB, initiating the SRVCC handover.

The eNodeB then identifies the best available network to receive the service, and sends the
handover request (specifying that it is the SRVCC type) to the MME.

The new voice call request is then sent to the IMS, using a SR STN (Session Transfer
Number for SRVCC) - a unique number that is generated by each UE, and is stored in HSS.

This unique number is sent by the MME to the HSS when the UE firts comes into contact
with the network.

Upon receiving the STN SR number, the SCC AS believes that the corresponding call should
be transferred to a different network network, and starts the redirecting process for the
transfer point (handover) to the legacy network.

After resource preparation is completed, the MME confirms the handover request, previously
provided by the eNodeB.

The eNodeB then transmits this acknowledgment to the UE, while still providing the required
information about the target network.

In the final stages, the UE is detected in legacy networks, and the call is re-established in it.

Thus we have the completion of the SRVCC handover.

Voice packets and also packets that are not voice can be transferred using this method, but
the data rates will be limited by the capabilities of the legacy networks.

Once the SRVCC is a procedure for inter-RAT handover based on IMS LTE network to the CS
legacy network 2G/3G, it is much more complex than that of handovers legacy networks 3G
/ 2G. The question is how to maintain a handover performance comparable to or better
acceptable.

In order to improve the performance of the SRVCC handover, one WI (Work Item) called
eSRVCC (SRVCC enhancement) was established in the 3GPP SA2 in Release 10. The
anchoring solution is based on the IMS, and introduces new entities ATCF (Transfer Control
Access Function ) and ATGW (Transfer Access Gateway).

Again, the deepening of this subject escapes from our goal today.

Finally, we will enumerate some of the main advantages and disadvantages (or pros and
cons) of each alternative.

Advantages and disadvantages of each alternative

Call setup time: When operators use CSFB, one of the biggest problems faced (and one of
the major disadvantages of CSFB) is the increase in call setup time due to retuning
procedures in 2G/3G radios.

An efficient CSFB solution requires the the TAC -> LAC mapping is so that the fallback to an
external MSC/LAC be avoided, since this will further increase the call setup time.

Call quality: call quality in LTE is better when compared with the same third-party
applications (OTT). This is due to specific QoS allocated to the call IMS, which may not be
present in common data applications.

Resource limitations for VoLTE: AMR-NW LTE requires much less resources and datarate
than GSM, and we will have many more users on the same bandwidth (spectral efficiency).

Investment x Current Network: if everything is 'working well', what would be the reason
for investment, since surely such investments generate resistance from commercial and
business areas?

The comparison that must be done is: Investment versus (all) Benefits of IMS/MGW/BGCF.

Future:In any way all that discussion hereafter will more significance. Currently we still
have extensive legacy networks, capable of supporting these voice calls.

In this case, it is no problem to continue using this available infrastructure. Resistance will
only decrease when such capacity also decrease. But in an LTE network, if the IMS is
supported can make a VoIP call. So why would we need to make a CS voice call?

CSFB x SRVCC:

 It is not necessary to implement both solutions (CSFB and SRVCC) at the same time, if the network
has a wide LTE coverage and a complete IMS backbone.
o If we implemente CSFB, it means we will not make the call setup using existing IMS Core,
and that could take care of that call in LTE.
o In respect to the SRVCC: assuming the Backbone IMS is available. In this case, if the
register in the IMS is successful, the user do not need to do CSFB - A voice call can be
simply initiated in LTE network using IMS.
 CSFB is a service handover procedure while SRVCC is a coverage handover procedure.

Case Studies and Analogies

With all that we have seen today, let's imagine some scenarios.

First, imagine that you are in a network that does not have LTE IMS. Then the only way to
make a voice call, whether originated or terminated, is through using legacy 2G/3G.

You need to be redirected/released from LTE to legacy 2G/3G network to make a voice call.
Like a 'reselection' from cell LTE to the 2G/3G. Once the legacy network, you can make the
call normally, as you're already used to.

And so, you just saw the CSFB in practice!

Now suppose you are watching a video stream on 4G network, and receive a voice call. In
this case, you need to go to the 3G network (in idle mode), and get the resources for to
make that call in 3G.

After you end your voice call, you keep watching the video stream, but now in the 3G
network (the handover from 3G to 4G is not yet defined).

You just saw the CSFB with an active data call!

Now let's imagine that you are in another LTE network, this time with IMS. In this case, you
can make a voice call using IP packets.

We have just seen a VoLTE call!

Further, imagine that you are in one of these voice calls using packets in 4G. Suppose
further you reach your 4G cell coverage edge. So the only option to keep your call is to
handover it to the 3G (assuming this is the existing coverage). Your call will then continue
on the 3G network, but now as one CS voice call. SRVCC!

If the SRVCC is not supported, the call is dropped as soon as it leaves the LTE coverage
area.

If the SRVCC is supported, a set of messages are exchanged, and the voice call is
transferred (handover) from LTE IMS to CS domain of the 2G/3G network.

And so, we have just seen an example of SRVCC handover!

And that's all for today. We hope that the tutorial has managed to be useful for you that
somehow are interested voice in LTE networks.

Conclusion

We saw in this tutorial today, in a very general way, the main ways to make voice calls (and
SMS) in LTE networks.

The options or alternatives depend on several factors, such as available network topology
and the operator's strategy.

Depending on the situation, the call can be originated in LTE via data applications (OTT
VoIP), be purely originated on LTE IMS (VoLTE), sent to be performed on other networks
through mechanisms developed for this purpose (CSFB) or transferred via handover - if
active VoLTE call - to a legacy network (SRVCC).

So, for a user who is a LTE coverage area, a number of considerations should be checked,
as the type of device that it uses (whether supports CSFB), if the LTE network has an IMS
that allows outgoing calls, if the cells supports SRVCC, etc.
Based on the concepts seen here today, we hope you have a position to fully understand
what happens when a user performs a voice call from an LTE network.

What is VoLTE?
VoLTE stands for voice over LTE and it’s more or less exactly what it says on the tin. It's
voice calls over a 4G LTE network, rather than the 2G or 3G connections which are
usually used. We tend to think of 4G as mostly being about downloading, streaming and
web browsing, and indeed that’s primarily what it’s been used for so far, but it can also
be used to improve calls.
What are the benefits of VoLTE?
Superior call quality - The big advantage of VoLTE is that call quality is superior to 3G
or 2G connections as far more data can be transferred over 4G than 2G or 3G. Up to
three times as much data as 3G and up to six times as much as 2G to be precise,
making it easier to make out not only what the person on the other end of the line is
saying, but also their tone of voice. Essentially it’s an HD voice call and it’s a much
richer experience over all. Improved coverage and connectivity - VoLTE can connect
calls up to twice as fast as the current methods and as 2G and 3G connections will still
be available when there’s no 4G signal it simply means that there’s greater mobile
coverage overall, as currently places with a 4G signal but no 2G or 3G means that one
you can’t make or receive calls. You might think that would be a rare occurrence, but
some of the frequencies that 4G operates on, such as the 800MHz spectrum, have far
greater reach than 2G or 3G spectrum, so you’ll be able to get signal further away from
a mast or in buildings which other signals struggle to penetrate. Indeed, Three is fully
relying on its 800MHz spectrum for VoLTE calls.
However, while 2G and 3G services would likely remain they wouldn’t be as necessary
as they are now and much of the spectrum used for 2G in particular could potentially be
repurposed to increase capacity on 4G networks.
Better battery life - Anyone who currently uses 4G could also find their battery life
increased with VoLTE, as right now whenever you make or receive a call your phone
has to switch from 4G to 2G or 3G, since 4G calls aren’t supported (other than on Three
Super-Voice) and then once the call is finished it switches back again. All that switching,
plus the need to search for a different signal each time, can give the battery a significant
hit.
Video calling - It’s also theoretically possible to make video calls over 4G, much like a
Skype call except you’d just use your mobile number and be able to use the regular
dialler and call interface, so you can make and receive video calls from anyone else
with VoLTE, rather than relying on separate accounts.
In fact you may have noticed that Skype and other existing video calls services often
seem to have superior audio quality to voice calls. That’s because like VoLTE they use
more data as part of a similarly named VoIP system, so you can expect your voice calls
to start sounding more like Skype calls, but they won’t hit your battery life as much as
Skype does.
Not only could video calls become native to the dialler, but other Rich Communication
Services (or RCS’s) could as well, such as file transferring, real time language
translation and video voicemail and there may be applications which haven’t even been
thought up yet.
As VoLTE is tied to data it could also mean that you won’t have to worry about how
many minutes you use, as everything will fall under data use.
Are there any limitations of VoLTE?
Initially there are a few. For one thing in some implementations it only works if both the
device making a call and the one receiving it support VoLTE, so in the early days you
might find yourself quite limited in terms of who you can actually use it to contact.
Depending on how the networks set it up there also may or may not be network
interoperability at first, so it’s possible that initially you may only be able to use VoLTE to
call people on the same network as you.
VoLTE also potentially requires both participants on the call to have 4G coverage. As
that’s not yet as widespread as 2G and 3G it means that VoLTE calls won’t always be
available and if someone moves out of 4G coverage during the call there’s a chance
that the call will be dropped.
Finally, pricing may be an issue. Since VoLTE ties into data it may be the case that
calling becomes more expensive than it currently is, depending on what networks
decide to charge. All of these problems are likely to be short-lived though as 4G
coverage increases, more devices begin to support VoLTE, prices stabilise and
networks align their technology. In fact Three's version of the service (which is the only
live one in the UK) suffers from very few of these problems, with the only major ones
being lack of VoLTE coverage and limited support for existing handsets.
Why haven't we been making calls over 4G with
VoLTE all along?
The problem with Voice over LTE is that 4G LTE is a data-only networking technology,
so it doesn’t natively support voice calls. While 3G and 2G were primarily designed with
voice calls in mind and data was added to them.
As such it’s been necessary to create new protocols to support voice calling over 4G
and it’s a big job, requiring upgrades across the entire voice call infrastructure. There’s
no one standard for this, with different networks creating their own solutions.
There are some common problems and solutions though, most notably the requirement
for Single Radio Voice Call Continuity (SRVCC), which simply means that the phone will
be able to switch back to a 2G or 3G signal if you move out of a 4G signal zone during
the call.
This needs to be seamless or the call will cut out, so it requires masts to pre-emptively
deliver a 3G signal whenever the 4G signal drops dangerously low, but also to keep the
4G signal running at the same time, so it can either stay on 3G if it loses the 4G signal
entirely or drop the 3G if the 4G signal becomes strengthened again.
Who offers VoLTE in the UK and how do I get it?
Three is the first and currently only network to offer VoLTE, through a service dubbed
4G Super-Voice. Currently its Super-Voice service is available to over 50% of the UK
population indoors and by 2017 the network plans to bring it to 5.5 million
customers. The selection of handsets you can get it on is limited but growing, but
includes the iPhone 6, LG G4, Galaxy S7 and many more.
Check Three 4G Super-Voice supported handsets and coverage here
EE and Vodafone are likely to be next in line for a VoLTE launch. The two networks
both predicted a summer 2015 launch and while they’re both running late we may well
still see their 4G calls services before too long. Especially as Vodafone is now trialling
VoLTE on a live network and EE has moved to nationwide trials. Don’t expect to see
VoLTE from O2 quite so soon, but the network is working on it, with trials completed late
last year and a commercial rollout planned for sometimes in 2016. For a more in depth
look at when and where VoLTE is available you can head over to our dedicated guide.
How do I know if my phone supports VoLTE?
If you’re using a 3G handset then you’ll definitely need to upgrade, but if you have a
newer 4G enabled one then it might work with VoLTE (though it will require a software
update first).
So far in the UK only Three offers VoLTE and it’s limited to a handful of flagship
handsets, but it’s working to push out software updates for a range of other phones.
See the full list of supported handsets here
In theory most existing 4G phones should work with VoLTE, but some may encounter
problems and as they all require a tailored software update it’s dependent on each
network to roll one out, so if you don’t have a new and popular phone then you may be
left waiting indefinitely.

Voice over LTE - VoLTE Tutorial

- operation of Voice over LTE VoLTE system for providing a unified
format of voice traffic on LTE, and other systems including CSFB, and
SV-LTE.
IN THIS SECTION

The Voice over LTE, VoLTE scheme was devised as a result of operators seeking a standardised
system for transferring traffic for voice over LTE.
Originally LTE was seen as a completely IP cellular system just for carrying data, and operators
would be able to carry voice either by reverting to 2G / 3G systems or by using VoIP in one form or
another.

From around 2014 Phones like this iPhone6 incorporated VoLTE as standard
However it was seen that this would lead to fragmentation and incompatibility not allowing all phones
to communicate with each other and this would reduce voice traffic. Additionally SMS services are
still widely used, often proving a means of set-up for other applications.
Even though revenue from voice calls and SMS is falling, a format for voice over LTE and
messaging, it was as necessary to have a viable and standardized scheme to provide the voice and
SMS services to protect this revenue.
Options for LTE Voice
When looking at the options for ways of carrying voice over the LTE system, a number of possible
solutions were investigated. A number of alliances were set up to promote different ways of
providing the service. A number of systems were prosed as outlined below:

 CSFB, Circuit Switched Fall Back: The circuit switched fall-back, CSFB option for
providing voice over LTE has been standardised under 3GPP specification 23.272.
Essentially LTE CSFB uses a variety of processes and network elements to enable the
circuit to fall back to the 2G or 3G connection (GSM, UMTS, CDMA2000 1x) before a circuit
switched call is initiated.

The specification also allows for SMS to be carried as this is essential for very many set-up
procedures for cellular telecommunications. To achieve this the handset uses an interface
known as SGs which allows messages to be sent over an LTE channel.
 SV-LTE - Simultaneous Voice LTE: SV-LTE allows packet switched LTE services to run
simultaneously with a circuit switched voice service. SV-LTE facility provides the facilities of
CSFB at the same time as running a packet switched data service. It has the disadvantage
that it requires two radios to run at the same time within the handset which has a serious
impact on battery life which is already a major issue.
 VoLGA, Voice over LTE via GAN: The VoLGA standard was based on the existing 3GPP
Generic Access Network (GAN) standard, and the aim was to enable LTE users to receive a
consistent set of voice, SMS (and other circuit-switched) services as they transition between
GSM, UMTS and LTE access networks. For mobile operators, the aim of VoLGA was to
provide a low-cost and low-risk approach for bringing their primary revenue generating
services (voice and SMS) onto the new LTE network deployments.
 One Voice / later called Voice over LTE, VoLTE: The Voice over LTE, VoLTE scheme for
providing voice over an LTE system utilises IMS enabling it to become part of a rich media
solution. It was the option chosen by the GSMA for use on LTE and is the standardised
method for providing SMS and voice over LTE.

Voice over LTE, VoLTE formation

Originally the concept for an SMS and voice system over LTE using IMS had been opposed by
many operators because of the complexity of IMS. They had seen it as far too expensive and
burdensome to introduce and maintain.
However, the One Voice profile for Voice over LTE was developed by a collaboration between over
forty operators including: AT&T, Verizon Wireless, Nokia and Alcatel-Lucent.
At the 2010 GSMA Mobile World Congress, GSMA announced that they were supporting the One
Voice solution to provide Voice over LTE.
To achieve a workable system, a cut down variant of IMS was used. It was felt that his would be
acceptable to operators while still providing the functionality required.
The VoLTE system is based on the IMS MMTel concepts that were previously in existence. It has
been specified in the GSMA profile IR 92.
Voice over LTE, VoLTE basics
VoLTE, Voice over LTE is an IMS-based specification. Adopting this approach, it enables the system
to be integrated with the suite of applications that will become available on LTE.

Note on IMS:
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem, IMS is an architectural framework for
delivering Internet Protocol, IP multimedia services. It enables a variety of services to be run seemlessly rather than
having several disparate applications operating concurrently.
Click for an IMS tutorial
In order that IMS was implemented in fashion that would be acceptable to operators, a cut down
version was defined. This not only reduced the number of entities required in the IMS network, but it
also simplified the interconnectivity - focussing on the elements required for VoLTE.

Reduced IMS network for VoLTE

As can be seen there are several entities within the reduced IMS network used for VoLTE:

 IP-CAN IP, Connectivity Access Network: This consists of the EUTRAN and the MME.
 P-CSCF, Proxy Call State Control Function: The P-CSCF is the user to network proxy. In
this respect all SIP signalling to and from the user runs via the P-CSCF whether in the home
or a visited network.
 I-CSCF, Interrogating Call State Control Function: The I-CSCF is used for forwarding an
initial SIP request to the S-CSCF. When the initiator does not know which S-CSCF should
receive the request.
 S-CSCF, Serving Call State Control Function: The S-CSCF undertakes a variety of
actions within the overall system, and it has a number of interfaces to enable it to
communicate with other entities within the overall system.
 AS, Application Server: It is the application server that handles the voice as an
application.
 HSS, Home Subscriber Server: The IMS HSS or home subscriber server is the main
subscriber database used within IMS. The IMS HSS provides details of the subscribers to
the other entities within the IMS network, enabling users to be granted access or not
dependent upon their status.

The IMS calls for VoLTE are processed by the subscriber's S-CSCF in the home network. The
connection to the S-CSCF is via the P-CSCF. Dependent upon the network in use and overall
location within a network, the P-CSCF will vary, and a key element in the enablement of voice calling
capability is the discovery of the P-CSCF.
An additional requirement for VoLTE enabled networks is to have a means to handing back to circuit
switched legacy networks in a seamless manner, while only having one transmitting radio in the
handset to preserve battery life. A system known as SRVCC - Single Radio Voice Call Continuity is
required for this. Read more about .
VoLTE codecs
As with any digital voice system, a codec must be used. The VoLTE codec is that specified by 3GPP
and is the adaptive multi-rate, AMR codec that is used in many other cellular systems from GSM
through UMTS and now to LTE. The AMR-wideband codec may also be used.
The used of the AMR codec for VoLTE also provides advantages in terms of interoperability with
legacy systems. No transcoders are needed as most legacy systems now are moving towards the
AMR codec.
In addition to this, support for dual tone multi-frequency, DTMF signalling is also mandatory as this is
widely used for many forms of signalling over analogue telephone lines.
VoLTE IP versions
With the update from IPv4 to IPv6, the version of IP used in any system is of importance.
VoLTE devices are required to operate in dual stack mode catering for both IPv4 and IPv6.
If the IMS application profile assigns and IPv6 address, then the device is required to prefer that
address and also to specifically use it during the P-CSCF discovery phase.
One of the issues with voice over IP type calls is the overhead resulting from the IP header. To
overcome this issue VoLTE requires that IP header compression is used along with RoHC, Robust
Header Compression, protocol for voice data packet headers.