Transcript 09_Protocols_ telephone_networks

Communication
Systems
9th lecture
Chair of Communication Systems
Department of Applied Sciences
University of Freiburg
2008
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Communication Systems
administrative stuff (lectures and exercises)
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Please hand in the theoretical exercise sheets #4
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And grab the new sheet (#5) due for the 24th of June
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16th and 20th are lectures (all held here in this lecture room), next
practical course is 24th (again at the computer center, exact start
time will be announced the lecture before)
You might refer to the
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electures.informatik.uni-freiburg.de for the lecture recordings and
slides in different formats

www.ks.uni-freiburg.de for details on upcoming lectures and material
like the slides of the practical course (containing the “master”
solutions to the exercise questions and some hints for the actual
exercise sheets, the requirements on this lecture available from the
first practical courses slide set)
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Communication Systems
last lectures
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Lecture started with rather modern communication technologies
and introduced the Internet Protocol as a global orientated packet
switching network technology
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IP can be run over very different physical media and intermediate
protocols
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and IP is used for more and more networked services
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future IPv6 will solve any address scarcities
Very popular traditional service is telephony mostly 1:1 voice
communication
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with the more and more widely introduced “Voice-over-IP” we could
observe a merge of both networks
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and we will see a role change in the sphere of network providers
(telephony companies vs. IP services providers, (TV) cable
networks)
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Communication Systems
Upcoming lectures
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To get an idea how traditional and modern wireless telephony
networks work, we give an introduction to ISDN, GSM and UMTS
First traditional telephony networks its history and their concepts
in general
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Components of (digital) telephony networks
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Digitization of voice - PCM
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Then introduction to ISDN – a completely digitalized
communication infrastructure
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call setup and global routing in telephony networks
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network definitions and standards
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Communication Systems
plan for this lecture
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History of telephony networks and wireless information networks
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Line switching
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DTMF – dual tone multi frequency as traditional signalling
source
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Telephony protocol
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Standards in telecommunication
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Digital telephony networks – from analogous source to digitized
data streams - PCM
ISDN – Integrated Services Digital Network
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D channel protocol
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DSS1 layer 3 protocol
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Communication Systems
History of telephony networks
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Traditional analogous telephony networks
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1848: State Telegraphy System in Prussia (Siemens)
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1851: First trans-sea cable between Dover and Calais
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1858: Transatlantic line-based telegraphy between Europe and
America
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1866: Durable transatlantic cable
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1876: Bell patents the “phone” (Reiss in Germany – you will find
even more inventors of the telephone, for each nation one :))
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1880: 50.000 participants in US phone network
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1881: Berlin opens the first “Fernsprechamt”
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Communication Systems
History of wireless information networks
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Wireless signal transmission
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Morse codes transmitted by radio (Marconi)
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1901: Radio-based telegraphy between Europe and the US
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1914: Introducing the teletype/telex system
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1915: Wireless telephony NY – San Francisco
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1920: First public radio transmission in Königs-Wusterhausen
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1923: Start of entertaining radio in Berlin
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1929: First radio-based TV transmission (Funkausstellung in Berlin)
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1935: First regular public TV transmissions in Berlin
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Communication Systems
development of telephony equipment
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Traditional analogous telephony networks provides most of the
standards (partly) in use up to now
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bi-directional voice channel
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bandwidth to carry voice around 300Hz - 3,4kHz – just the
characteristics of the end user devices and their microphones and
earpieces
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you could hook up the old mid-thirties or sixties telephone set to your
wall socket of your telephony provider or your private
telephone
installation
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end devices are power supplied by the telephone exchange, so the
devices independent of local (power) sources (which have some
advantages in cases of power failures)
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Communication Systems
development of telephony equipment
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Local loop – connection of the end uses device to the telephony
exchange
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Device is without power when hook on cradle
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Call information is signalled with 65V alternating current
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When off-hook power supplied at around 60V by a current of 20 –
40mA
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Dial plate cuts the local loop for well defined periods to indicate dial
information (~60ms cut, ~40ms closed in between – try to dial via
cradle – system is rather robust in detection :-))
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Communication Systems
line switching
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End systems has to be connected somehow to each other
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In the early beginnings manual switch boards (you know the pictures
of old films with young ladies called operators plugging wires to
connect subscribers :-))
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around 1975
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Communication Systems
line switching
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Switch boards
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first direct-dial switch boards
appeared around 1900 used in
local area nets first and from
around 1920 for long distance calls
– dial plates (digits 1 – 9, 0) where
added to the telephone device
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using special relay boards with
contacts for each dialed digit
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system operated directly controlled
until around 1960s
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there is an impressive collection of
old (mechanical) switchboards in
the Nuremberg Communication
Museum
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Communication Systems
line switching and signalling
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Early phones used a hand generator to signal assistance by the
operator at the switch board
Now: Identification of each end device through numerical ID
composed of digits from decade system
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dial plates (digits 1 – 9, 0) where added to the telephone device
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automated line switching
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Switch boards - routers in the telephony world
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major drawbacks of this concept
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route of the call is fixed
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every dialed digit switches the next relay in the switching network
the (long distance) line was already occupied during call setup
(=expensive for the telephone companies, especially if no-one
answered the call)
Next step was introduction of indirectly operated switching
networks middle of the fifties
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before routing setup the dial information was collected and then
processed
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Communication Systems
line switching
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Analogous electronic switching networks appeared with the
beginning of the 1970s
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allowed new type of dial indication
DTMF – dual tone multi frequency was introduced for dial
information
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inband signalling
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pulse dial information has to be transported via copper wires and
require rather high currents
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puls dialing impossible over very long distances (resistor capacity of
wire) and wireless transmission
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major speedup for dialing (same amount of time spent for each digit
transmitted)
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DTMF
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voice frequency band to the call switching center – frequencies
selected in a way that no clash with “normal” voice
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multifrequency shift keying (MFSK)
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DTMF signalling
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still in use on analogous lines and for signaling e.g. on voice menu
systems – digital equipment uses out-of-band
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special codes for signaling other data (e.g. Pay card identification)
and for cost signaling between switching centers
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some people were able to produce the needed frequencies to switch
off payment or setup special connections (no cost, used by Telcos
for maintenance)
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“hacking/cracking” started not with computer networks but with
automated telephony equipment – challenge of the 1970s was to
setup connections/routes around the globe to call someone other in
the same city (and enjoy the delay because of the huge distances)
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telephony protocol
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Key dials were introduced to telephones – special optimized
layout (in contradiction to keyboard, counters layout used today)
So we have a well known “protocol of analogous telephony
connection”
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Protocol of analogous telephony connection
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standards in telecommunication
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But in telephony world mostly not talked on “protocols” but
interfaces
Interfaces are well-defined connection points where different parts
of the infrastructure/equipment talk to each other in a certain way
International standardization body is ITU (International
Telecommunication Union www.itu.int)
Process of standardization completely different to the workflows in
Internet bodies
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no bottom up, but top down decisions
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exclusive club of the big (state monopoly) Telcos
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high annual fees
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much less information publically available then for IP and other open
protocols
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standards in telecommunication
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Because of the old (nation state) monopolies there are many
differences within the several networks
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Numbering schemes
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Acoustical indication of dial states (busy, line-free, ...)
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Different use, assignment of the (wireless) frequency
spectrum
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Not really compatible equipment (branch exchanges, ...) every firm tries to use their own subset of “standards”
With the introduction of digital networks (ISDN and mobile)
agreement on global standards started
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standards in telecommunication
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Inter connecting of voice streams has lots of technical problems
Up to 1980s computerized switching centers but analogous voice
connections
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fault-prone to jamming and noise
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regeneration means amplification of noise too
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Allow data connections over telephony networks
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Next step: Fully computerized switching centers
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out of band signaling of call setup
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digital voice streams allow better/perfect regeneration
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digital telephony networks - PCM
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Voice/speech analogous signal
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continuous in time and value
domain
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characterized by amplitude
(signal strength) and frequency
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bandwidth in traditional telephony
networks 300Hz - 3,4kHz
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digital telephony networks - PCM
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Sampling of a signal
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rate at least twice the max
frequency of analogous signal
(Nyquest theorem)
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2* fmaxb = 2*3,4kHz = 6,8kHz
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internationally the sample
frequency was agreed on
fSample=8kHz=8000Hz=8000/s
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we get a sample period of
T=1/f=1/8000=125µs
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digital telephony networks - PCM
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Analogous signal
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Continuous in value domain
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Has to be translated into discrete
values
A/D converter quantizes the
signal
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Splitting the value domain into
equal intervals
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Every measured value is
approximated and assigned to
one of the defined intervals
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digital telephony networks - PCM
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PCM defines 128 different
levels for positive and 128
negative amplitude of the
signal
thus resolution is 256 bit
Sample rate is 8000 per
second
so we get 8000 Byte per
second and a bit stream of
64kbit/s
So we have the B channel
bandwidth for ISDN ...
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ISDN – Integrated Services Digital Network
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The development of digital switching networks led to
standardization and integration of additional services into the
same network
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three virtual multiplex channels over the same two wire infrastructure
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digital telephony (two independent lines on basic rate interface)
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fax, telex
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video telephony (H.323 devices may use ISDN as transport layer for
their applications)
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data communication of 64 or 128kbit/s
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ISDN – Integrated Services Digital Network
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Prerequisite for ISDN was digitalized infrastructure
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The ISDN standard was defined in the early 1980s by the ITU
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several national standards evolved, 1TR6 in Germany, NI-1/2
in United States, DACS in UK, ...
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DSS1 is the “EURO-ISDN” used in many other countries too
available from 1993
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EURO ISDN was defined by the new founded ETSI (European
Telecommunication Standards Institute in 1988)
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ISDN – Integrated Services Digital Network
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ISDN is commonly used in all European countries since 2000
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all switching centers use ISDN backends
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so called “analogous” telephony devices (POTS – plain old
telephony service) are converted to digital service at the local
switching center
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50% of the European BRI connections are in Germany
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Germany has a 30% worldwide share
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ISDN – and the OSI protocol stack (mostly D channel)
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ISDN – Basic Rate Interface
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BRI provides a total data rate of 160kbit/s
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standard end user connection
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2 B channels (“bearer” - for data, digitized voice, ...) of 64kbit/s each
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1 D channel (data channel for out-of-band signaling) of 16kbit/s
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synchronization of 16kbit/s
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ISDN – Basic Rate Interface
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Physical layer specifications of the Uk0
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operates over two-wire cable up to 5 km (depending on cable
diameter and quality)
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switching center provides a 90V current to power the NTBA and one
device (emergency function – to be independent on local power
supply for at least one telephone)
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other physical layer specifications for alternate U interfaces
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Communication Systems
ISDN – Basic Rate Interface
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BRI network termination is defined by the Uk0 interface
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a special encoding (4B3T) is used: 4 bit digital to 3 baud ternary
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4B3T is a "block code" that uses Return-to-Zero states
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allows reduction of symbol rate to 120 kBaud (¾th) and thus
distances up to 8km
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reduction of low frequencies in the signal spectrum
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better detection of code errors
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three states: negative pulse, no pulse, positive pulse
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ISDN – Basic Rate Interface
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Next state (S1 - S4) to be transmitted is indicated in column
labeled Go
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ISDN – Basic Rate Interface
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Alternate encoding: 2B1Q – 2 bit digital to 1 baud quaternary
representation
2B1Q transmission can be simply described as an amplitude
modulation scheme for DC pulses
Ordering of data blocks depends on the encoding used
Bits
00
01
10
11
Voltage
-2.50
-0.83
2.50
0.83
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Uk0 – bit streams from switching center to NTBA
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Each frame consists of 120 ternary steps
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2*B+1*D takes 108 steps in 4 ternary blocks (tb) with 27 steps each
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sync channel occupies 11 steps and a “maintenance” channel (mc) 1
step
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Uk0 – bit streams from NTBA to switching center
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Connection is full-duplex over the two wires
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echo compensation and terminating set is needed
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NTBA splits the data streams to separate up and down onto the S0
bus
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ISDN – Basic Rate Interface
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Instead of the traditional wall socket a NTBA (network terminal
base adapter) is needed at end users site
NTBA provides the S0 bus to which end user devices are
connected
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unidirectional – on pair of wires for each direction
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allows up to 12 wall sockets, 8 ISDN devices (or analogous devices
via a/b converter)
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provides device power up to 4,5W
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ISDN – S0
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Provides the same B and D channels as Uk0
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maintains the step and octet frequency
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handles the device plugging and device activation, deactivation
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has to be terminates with resistors of 110 Ohm
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uses modified AMI code with currents of -0,75 and 0,75V
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S0 – AMI code
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Modified AMI code (avoid long sequences of symbols of the same
type)
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data link layer for the D channel
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No distinct layering for B channels – PCM or data directly put into
frames as shown on previous slides
LAPD – Link Access Procedure on D channel
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derived from High-Level Data Link Control Protokoll (HDLC)
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broadcasts only for network termination device
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D2 frame margin – octet of binary pattern: 01111110
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Keeping of frame sequence
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Error discovery
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Multiplexing of more than one logical D2 connections
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Flow control
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higher layer protocols for the D channel
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ITU Recommendation Q.921
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layer 2 for the D channel
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Flag
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character is part of the Header information, hexadecimal 7E
Address is two bytes (octets) long, and consists of three fields
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Service Access Point Identifier (SAPI)
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Command/Response (C/R) bit
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Terminal Endpoint Identifier (TEI)
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layer 2 for the D channel
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Control one or two octets (bytes) in length, indicates one of three
frame formats
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information
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supervisory
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unnumbered
Information carries Layer 3 Call Control (Q.931) data

it may carry Unnumbered Information data (TEI assignment) or XID
(Connection Management/parameter negotiation) information
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data link layer for the D channel
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Protocol handles the TEI (Terminal Endpoint Identifier) allocation
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all devices on S0 using the same bus and have to be addressable
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TEI assignment is started by the connected devices after successful
initialization of physical layer synchronization
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non automatic assignment uses ID0 – 63, automatic 64 – 126

there is a special group TEI 127
Protocol elements
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information lowermost bit is set to 0
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data link layer for the D channel
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Protocol elements
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Receive Ready - (01)
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Set Asyncronous Balance Mode Extended - (6F/7F)
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Unnumbered Information - (03)
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Disconnect - (43/53)
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Unnumbered Acknowledgement – (63/73)
Flow control uses sequence numbers for sending and receiving
00:E1:04:00:...

Octets #4 for sending and #5 for receiving in the information
frame
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data link layer for the D channel error detection
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D channel protocol uses rather sophisticated error detection
protocol
Generates frame checksums
Generator polynom
g(x) = (x +1)(x15+x14+x13+x12+x4+x2+x +1)
g(x) = x16+x12+x5+1

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16 bit frame checksum
Inverted residue of binary division
p1(x) = xk (x15+x14+...+x2+x +1)
p2(x) = x16d(x)
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data link layer for the D channel error detection
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Checking for added or lost binary zeros
Thus cyclic Hamming codes implemented
Error detection for one, two and three bit error
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network layer for the D channel
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DSS1 protocol handels the call setup of the calling and
called site
Call destruction after finishing the session
Restaring and parking if required
Error handling
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DSS1 layer 3 protocol
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Protocol Discriminator
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part of the Layer 3 header information
single byte (octet) that is usually set to a value of 00001000
(hexadecimal "08") - meaning Q.931 call maintenance
Reference Value consists of either two or three bytes (octets)

BRI systems have a 7-bit Call Reference value (127 references)
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no particular end-to-end significance
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Either end can assign an arbitrary value
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used to associate messages with a particulary channel connection
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DSS1 layer 3 protocol
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Message Type single byte (octet) that indicates what type of
message is being sent/received
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DSS1 layer 3 protocol – message types
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Message Type – four categories
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Call Establishment
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Call Information
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Call Clearing
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Miscellaneous
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DSS1 layer 3 protocol – information elements
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Each type of message has Mandatory and Optional Information
Elements, identified with single byte (octet)
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bearer Capability (identifies transport requirements of the requested
B-Channel)

cause (identifies reasons for disconnect or incomplete calls)
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channel Identification (identifies type and number of B-Channel(s)
requested)
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progress Indicator (indicates status of outgoing call)
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Communication Systems
DSS1 layer 3 protocol – information elements


Network Specific Facilities (Useful for North American PRI calls identifies network type, Carrier ID, Carrier Service Type
[WATS/SDN/ASDS,etc.])

Calling Party Number (caller ID)
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Calling Party Number sub address
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Called Party Number (destination number, type of number[unknown],
numbering plan)

Called Party Number sub address
When Information Elements (IE) consist of multiple octets, the
following octet describes how many bytes (octets) are in the
Information Element
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literature on telephony networks

E. Pehl, Digitale und analoge Datenübertragung
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See previous season lectures literature hints:
http://www.ks.uni-freiburg.de/php_termindetails.php?id=180
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Next lectures: 16th , 20th June (here), next exercise 24th
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