Compare at three levels:
- Data--continuous (audio) vs. discrete (text)
- Signaling--continuously varying electromagnetic wave vs. sequence
of voltage pulses.
- Transmission--transmit without regard to signal content vs. being
concerned with signal content. Difference in how attenuation is handled.
Look at Table 2.2 in Fig 2.13 in Stallings.
Seeing a shift towards digital transmission despite large analog base. Why?
- improving digital technology
- data integrity. Repeaters take out cumulative problems in
transmission. Can thus transmit longer distances.
- easier to multiplex large channel capacities with digital
- easy to apply encryption to digital data
- better integration if all signals are in one form. Can integrate
voice, video and digital data.
Because of the role played by phone companies in data
transmission, analog transmission has dominated communications for
quite some time. Today, everyone agrees that fiber is the way to go.
As much as we prefer fiber, analog communications will be with us for
a long time. Consider the phone system. It is characterized
by:
- Low bandwidth: It carries a bandwidth of about 3 kHz. That
is, the system only allows signals between 0-3kHz to pass through --
all higher frequencies are chopped off. The 0-3kHz spectrum covers
the most important frequencies of human voice, which is precisely what
the phone system has been designed to carry.
- High error rate: Relative to LANs, the error rate is roughly 11
orders of magnitude higher! Errors don't matter as much to analog
communication, especially when voice is involved. For digital
communications, of course, a 1-bit error can have devastating
consequences.
The phone system is organized into a hierarchy:
Must convert digital data to analog signal. One such device is a
modem to translate between bit-serial and modulated carrier signals.
To send digital data using analog technology, the sender generates a
carrier signal at some continuous tone (e.g. 1-2 kHz in
phone circuits) that looks like a sine wave. The following techniques
are used to encode digital data into analog signals (Fig 2-18)
Resulting bandwidth is centered on the carrier frequency.
- amplitude-shift modulation (keying): vary the amplitude (e.g.
voltage) of the signal. Used to transmit digital data over optical fiber.
- frequency-shift modulation: two (or more
tones) are used, which are near the carrier frequency. Used in a
full-duplex modem (signals in both directions).
- phase-shift modulation: systematically
shift the carrier wave at uniformly spaced intervals.
For instance, the wave could be shifted by 45, 135, 225, 315 degree at
each timing mark. In this case, each timing interval carries 2 bits
of information.
Why not shift by 0, 90, 180, 270? Shifting zero degrees means no
shift, and an extended set of no shifts leads to clock synchronization
difficulties.
Another variation, called Quadrature Amplitude Modulation
(Quadrature Phase-Shift Keying), has the following characteristics:
Phone communication gives rise to further complications. Under some
conditions, part of a signal may reflect back to the sender. In the
phone system, the result is an annoying echo of the speakers voice.
Echo suppressors are used remedy the echo problem. They
monitor the line, and when one side is transmitting, the suppressor
disables transmission in the reverse direction. However, now only one
person can talk at a time.
For modems, lack of full-duplex communication is can be a significant
problem. To get around this problem, most phone lines will disable echo
suppression when a 2100 Hz signal is sent on the line. This is a form of
in-band signaling because the data channel itself is used to
carry information.
Also out-of-band signalling--signal doesn't interfer with normal
transmission.
Touch tone dialing and pulse dialing are other examples of in-band
signaling.
Thought for the future: Will fiber replace the 2-wire local loop, or
are there cheaper alternatives? Cable systems may well jump in
and provide a high-bandwidth alternative to the phone companies.
Can actually transmit analog data in a similar manner with amplitude-, phase-
and frequency-modulated waves. Stallings Fig 4.20.
Two reasons:
- Transmission media may need to use a higher frequency than that used
by the data (such as voice)
- Modulation permits frequency-division multiplexing.
The EIA RS-232-C standard provides an example physical layer protocol,
describing the pins, signals, and protocols for the interaction between
a computer and modem.
The standard views the communication world from the two perspectives:
- computers (or terminals) called Data Terminal
Equipment (DTE)
- modems called Data Circuit Terminating
Equipment (DCE)
The standard specifies a 25-pin DB-25 connector including:
- mechanical specification: 25 pins, including each pins
thickness, length, etc.
- electrical specification: a binary ``1'' indicated by voltage
less than -3, and a binary ``0'' indicated by a voltage of greater
than +4
- functional specification: what each of the pins means
Digital transmission has several advantages over analog transmission:
- Analog circuits require amplifiers, and each amplifier adds
distortion and noise to the signal.
- In contrast, digital amplifiers regenerate an exact
signal, eliminating cumulative errors.An incoming (analog)
signal is sampled, its value is determined, and the node then
generates a new signal from the bit value; the incoming signal is
discarded. With analog circuits, intermediate nodes amplify the
incomiing signal, noise and all.
- Voice, data, video, etc. can all by carried by digital circuits.
What about carrying digital signals over analog circuit?
The modem example shows the difficulties in carrying digital over
analog.
A simple encoding method is to use constant voltage levels for a ``1'' and
a ``0''. Can lead to long periods where the voltage does not change.
With digital transmission, one problem that continually arises is
clock synchronization. The receiver must be able to
determine when one bit time ends and the next one starts, so that it
samples one pulse, rather than part of one pulse and part of the
next.
Note: quartz clocks are not accurate enough. Eventually, the sender
and receiver's clock will drift apart.
Possibilities:
- include timing information in the data signal
- use a separate channel (e.g., wire) to transmit timing information
Manchester encoding is one technique that provides
clocking information. The encoding splits each sampling unit into 2
halves where:
- a binary ``1'' is sent as a high-low voltage sequence
- a ``0'' is sent as a low-high sequence
- because each sampling time contains one transition, the
receiver can easily synchronize its clock to the sender's.
In a related technique, differential Manchester
encoding, a ``1'' bit is indicated by the absence of a transition at
the start of the bit time, while a ``0'' is indicated by the presence
of a transition.
Drawback of Manchester encoding:
- half the bandwidth is wasted because it takes two transitions to
represent one bit
Advantages:
- reduced complexity of transmitter and receiver components
See Fig 4-20.
Although most local loops are analog, end offices increasingly use
digital circuits for inter-trunk lines. A codec
(coder/decoder) is a device that converts an analog signal into a
digital signal.
To convert analog signals to digital signals, many
systems use Pulse Code Modulation (PCM):
- PCM samples the 4kHz signal 8,000 times per second. Why?
PCM takes advantage of Nyquist's result, sampling the 4kHz
bandwidth signal at 2H = 8 thousand times per second. (Here we
assume the use of a standard voice grade line.)
- Each sample measures the amplitude of the signal, converting it
into an n-digit integer value.
- The digital channel carries these n-digit encodings.
One popular product is Bell's T1 carrier (Figure 2-26)
- It multiplexes 24 voice channels over one digital channel.
That is, it carries 24 voice channels at the same time
over one digital channel.
- Each of the 24 analog inputs is sampled in round-robin fashion
and its n-bit encoding is sent down the wire.
- Each encoding consists of 7 bits of sampled data, plus 1 bit of
signaling information (e.g., out-of-band information).
- Each voice grade sub-channel carries (7 bits X 8000 samples) =
56kbps of data, plus 8000 bps of signaling information,
requiring a digital data rate of 64kbps.
- Samples are transmitted in 193-bit units (frames).
- Each 193-bit frame consists of bits of information; the
extra bit of information carries synchronization
information, which is similar in purpose to the start bit in
RS-232-C. It alternates between a ``0'' and ``1'' allowing the
receiver to verify that it is properly recognizing the start and end
of frames.
- A T1 channel has an aggregate carrying capacity of 1.544 Mbps.
As for the international standard, CCITT felt that 8 kbps signaling
was overkill, so their standard encodes digital signals differently:
- In Common Channel Signaling, all 8 bits carry
data, and the extra frame bit is used to carry framing and signaling
information
- Channel Associated Signaling is yet another
variation on the same idea. Here, five of six samples carries 8
bits of data, while every sixth sample carries seven bit of data and
one for signalling.
Note: Other Bell standards specify how T1 trunks are to be multiplexed
over higher capacity trunks, such as: T2 (6.3 Mbps), T3 (44.7 Mbps)
and T4 (274.2 Mbps).
It turns out that 8 bits of data can be reduced through
compression. For compression, the assumption is that the
signal changes relatively slowly compared to the sampling frequency:
- In differential pulse code modulation, each
sample contains the (signed) difference between the current and
previous amplitude value.
It only requires 5 bits, and works well in practice with voice
traffic.
- Delta modulation assumes that each sample differs by
either +/- 1 relative to the previous sample, requiring only
a single bit to represent each sample. See Figure 2-27.
- Predictive encoding attempts to predict what the next sample
will look like, transmitting the difference between the actual
measured sample and the expected sample.
Finally, X.21 is a CCITT standard that specifies the interaction between a
digital DTE (customer's computer) and DCE (carrier equipment). X.21
is the physical layer specification of the X.25 protocols. It includes
specifications for dialing calls, returning the status of a call, hanging
up calls, etc.
Problem: Given a channel of large capacity, how does one subdivide the
channel into smaller logical channels for individual users?
Multiplex many conversations over same channel. Three
flavors of solution:
- Frequency division multiplexing (FDM):
- Divide
the frequency spectrum into smaller subchannels, giving each
user exclusive use of a subchannel (e.g., radio and TV). One
problem with FDM is that a user is given all of the frequency to use,
and if the user has no data to send, bandwidth is wasted -- it cannot
be used by another user.
- Time division multiplexing (TDM):
- Use time
slicing to give each user the full bandwidth, but for only a fraction
of a second at a time (analogous to time sharing in operating
systems). Again, if the user doesn't have data to sent during his
timeslice, the bandwidth is not used (e.g., wasted).
- Statistical multiplexing:
- Allocate bandwidth to arriving
packets on demand. This leads to the most efficient use of
channel bandwidth because it only carries useful data. That is,
channel bandwidth is allocated to packets that are waiting for
transmission, and a user generating no packets doesn't use any of the
channel resources.
Now we see another reason why the phone system limits the
bandwidth voice grade lines to 3kHz. FDM is used on the trunk
lines, allocating 4 kHz to each channel.
Only 3 kHz is consistently usable, with 500 Hz of guard
bandwidth on each end of the spectrum.
One common organization of channels is as follows:
- Bundle 12 voice grade lines into a unit called a group.
A group carries signals in the 60-108 kHz spectrum.
- Combine 5 groups into supergroup.
- Combine 5 supergroups into a mastergroup.
Both TDM and FDM work well with continuous transmission,
in which data is generated at a constant rate (e.g. voice).
How well does it work for computer traffic?
Not so well. Computer traffic is extremely
bursty, characterized by alternating periods of idleness and
heavy data transmission.
The phone system uses a technique called circuit
switching (see Figure 2-35).
- Once a call has been completed, the user sees a set of
``virtual wires'' between communicating endpoints.
- The user sends a continuous stream of data, which the channel
guarantees to deliver at a known rate.
- Data transmission handled elegantly using TDM or FDM. Note
that TDM/FDM work well because the data rate is predictable -- the
voice grade signal is sampled using PCM generating a steady stream of bits.
- Call setup required before any data can be sent,
allowing network to set up the path, allocate subchannels, etc.
Call setup also used to decide who to charge for the call.
- Call termination required when parties complete
call, allowing the network to reclaim resources. At this point,
a billing record is saved somewhere that records where the call was
made, its duration, etc.
Advantages of circuit switching:
- Fixed bandwidth, guaranteed capacity (e.g., no congestion).
- Low-varience end-to-end delay (e.g., delay nearly constant).
Drawbacks:
- Connection setup introduces delay before communication can begin.
- User pays for circuit, even when not sending any data.
- Other users cannot use bandwidth of other circuits that are not
actually being used(e.g., in most conversations, only one person
speaks at a time. Thus, half the underlying bandwidth is wasted!
Entire message stored at each node. Each message is received in its
entirety before forwarding. A store-and-forward network (UUCP is a
store and forward network).
In contrast, packet switching systems use
statistical multiplexing to make better use of a channel:
- Data is sent in individual messages (packets).
- Each message is forwarded from switch to switch, eventually
reaching its destination.
- Each switch has a small amount of buffer space to temporarily
hold messages. If an outgoing line is busy, the packet is queued
until the line becomes available.
Packet switching vs circuit switching:
- (Current) packet switching system do not provide
known delay or capacity characteristics. Some applications, like
those making use of real-time voice and video, cannot tolerate high
variation in delays.
- If many sites send data at the same time, end-to-end delay
increases. That is, per-user response and throughput drops as
more users share a channel.
- Packet switching utilizes resources more efficiently (similar
to multiprocessing in operating systems). In particular, with
circuit switching, bandwidth can be allocated but unused, as when no
one talks.
- Packet switching systems doesn't usually require opening a
connection before sending data. This important for applications
that send only a single packet of data; the cost of opening and
closing a connection may exceed the cost of sending the data.
- Billing algorithm more complex in packet switching systems.
It's easy to bill for a connection, because one can figure out
who to charge during the connection set up. With packet-switching,
each packet must be accounted for individually.
Hybrid switching systems attempt to combine the advantages
of both approaches.
For instance, phone companies have developed fast
connect circuit switching systems that
establish connections quickly (e.g. on each interactive input line).
However, there is still much debate as to whether these ``fast''
systems are really fast enough.
Another variation, virtual circuits, requires users to
open a connection before sending data, but transmits packets.
The call allows the network to establish a path, and once
established, all packets follow the same path. Because all packets
follow the same path, packets can be delivered in order, and
accounting is simplified.
The goal of ISDN is to integrate voice and non-voice services over the
phone system. ISDN is an international effort, with CCITT in
charge of setting the standards. The idea is to have a ubiquitous
system (e.g. like today's phone system) that allows
computer-to-computer interaction between machines anywhere in the
world.
ISDN opens the door to many services:
- Caller identification (displaying the phone number of the person
who is calling you).
- Facsimile (fax) transmission.
- Automated meter reading (e.g. electricity, water, etc.).
- Fire and burglar alarms that notify appropriate authorities instantly.
- Electronic mail.
- Answering machines.
ISDN's goal is to integrate all services under one technology. Hurdles?
How long will it take to deploy? Probably decades.
In addition, ISDN only specifies the physical layer standards;
additional standards needed for higher layers.
The ISDN architecture provides a digital bit pipe
between a phone and the end office. The bit pipe multiplexes
different types of data (e.g. TDM) allowing transmission of voice and
data simultaneously.
An NT1 (network terminating device) is placed on the customer's
premises. The NT1 connects to the end office over the existing
two-wire circuit and has 8 (bus) cables for connecting to phones,
terminals, computers, etc.
New technology where broadband ISDN is based on ATM.
Switching on messages (cells) at 155Mbps and higher.
Packet-switched
Fig 2-43, switches hold information, rather than reserving bandwidth in a
circuit-switched network. Can have switched (temporary) or permanent
virtual circuits.
ATM maybe carried on any carrier--T1, T3, SONET, FDDI.
SONET - Synchronous Optical Network. Figure 2-32 gives SONET rates. OC-3
and OC-12 networks used on campus.
Fast switching is essential. Fixed length cells are the key to switching.
Goals:
- switch all cells with as low as discard rate as possible
- never reorder the cells in a virtual circuit
Different approaches:
- block at input to switch (Fig 2-46), can cause head-of-line blocking
- output queueing (Fig 2-47)
- knockout switch (Fig 2-48) Select up to n cells for each line on each
cycle. Extra cells are dropped. By varying n, can trade off switch cost
for expected cell rate loss.
- batcher banyan switch -- more complex