Transm ission

This document runs through some transmission basics:

An alo gue Connectivi ty

Li ne Types

This uses mainly PSTN dialup lines of varying quality and modems to provide
digital connections. Circuit switching is used to provide a link for a particular call
and levels of quality line types can be offered by the telephone companies

e.g.
1. Line Type 1 - Basic voice,
2. Line Type 5 - Basic data,
3. Line Type 7 - voice and data over private lines.

In addition, you can obtain conditioned lines with improved communications.

There is D Cond iti oning and levels 1 to 8 of C Cond it ioning . A more
expensive dedicated analogue line can be bought where the circuit is fixed and is
not different every time that you dial up, thereby giving you a more consistent
service.

Circu it S wit ch ed P ath

When making an analogue phone call you first obtain a dial tone, then you dial a
number. This number is sent to the local switch containing a D-Channel Bank
using touch tone Dual Tone Mult ifr equency (D TMF) signals. That is as far
as DTMF gets. The voice call is converted to the digital Pulse Code Modulation
format and analogue signalling to digital signalling by the D-channel bank. The
switch routes the call from this point through the digital switch network using the
Mana gement (M)-Plane protocol called Signa ll ing System Number 7
(S S7) which is a form of CCS. SS7 sends messages to the switch which is
connected to the destination phone and this far end switch sends a Cont rol
(C)-Plane message that rings the far end phone. When the phone is picked up
the C-Plane mechanisms send the message that the path is available. The
digitised voice is the User (U )-Plane data flow.

Digit al Signal ling

DS0

The Modern Telco digitizes speech using Pulse Code Mo dulat ion (PCM) on
64K (DS0) channels. 64 Kbps is considered to be Digi tal Signa l L ev el 0. Each
channel is sampled 8000 times/second according to Nyquist's Theorem, and
incorporates 8 bits per sample (hence 8bits x 8000 giving 64,000 bits/sec). This
figure of 8000 comes from the fact that the valuable range of telephone signals
is 100Hz to 4kHz, and the sampling rate is twice that of the highest signal. The
standard G.711 defines the Pulse Code Mo dulat ion (PCM) 64Kb/s voice
channel. DS0 trunks make up the trunks around the digital network that can
carry data or voice. For voice the conversion to 2-wire analogue occurs at the
switch closest to the user. The call handling in the 'external' network is dealt with
by SS7 .

Straight digital signals (bipolar) are used across these lines so no modem is
required. A Channel Servi ce U nit/Data Servi ce Unit (CS U/DSU) provides
the interface for the end user and converts the DTE's digital signals into the
Synchronous digital signals used over the WAN. Kilostream services (BT's version
of Digi tal Data Servi ces ) offer from 2.4Kb/s to 64Kb/s (56Kb/s in the USA)
whilst Kilostream N allows multiples of between 2 times and 16 times 64Kb/s to
give a range of bandwidths from 128Kb/s to 1024Kb/s. BT's Megastream offers
2, 8, 34, 45, 140, 155Mb/s bandwidths for really bandwidth intensive traffic.

T1 (DS -1)

Fr aming

For the DS-1, also called T1, Time Division Mu lti ple xing (TDM) is used to
transport multiple channels over one line. Clocking of the serial transmission
needs to occur at one end of the link or the other, sometimes you will see the
clocking options as interna l i.e. provided by the local device, or line meaning
that the clock is provided by the remote device. Two-pairs are used in a T1 link.
The T1 link can operate in full-duplex mode where one pair transmits and the
other pair receives. 24 channels are available for transmission and these are
grouped together to form a Fr ame i.e. the 24 time slots (8 bits each) plus one
framing bit form one T1 frame (193 bits, the 193rd bit being the
synchronisation/framing bit). For 8000 samples a second, a T1 frame must be
transmitted every 125 usecs, we can therefore calculate the T1 line rate as 193 x
8000 = 1.544 Mbps (A DS0 line rate is 8 bits x 8000 = 64 Kbps).

The frames can also be grouped into 12 sequenced frames to form a

Super frame (S F) (also called a D4 ) which means that 12 framing bits are used
per SF. These 12 framing bits are also called F bits. They form the sequence
10 00 11 01 11 00 and are used to sequence the SF within 4 frames. In one
second 8000 'F' bits are used for framing. This is encapsulated in the G.704
framing standard. A D4 contains 288 channels.

The frames could also be grouped into 24 to form the newer framing format
called the Exten ded Super frame (ES F) . The 8000 'F' bits are used differently
in ESF where 2000 'F' bits are used for framing, 2000 are used for CRC-6 error
checking and 4000 are used as a supervisory channel for things such as loopback
and error reporting. An ESF contains 576 channels.

Channel As soc iated Signa ll ing (CAS)

T1 signalling can take the form of CAS using Robbed Bit Signa ll ing where bits
are 'robbed' from the channels carrying the voice. This is called In-ban d
Signa ll ing . In the SF, the LSB is 'robbed' from each of the 24 x 8-bit timeslots in
the 6th and the 12th frames. The A bit comes from the 6th frame timeslots
whereas the B bit comes from the 12th frame timeslots. These 'robbed' bits are
used for call supervision and trunk signalling in the voice environment e.g. the 'A'
bit is commonly used in the same way that the 'M' lead is used in E&M signalling
i.e. signalling by pulsing the 'A' bit. This Bit Robbing is fine if the channels are
used for voice because the 8 bit samples being reduced to 7 bits every 6 frames
does not significantly impact on voice quality. Data is of course not so forgiving
with the lowered quality line so each channel is reduced to 56kbps for data (In
the US a type of ISDN called Swit ched Servi ces uses bit-robbing technology
that results in a 56kbps B-channel). The problem with using CAS is that these
robbed bits are really only used when setting up and establishing a call, the rest
of the time the bandwidth is wasted. The only messages used are Wink ,
Ringin g, Hang up and Pulse Dig it Dia ll ing .
The ESF operates a similar manner to the SF other than bits are robbed from the
18th frame (C bits ) and the 24th frame (D bits ).

The main difference between channelised lines (analogue) and non-channelised

lines (ISDN) is that they do not have a built-in D-channel. For example, all 24
channels on a T1 line only carry data. The signalling is in-band or associated to
the data channels (Channel A ssociat ed Signal lin g (C AS) ). Traditional
channelised lines do not support digitized data calls (for example, BRI with
2B+D). Channelised lines support a variety of in-band signal types, such as
ground start, loop start, wink start, immediate start, E&M and R2.

Common Chan ne l Signal ling (CCS)

T1 signalling can also take the form of CCS which is normally Common
Channel Signa ll ing Number 7 (S S7) or Primary Rate ISDN where one
channel (D-channel, channel 24) is used for Q.931 signalling. This is called Out -
of -band Signa ll ing since the signalling is in a channel that is separate from the
voice channels. This speeds up call setup by up to a factor of 5, to 1-3 seconds.
One signalling channel can handle up to 1500 calls. SS7 is a protocol in its own
right, very akin to X.25 where switches exchange billing, switching and signalling
information.

With CCS, PRI does not operate Bit Robbing but takes one of the channels and
uses that for signalling (D-channel) instead leaving 23 channels for the data. The
line encoding coding scheme used to allow both data and voice is ususally based
on a pseudo-ternary bipolar code called Bipolar wi th 8 -Zer os Substitut ion
(B 8ZS) . This is called Clear Channel . Another coding scheme called B7 exists
for voice only applications and yet another called Al ternate Mark Inversion
(AMI) is commonly used.

E1

Fr aming

The European standard E1 is another way of using TDM to transport multiple

channels over a single line. The E1 interface has 32 channels or time slots.
Framing is carried out in time slot 17, the 32 x 8-bit slots are grouped into a 256
bit Fr ame and then 16 frames are grouped into a Mul ti frame . Time slot 17
(channel 16) in the first frame of the Multiframe indicates the beginning of the
Multiframe. In the first frame there are the 4 signalling bits (ABCD) for channel 1
and 4 bits (ABCD) for channel 17. In the second frame there are 4 signalling bits
(ABCD) for channel 2 and 4 bits (ABCD) for channel 18. This carries on 16 times
to form the Multiframe. G.704 covers this framing for E1 (as well as T1).

The E1 digital line operates at 2048Kb/s and is made up of 32 x 64Kb/s. Often,

the E1 is broken up into 64Kb/s channels and is known as Fr actional E1 (or F-
T1 in the States) if the customer is presented with a group, for example a
384Kb/s circuit (made up of 6 x 64Kb/s channels). In this instance, the carrier
determines the grouping of the channels, one E1 may server a number of
different clients.

A channelised T1/E1 (CT1/CE1) line is an analogue line that was originally

intended to support analogue voice calls, but has evolved to support analogue
data calls. ISDN does not transmit across channelised T1/E1 lines.

Al ternate Mark Inversion (AMI) can be used with E1 but the most common
E1 line-encoding scheme used is called Hi gh-Densit y Bipo lar with 3- zer os
(HDB3) . The error checking used is called Cy clic Redundancy Check with
lev el 4 checking (CR C-4 ), although this can be turned off (no-CRC4) with
some providers. Australia has a different way of E1 framing from the rest of the
world.

Channel As soc iated Signa ll ing (CAS)

In CAS, Time slot 17 (E0 channel 16) is used for signalling and time slot 1 (E0
channel 0) is used for framing synchronisation and alarms, the other 30 are used
for voice and data. The CAS signalling is considered in-band and is very simple as
it just identifies four states:

• 00 - Idle
• 01 - Seizure
• 10 - Disconnect
• 11 - Busy

Common Chan ne l Signal ling (CCS)

E1 can also be set up for CCS where channel 16 carries signalling such as Q.931
for Primary Rate ISDN (I.421). This signalling, based on HDLC type protocols can
often be vendor proprietory signalling that needs to be transparently passed by
network equipment. Examples include BT's Dig ita l Priv ate Network
Signa ll ing System (DPNS S) , Nortel's Mer idian Customer Def ined
Netwo rking (MC DN) , QSIG and Signa ll ing System 7 (S S7) which is a
standard for CO to CO signalling used throughout the world.

Sig nal ling Band widt hs

One channel can carry a different data signal from another and therefore allows
multiplexing to occur to give 32 simultaneous data transmissions. The 64Kb/s
data rate is known as Digi tal Signa l L ev el 0 (DS-0) and the 1.544Mb/s rate
is known as DS-1 (T1), the following table details some of the transmission
rates:

Signal No . of T1 No . of V oice Data R ate

Carrie r
Type channels channels (Mb/s
DS-0 1 0.064
DS-1 T1 1 24 1.544
DS-1C T1-C 2 48 3.152
DS-2 T2 4 96 6.312
DS-3 T3 28 672 44.736
DS-4 T4 168 4032 274.760

Multiplexers combines several channels into a single bit stream.

Synch ronous data transmission differs from Asynchronous transmission in

that data is sent as a continuous stream of data packets separated by start and
stop bits conforming to precise clocking which is governed either by one end of a
link, or the other. Asynchronous transmission has no central precise clocking
and packets can come and go at undefined times and in different orders with
different frequencies. Each end of an Asynchronous link is responsible for its own
clocking.

For interest sake, Isynch ronous transmission is when asynchronous data is

sent over synchronous transmission. Plesio chr onous transmission is when
digital signals with different, but reliable, clock signals are being used.
Up until this point we have looked at Circui t S witched Netw orks , however,
there are also Pack et Swit ched Networks where data is split into individual
packets and each packet is switched separately through the WAN. These means
that packets can end up at the destination in a different order from when they
left. For this reason, each packet is tagged with a destination address and order
number. Often called An y-to- An y connect ions , Packet switched networks are
only switching small packets, so resends are fast and the whole switching
process is a fast one.

Virtual circuits are set up to provide either a temporary path Switche d Virtual
Ci rcu it (S VC) or a Permanent path Permanent Virtual Circui t (P VC) .

QS IG

The QSIG CCS protocol is significant because it is based on Q.931 and Q.933 and
is being developed to cover the first three layers of the OSI model with layer 3
being the messaging layer. The messaging facilities are more complex allowing
for features such as Call Forwarding, CTI and ACD to be extended across the
digital trunks between different PBX manufacturers.

Sign al ling S ys tem 7 (S S7)

SS7 is a form of CCS called CCS number 7 and has been defined by the ITU as
ITU -T#7 . SS7 uses the signalling channel on T1 (24) or E1 (17) to carry
signalling, billing and switching information between SS7 capable switches. This
is carried out-of-band in parallel to the calls.

Terminal s and Mo dems

Various types of Terminal access over the Wide Area network use different
methods to mimic being directly connected to the host:

• Telnet/ rlo gin - provides a virtual terminal connection to a computer. The

protocol rlogin is specific to Unix systems.
• Local- area T ran spo rt ( LAT) - is provided by DEC to allow terminals to
connect to a number of hosts at one time.
 • TCP/IP T elnet 3 27 0 (TN32 70) - is the virtual terminal protocol that
allows one to access 3270 applications.
• X.2 5 P ack et As semble r/Disassembler (P AD) - The PAD translates
the character-based terminal output into X.25 packets that can be
switched on the Wide Area Network. X.25 has versatility due to its
addressing schemes.

Dial-in Modems are used when there are Asy nch ronous connections to the
Wide Area Network. Protocols that run over the asynchronous connections
include Ser ial Line Pr ot oco l ( SLIP) , Point to Point Pr ot oco l (PPP) and
App leTalk Remote Access Pr oto col (ARAP) .

Traditionally, modems converted digital signals to analogue to traverse the local

loop where they are converted to digital as they are routed across the network
before being converted back into analogue for the local loop at the remote end
where the modem at the remote client converted from analogue to digital again.
The theoretical speed limit for data throughput using this scenario is 33kbps.
With the V.90 56kbps standard the the digital-to-analogue conversion still occurs
at the client end but instead of converting back to analogue at the remote end,
say at an ISP, the signal remains digital and can do so provided that the service
provider has a digital link to the telco. This means that although the data speed
in one direction from the client to the ISP is still limited to a maximum of 33kbps,
the data speed from the ISP to the client (download) can theoretically reach
56kbps. If both ends have to convert back to analogue signalling then the
maximum speed in either direction will only be 33kbps, for example modem to
modem connection between two private individuals.

The rate between a computer and a modem can be up to 57.6 kbps for a
V.32bisPlus modem or 115.2 kbps for a V.34 modem. These higher speeds are
achieved using V.42bis compression (level 5 of the Mi cr ocom Networking
Protoc ol (MNP) ). Modems commonly monitor the line quality and apply
Adap tiv e Speed Lev el lin g ( ASL) and lower the speed if the line quality
deteriorates thereby keeping the modem connection open.