Brief Overview of Evolution of Voice Network

Public Switched Telephone Network (PSTN) Have you ever imagined what happens when you pickup
your phone and start to make a call . Well your phone has a microphone attached and the purpose of
the microphone is change sound energy to an electrical signal .Analogue electrical voice signal is then
transported using the wire connected to your Telephone.Your phone is connected to the wall socket with
a RJ-11 jack. RJ-11 is the standard connector utilized on 2-pair (4-wire) telephone wiring. Pin three and
four are used for ‘Tip’ and ‘Ring’ wires of the telephone line. Tip is the ground wire (positive), and Ring is
the battery (negative) wire of a phone circuit. 2and 4 call also be used. Analogue electrical signal is
transmitted using pins 3 & 4

Now what happens when you dial a number? Each number has a unique sound or tone when you press
the Telephone numeric keypad. This sound has a name DTMF or dual tone multi frequency. DTMF are
two different tones at two ends of a spectrum that are used to send information in telephonic
communication mediums. The DTMF tones represent numbers 0-9 and the symbols * and #. seven
different frequencies are assigned. Some numbers will have the same lower frequency, but different
higher frequency. Others will have the same higher frequency but a different lower frequency. The range
is between 697 Hz and 941 Hz for the lower frequencies and 1209 Hz and 1633 Hz for the higher
frequencies.End customer or a subscriber of a Telephone line is not aware of what lies behind the
socket. To give a brief over view a simple telephone network diagram is shown below Pic-1. This
network is called the PSTN network or Public switched telephone Network.

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Everyone who subscribes to the PSTN is connected to the closest telephone office through a local loop.
Local loops connected to the public switched telephone network are two-wire metallic cable pairsEach
Telephone line has two copper wires (tip and ring) which run from a home or other small building to a
local telephone exchange. There is a central junction box MDF or Main Distribution Frame for the
building where the wires that go to telephone jacks throughout the building and wires that go to the
exchange meet. There could be 50 or maybe 100 lines in a building that run from every building unit to
the MDF. The wires between the junction box (MDF) and the exchange are known as the local loop or
ULL .The ULL or Unconditional local loop is simply the dedicated cable facility used to connect a
telephone at a subscriber's station to the closest telephone office. An exchange is a central location
where subscribers are interconnected, either temporarily or on a permanent basis. Telephone company
switching machines are located in exchanges. Switching machines are programmable matrices that
provide temporary signal paths between two subscribers. To connect one phone call to another, the
phone call is routed through numerous switches operating on a local, regional, national or international
level.See pic1 Switches reside in every exchange. The transmission link could be copper wire, optical
fibre, mobile, wireless or satellite technology. The connection established between the two phones is
called a circuit. The PSTN relies on circuit switching. So what is circuit switching? Well every time a
call connection is made a circuit or channel is established between nodes and terminals before the
users can communicate, as if the nodes were physically connected with an electrical circuit. Each circuit
cannot be used by other callers until the circuit is released and a new connection is set up. Even if no
actual communication is taking place in a dedicated circuit that channel remains unavailable to other
users
Evolution of Digital Communication Signals can be either analogue or digital. In an analogue
signal, the signal is varied continuously with respect to the information. In a digital signal, the information
is encoded as a set of discrete values (for example ones and zeros). Analogue signals degrade by noise
where as Digital signals are more imune to noise unless it exceeds a certain threshold.

The usable voice frequency band ranges from approximately 300 Hz to 3400 Hz or 3.4 KHz
The bandwidth allocated for a single voice-frequency transmission channel is usually 4 kHz, including
guard bands.

(Nyquist sampling theorem) was used to sample analog signal or take fragments of analog signal. The
speed of taking fragments or samles was twice the bandwidth for voice 2x4Khz =8Khz . This is the basis
of the pulse code modulation system used for the digital PSTN.

Pulse-code modulation (PCM) is a digital representation of an analog signal where the magnitude of
the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a numeric
(usually binary) code

In the picture below for 4Bit PCM analogue signal represented in a sin wave is sampled and quantized
for PCM. For the sine wave example, we can see that the quantized values at the sampling moments
are 7, 9, 11, 12, 13, 14, 14, 15, 15, 15, 14, etc. Encoding these values as binary numbers would result in
the following set of nibbles: 0111, 1001, 1011, 1100, 1101, 1110, 1110, 1111, 1111, 1111, 1110, etc.
These digital values could then be further processed or analyzed by a purpose-specific digital signal
processor or general purpose CPU. Several Pulse Code Modulation streams could also be multiplexed
into a larger aggregate data stream, generally for transmission of multiple streams over a single physical
link. One technique is called time-division multiplexing, or TDM,

Note for Voice we use 8 Bit PCM not 4 Bit

TDM: Time Division Multiplexing

TDM is a technology that transmits multiple signals simultaneously over a single transmission path
Multiplexing in other words is sharing of the same medium by multiple users. In TDM two or more
signals or digital bit streams are given access to the medium by physically taking turns on the channel.
There are sub channels created with in a channel and each sub channel is used to carry bits from digital
signals. Each sub-channel has a capacity to carry a certain number of bits. Basically a TDM frame is a
cyclically repeated data block that consists of a fixed number of time slots. One TDM frame consists of
one timeslot per sub-channel

Digital Signal 0 (DS0) is a basic digital signaling rate of 64 kbit/s used for transmission of voice using
TDM. The DS0 rate was introduced to carry a single digitized voice call. For a typical phone call, the
audio sound is digitized at an 8 kHz (2x4KHz) sample rate using 8-bit pulse-code modulation for each of
the 8000 samples per second. This resulted in a data rate of 64 kbit/s ( 8KHz x 8-Bit). Note Each DS0 is
8 Bit wide (64K rate per sec)

E1 lines are formed when 30 digitized phone lines are combined into a single digital data stream. The
standard frame is 32 timeslots with 30 Timeslots for voice calls. The timeslots are numbered 0 to 31.
TS0 Is used for Framing and Synchronization and TS16 Is for Signalling .Single frame is 256 (32x8)
bits wide (125 microseconds) and Each DSO is 8 bits wide (64K rate per second).
The E1 data stream is Time Division Multiplexed [TDM] by channel; 8 bits [one sample] of channel 1
followed by 8 bits of channel 2 and so on. E1 trunks are bi-direction, one twisted pair for transmit and
one twisted pair for receive.
ISDN (Integrated Services Digital Network)

ISDN – Integrated Services Digital Network – is a type of circuit switched telephone network system,
designed to allow digital (as opposed to analogue) transmission of voice and data over ordinary
telephone copper wires, resulting in better quality and higher speeds, than available with analogue
systems

ISDN is a circuit switched Telephone Network System designed to allow digital transmission of voice
and data over ordinary telephone copper wires. ISDN provides users with digital access to switched
voice and switched data on a single integrated network connection.

There are two types of ISDN services ISDN-BRI and ISDN-PRI

BRI stands for basic rate interface and PRI stands for Primary rate interface

The Basic Rate Interface consists of two 64 Kbps B-channels and one 16 Kbps D- channel. With a
maximum speed of 128 Kbps or 64 x 2 Kbps using bonding of 2 channels.

The Primary Rate consists of 30x 64 Kbps B-channels and one 64 Kbps D-channel in Europe Australia .
PRI is an ISDN service, delivered over a high capacity circuit (E-1) that provides 30-64K B channels for
voice and data traffic and a 64K D channel for signaling and call setup.
In total there are 31 Channels.

The Bearer ("B") channel is a 64 kbps channel which can be used for voice, video, data, or multimedia
calls. B-channels can be aggregated together for even higher bandwidth applications.

The Delta ("D") channel can be either a 16 kbps or 64 kbps channel used primarily for communications
(or "signaling") between switching equipment in the ISDN network and the ISDN equipment at your site

It offers circuit-switched connections (for either voice or data), and packet-switched connections (for
data), in increments of 64 kilobit/s

In ISDN PRA services trunking concept is used by which a communications system can provide
network access to many clients by sharing a set of lines or frequencies instead of providing them
individually. This is analogous to the structure of a tree with one trunk and many branches
For Example: 100 number range is allocated to 100 individual extensions and 100 extensions share 30
channels.

PHYSICAL CONNECTION ISDN

ISDN services can be terminated on the following types of physical connection:

Two types of physical delivery

Interfaces: G.703/G.704 120 Ω balanced
-Unbalance 120 W
-Copper delivery on 4 wires
-one pair for RX (1+2)
-one pair for TX (4+5)
Interfaces: G.703/G.704 75 Ω unbalanced
-Balanced 75 W
-Coax with BNC connectors
-one cable for RX
-one cable for TX
FEATURES OF ISDN SERVICES

Access Capacity ISDN PRA : 10 channel, 20 channel & 30 channel

Access Configuration: Both way, outgoing only

Standards: ETSI ISDN (TS038), TS014 ISDN

Overlap is where you dial and it sends each digits as you dial it. Enbloc stores the digits and then
sends.

LCR- Least cost routing:: In voice telecommunications, least cost routing (LCR) is the process of
selecting the path of outbound communications traffic based on cost. Within a telecoms carrier, an LCR
team might periodically (monthly, weekly or even daily) choose between routes from several or even
hundreds of carriers for destinations across the world. This function might also be automated by a
device or software program known as a "Least Cost Router."

Extension Level Billing: This feature is used when a company would like to get a detailed analysis of
the billing data at an extension level. ISDN billing is done normally under 1 single prime number. If you
would like to know what each separate extension is billing on your monthly bills you would request
the extension level billing feature.

CRC-Cyclic redundancy Check:: A CRC-enabled device calculates a short, fixed-length binary

sequence, known as the CRC code or just CRC, for each block of data and sends or stores them both
together. When a block is read or received the device repeats the calculation; if the new CRC does not
match the one calculated earlier, then the block contains a data error and the device may take
corrective action such as rereading or requesting the block be sent again, otherwise the data is
assumed to be error free. Framing format CRC-4. CCS –

Common Channel Signalling is used for ISDNs Signaling is used to indicate Status Incoming
Calls,Channel Status,Available or Out-of-Service

HDB3 - High Density Bipolar coding is used for bits 3 is used.. HDB3 coding replaces groups of 4 zeros
with either 1 or 2 marks ensuring that there is never more than 3 spaces between the marks and
therefore maintaining the clock synchronisation

Super trunk: It is a feature when activated between two or more ISDN services will distribute incoming
calls evenly between the channels of ISDN services…Example first call hits first circuit and the second
call hits the second circuit.

Cause Code based:: if it is disabled instead of sending a tone for engaged signal we will send them a
code for engaged signal

Sequential and non sequential : For sequential the first call goes to first channel and the second goes
to second channel and then so on .For non sequential it could be any random channel

In-dial, Calling Line Identification (CLI) : calls made from ISDN numbers will be displayed on
receivers handset or mobile

Number Redirection : Numbers can be diverted to customers mobile,landline , national number or

even international number
ISDN NETWORK ARCHITECTURE
VOICE SWITCHES

Voice switches are located at the POP

What is POP?

Point of Presence is the location where a long distance carrier would terminate Services and provide
connections into a local Telephone Network. It is an artificial Demarc point or interface point between
communication entities.

NORTEL DMS: Digital Multiplex system / PDTC: Digital Trunk Controller

The DMS 100 is an electronic switching system establishing a connection between two telephone lines,
or two switching systems. The purpose of the DMS-100 Switch is to provide local service and
connections to the PSTN public telephone network. It is designed to deliver services over subscribers'
telephone lines and trunks. The DMS-100 Switch is the biggest seller of a line of Digital Multiplex
System (DMS) telephone exchange switches manufactured by Nortel Networks.

It provides Plain Old Telephone Service (POTS), mobility management for cellular phone systems,
sophisticated business services such as Automatic Call Distribution (ACD), Integrated Services Digital
Network (ISDN), and Meridian Digital Centrex (MDC), formerly called Integrated Business Nework
(IBN). It also provides Intelligent Network functions (AIN, CS1-R, ETSI INAP).

DMS switches were not designed to accept traffic from VoIP H.323, SIP trunks, MGCP. Below is a
picture of Nortel DMS switch at the POP site.

DMS switches ports are wired to a DSX frame. A technician is required to do the cross connect between
DMS port and port of another equipment which could be SDH port or MGX-CESM port

Nortel :SOFT SWITCH Communication server 2000 ::CS2K

Softswitch is an electronic switching system designed to support next generation networks that rely on
packet-based voice, data and video communications technologies that can interface with a variety of
transport technologies including copper, wireless, and fibre. A softswitch connects calls from one
phone line to another, entirely by means of software running on a computer system. A softswitch is
typically used to control connections at the junction point between circuit and packet networks. The
softswitch generally resides in a building owned by the telephone company called a central office.

The CS 2000 can be used in a packet-only or a hybrid TDM or Packet (ATM and IP) applications.
Soft switch can accept traffic from TDM and also IP based network (from VoIP H.323, SIP trunks,
MGCP). It uses Media Gateway 7400/15000 or PVG –Passport packet voice gateway which acts as
media gateways between IP packet network and the TDM switches.

The PVGs act as media gateways between the IP packet network and the TDM switches and PBXs
(PRI) in service provider networks. The PVGs terminate TDM-based DS0s on E1, DS3, and STM-1
interfaces, and convey the media on these DS0s in Real Time Protocol (RTP) packets over an IP
interface.
The IP network and the PSTN are connected through a gateway.The gateway converts a PCM (pulse
code modulation) voice data applied from the PSTN into a packet to transfer it to the IP network.
Conversely, when the packet is received from the IP network, the corresponding packet is converted
into PCM voice data to be transferred to the PSTN

ISDN PRI via VOIP H.323 has a draw back as Only 10-20 channels are supported not 30 channels
using this method due to limitations in ULL ..IAD type is One access 400
CS2K - IP Switch. Supports or accepts traffic from VoIP H.323, SIP trunks, MGCP, and PRI 10,20,30
are supported (via PVG)

The components of the CS 2000 use the packet network for signalling to each other and to the
gateways. Because the components of the CS 2000 are connected through a packet network, they can
be located close together at a single site or room, or spread far apart in distant locations.
In an internet protocol (IP) network environment, the distance between network components is less of
an issue than in a time division multiplexing (TDM) environment.

The gateway controller (GWC) acts as a protocol converter to create a bridge between media gateways
and the call processing function provided by the CS 2000 XA-Core. The GWC is based on the Motorola
MCPN750 or MCPM905 single board computer (SBC).

Example of Gateway controller card position in the POP

ISDN30 CLLI Naming Convention:

The PRI ISDN naming convention is broken down in the following Table:

Byte Position Field Name Position

1 Connecting Device Type
2 State of physical service
3-5 Customer Identifiction
6 Type of Circuit
7 Control Protocol
8 Product Description
9-11 Trunkgroup Number
P = PRI PABX connected device
B = BRI connected device
R = Remote Access Server / Router (Dial-up
Modem)
W = Wholesale Inter-Connect
V = Voicemail
T = Telstra Usermode
2 = NSW
3 = VIC
4 = QLD
5 = SA
6 = WA
Three Letter Abbreviation (TLA)
H = H.323 VoIP BRI/PRI Circuits.
V = PVG PRI Circuits.
Z = PDTC/SPM (Historical letter)
3 = ETSI-ISDN-PRI (1998)
4 = Australian TS0 014 – ISDN-PRI
D = Bi-Directional Traffic Circuits
M = Wholesale Inter-Connect System Number
R = Disaster Recovery Serices (MCT Only)
I = Telstra Incoming (Special)
C = Call Termination Service
001-999

Below shows how an average PRI (H.323) switch CLLI is derived:

P - PRI PABX Connected Device

4 - QLD Service

FIV - 3 Letter Abbreviation for “Five Star”

H - Defines a H.323 service

3 - Utilises ETSI-ISDN-PRI Control Signalling

D - Bi-Directional Circuit

002 - This is the 2nd Trunkgroup for this customer/region/type

Once all of the Elements are added the CLLI looks like:
> P4FIVH3D002
TECHNICAL TERMS AND MEANINGS

G.703/G.704

G.703 is a specific standard covering physical and electrical characteristics of a digital interface. In the
case of G.703, it presents a standard method for encoding clock and data into a single signal. The
principles in G.703 are applicable to interfaces with data rates ranging from 64 kbps to 2.048 Mbps

Carrier will provide with a point-to-point link that is"unstructured", meaning that you must supply the
timing ("structured" service is called G.704). You would use your G.703 link for interconnecting data
communications equipment, such as bridges, routers, and multiplexers at a data rate of 2.048 Mbps.

Since G.703 requires that the user supply the timing (rather than the carrier), how is this accomplished?

In most cases, the user equipment at one end of the line is set up as a clock master and the equipment
at the other end is set up as a slave. This is particularly important when the DTE interface is X.21 (which
only has a single clock signal), as both directions of the G.703 link signal must use the same timing. If
the DTE equipment has two clock signals, as is the case with EIA-530, then both ends can be set up as
clock master

The unstructured G.703 signal is comprised of 32 time slots of 64Kbps (32 x 64Kbps = 2.048Mbps).
Time slot zero (TS0) is used for synchronization Since the carrier provides no timing, all 32 time slots
(2.048Mbps) are available to the DTE for data. However, G.704 is a structured service,so the carrier
"steals" the 64Kbps in TS0 to provide timing. Therefore, only 31 time slots (1.984 Kbps) are available to
the DTE for data when G.704 service is provided.

DLSAM

The DSLAM equipment at the telephone company (telco) collects the data from its many modem ports
and aggregates their voice and data traffic into one complex composite "signal" via multiplexing.
Depending on its device architecture and setup, a DSLAM aggregates the DSL lines over its
Asynchronous Transfer Mode (ATM), frame relay, and/or Internet Protocol network (i.e., an IP-DSLAM
using PTM-TC [Packet Transfer Mode - Transmission Convergence]) protocol(s) stack.

The aggregated traffic is then directed to a telco's backbone switch, via an access network (AN) also
called a Network Service Provider (NSP) at up to 10 Gbit/s data rates on the Internet backbone.

The DSLAM acts like a network switch since its functionality is at Layer 2 of the OSI model. Therefore it
cannot re-route traffic between multiple IP networks, only between ISP devices and end-user connection
points. Customer Premises Equipment that interfaces well with the DSLAM to which it is connected may
take advantage of enhanced telephone voice and data line signaling features and the bandwidth
monitoring and compensation capabilities it supports.

DSLAM- Cisco
Cisco – 1xDSLAM will have 240 Ports
Cisco - 2xDSLAM 408 Ports
512 POI Pairs if site can support 2DSLAMS. If one DSLAM – 256 pairs

Madisson Frame has 4x16 Blocks . Each Block has 8 Pairs (4x16x8=512 pairs)
512 POI pairs in each DSLAM.
Madisson Block has 8 Pairs

DSLAM- Lucent
Madisson Frame has 672 Pairs terminated on it. (672 POI pairs) . 216 for SHDSL and 432 for ADSL
432 Ports = (3x72 SHDSL) + (3x72 ADSL)

IAD- Integrated Access Device

An Integrated Access Device (IAD) is an access device that can simultaneously deliver traditional PSTN
voice services, packet voice services, and data services (via LAN ports) over a single WAN link. It
aggregates multiple channels of information including voice and data across a single shared access link
to a carrier or service provider PoP (Point of Presence). The access link may be a E1 line, a DSL
connection, a cable (CATV) network, a broadband wireless link, or a metro-Ethernet connection.

We order SHDSL (Symetric High Bit Rate Digital Subscriber Line) to deliver ISDN and Leased line
services.The copper pair from the Exchange to customers end on which SHDSL is delivered is called
ULL

ULL - Unconditional Local Loop

Unconditioned Local Loop or ULL is where a third party rents from Telstra the copper
line between An exchange and a customer Premesis
ULL- Unconditioned Local Loop – Is the copper Wire that connects Telstra exchange in customers area
to customers site address. Analog voice leaves your phone over two wires and heads to the Central
Office [CO]. Once at the CO the signal is sampled at a 8kHz rate and digitized into 8 bits producing
64kbps per separate phone line. ULL is 4 wire or 2 Pairs . One pair to Tx and the other to Rx..Receive
1+2 and Tx 4+5

SHDSL (Symmetric High-bit-rate Digital Subscriber Line)

(SHDSL) is a form of DSL, a data communications technology that enables faster data transmission
over copper telephone lines than a conventional voiceband modem can provide. Compared to ADSL,
SHDSL employs frequencies that include those used by traditional Plain old telephone service (POTS)
to provide equal transmit and receive (i.e. symmetric) data rates. As such, a frequency splitter, or
microfilter, cannot be used to allow a telephone line to be shared by both an SHDSL service and a
POTS service at the same time.

It provides network connectivity between two connection points, delivering symmetrical throughput
rates. This means you receive the same download and uploads speeds.
Default deployment class while ordering SHDSL link is 9g (2320 Kbps) in SMS.
2320 Kbps is the Upstream and downstream speed. DSL refers to a digital signal carried over your
ordinary telephone line;

S, for Symmetrical, means that the digital signal travels at the same speed in both upload and
download. It utilizes a single copper wire pair to achieve speeds 192 to 2312 kbps
Its uses TC-PAM line coding.

SDH (Synchronous Digital Hierarchy)

SDH defines a standard rate of transmission at 155.52 Mbps on Optical Media

The basic unit of SDH is STM-1 (Synchronous Transport Module)

Data is packed in virtual containers (VC) in SDH frames and transmitted at speeds of 155 Mbps

STM-1->155 Mbps
STM-4 –> 4 x STM-1 =622 Mbps
STM-16->16 x STM-1 = 2588 Mbps

63 x (2Mbps) or E1 signals can be multiplexed and transported on a single STM-1 frame.

SDH Frame has 9 rows and 270 Columns .

Sampling rate of a 3400Hz voice channel is 8000Hz
8000Hz x 1 Byte = 64 Kbps
9 x 270 x 8 Bit x 8000 Hz = 155 Mbps which is basic data rate of SDH

ATM Asynchronous Transfer Mode (ATM)

It is a technology designed for the high-speed transfer of voice, video, and data through public and
private networks using cell relay technology. Size of ATM cell is 53 Bytes.

ATM uses switches to virtually connect to other nodes or devices. ATM networks are fundamentally
connection oriented. This means that a virtual connection needs to be established across the ATM
VPI: Virtual path identifier & VCI : Virtual Channel Identifier

A virtual path contains many virtual channels to same endpoint. For example the VPI is 3 for virtual
connection between ATM switch at site-A and Site-B or in other words the tunnel number 3 connects
Site-A to Site-B. Virtual Channels or VCI are analogous to small pipes with in the tunnel. ATM network
have the following Quality of Service (QoS) categories

-CBR (Constant Bit Rate) – The CBR service category is used for connections that transport traffic at a
consistent bit rate, where there is an inherent reliance on time synchronisation between the traffic
source and destination. CBR is tailored for any type of data for which the end-systems require
predictable response time and a static amount of bandwidth continuously available for the life-time of
the connection [2][5]. The amount of bandwidth is characterized by a Peak Cell Rate (PCR). These
applications include services such video conferencing, telephony (voice services) or any type of on-
demand service, such as interactive voice and audio. For telephony and native voice applications CBR
provides low-latency traffic with predictable delivery characteristics, and is therefore typically used for
circuit emulation

-rt-VBR (variable Bit rate) The rt-VBR service category is used for connections that transport traffic at
variable rates — traffic that relies on accurate timing between the traffic source and destination. An
example of traffic that requires this type of service category are variable rate, compressed video
streams. Sources that use rt-VBR connections are expected to transmit at a rate that varies with time
(for example, traffic that can be considered bursty). Real-time VBR connections can be characterized by
a Peak Cell Rate (PCR), Sustained Cell Rate (SCR), and Maximum Burst Size (MBS). Cells delayed
beyond the value specified by the maximum CTD (Cell Transfer Delay) are assumed to be of
significantly reduced value to the application

- Non-Real-Time Variable Bit Rate (nrt-VBR) The nrt-VBR service category is used for connections that
transport variable bit rate traffic for which there is no inherent reliance on time synchronisation between
the traffic source and destination, but there is a need for an attempt at a guaranteed bandwidth or
latency. An application that might require an nrt-VBR service category is Frame Relay interworking,
where the Frame Relay CIR (Committed Information Rate) is mapped to a bandwidth guarantee in the
ATM network

Available Bit Rate (ABR)The ABR service category is similar to nrt-VBR, because it also is used for
connections that transport variable bit rate traffic for which there is no reliance on time
synchronisation between the traffic source and destination, and for which no required
guarantees of bandwidth or latency exist. ABR provides a best-effort transport service, in which
flow-control mechanisms are used to adjust the amount of bandwidth available to the traffic
originator. The ABR service category is designed primarily for any type of traffic that is not time
sensitive and expects no guarantees of service. ABR service generally is considered
preferable for TCP/IP traffic, as well as other LAN-based protocols, that can modify its
transmission behaviour in response to the ABR’s rate-control mechanics

-UBR (Unspecified Bit Rate) : The UBR service category also is similar to nrt-VBR, because it is used
for connections that transport variable bit rate traffic for which there is no reliance on time
synchronization between the traffic source and destination. However, unlike ABR, there are no
flow-control mechanisms to dynamically adjust the amount of bandwidth available to the user.
UBR generally is used for applications that are very tolerant of delay and cell loss. UBR has
enjoyed success in the Internet LAN and WAN environments for store-and-forward traffic, such
as file-transfers and e-mail. Similar to the way in which upper-layer protocols react to ABR’s
traffic-control mechanisms, TCP/IP and other LAN-based traffic protocols can modify their
transmission behaviour in response to latency or cell loss in the ATM network

Service category Typical Application

CBR (Constant Bit Rate) Circuit emulation,videoconferencing
Real-Time Variable Bit Rate (rt-VBR) Compressed video/audio
Non-Real-Time Variable Bit Rate (nrt-VBR) Critical data
Available Bit Rate (ABR) LAN interconnection,
Unspecified Bit Rate (UBR) File transfer, message transfer
IAD: One Access 400 uses Cable: E1 Cross-Over Cable (blue)
• Please ensure E1 cable is not an ethernet cable. You can not use a crossover CAT-5 cable as a
crossover E1 cable – the pins are not connected properly. You either have to buy one or make
one.
• Blue appears in the middle (pins 4+5) at one end, at the other Blue must appear at the LHS (with
the connector clip facing down). And vice-versa.
• The PABX technician needs to connect the Cable from PABX to E1 port for One access 400 IAD
• Pins 4 & 5 are used to transmit Voice and Data
• Pins 1 & 2 are used to receive Voice and Data

Indicator Off Green Red Green Flashing

Lights
Status Switched Of Switched on Switched on not Reboot in
Operational Operational Operational
progress
xDSL Not used Synchronized Loss of Synchronization
Synchronization
in progress on 1
interface
IP Not used All the Connection
connections on Failure on an
IPoA oe PPP0A IPoA or PPPoA
are activated
Aux Not used Data Service Failure on Data
Oprtaional Service
(FRF,CES)
Voice Not used Service Service not
Operational operational
Comm No Voice Compression
communication in activated one or
progress several channel
IAD: Verilink 6450 A (40H IAD) Cable: Straight through Cable

• The PABX technician needs to connect the Cable from PABX to CBR port
• Pins 1 & 2 are used to transmit Voice and Data
• Pins 4 & 5 are used to receive Voice and Data

The WANsuite 6450 is a feature-rich, intelligent WAN access device that manages legacy applications
in an ATM network. The WANsuite 6450 terminates a standards based Symmetrical High-bit Rate
Digital Subscriber Line (SHDSL) that originates from a Digital Subscriber Line Access Multiplexer
(DSLAM) and provides interfaces for the end user’s communications equipment.
The WANsuite 6450 is equipped with the following interfaces: an SHDSL network interface; a Constant
Bit Rate (CBR) port configurable as T1 or E1; a serial port software configurable for V.35, X.21, RS-232,
RS-449 or EIA-530; a 10/100Base-T Ethernet port; and an asynchronous Supervisory port.

Indicator Description
Power The indicator lights green when power is applied to the unit
The indicator lights amber when the unit is in a test mode loop back
Alarm The indicator lights Red if an Alarm condition exists
The indicator lights amber if a yellow alarm condition exists
CBR The indicator is Off when the CBR port has not been configured.
The Indicator lights green when the CBR port link is up and is receiving AAL1 cells
The indicator lights red when the CBR port has been configured and AAL1 cells are
received
The indicator lights Amber when the CBR port link is up But AAL1 cells are not being
received
Net The indicator is Off not illuminated when the network port has not been
configured
The indicator lights green when the network port link is up and the ATM protocol
is established.
The indicator lights red when the network port link is down and the ATM protocol
is not established
The indicator lights Amber when the network port link is up but the ATM protocol
is not established
Serial console port for IAD configuration and management
IAD: Lucent Technologies Cellpipe (40H IAD)
Cable: Straight-Through Cable

• The PABX technician needs to connect the Cable from PABX to CBR port
• Pins 1 & 2 are used to transmit Voice and Data
• Pins 4 & 5 are used to receive Voice and Data

Indicator Description
Mode Normally this indicators lights green.The indicators light is amber while configuration is
being set by the front panel buttons or when the configuration is changed by SNMP or
through the WEB interface.The indicator will remain amber until the changed configuration
is saved , it will revert to green when the new config has been saved
CBR The indicator is Off when the CBR port has not been configured.
The Indicator lights green when the CBR port link is up and is receiving AAL1 cells
The indicator lights red when the CBR port has been configured and AAL1 cells are
received
The indicator lights Amber when the CBR port link is up But AAL1 cells are not being
received
NET The indicator is Off not illuminated when the network port has not been configured
The indicator lights green when the network port link is up and the ATM protocol is
established.
The indicator lights red when the network port link is down and the ATM protocol is
not established
The indicator lights Amber when the network port link is up but the ATM protocol is
not established
ALARM The indicator lights Red if an Alarm condition exists
The indicator lights amber if a yellow alarm condition exists
Power The indicator lights green when power is applied to the unit
The indicator lights amber when the unit is in a test mode loop back
Reset Provides a hardware reset to the unit
Config Sets the unit back to its default Ethernet or HDLC configuration this is same as
maintenance reset. To initiaite this function you must press and hold the CONFIG
button during a power up sequence
 Basic Settings for Pabx Maintainers
Interface – ETSI PRI signalling is used
Time slot 16 is used for signalling
Powertel Switch is set to Network , Customers interface - User
If more than one ISDN unless requested otherwise sequential circuit selection is used
Signalling Mode – Overlap or Enbloc, we prefer Enbloc
CRC-4 is always set to ON
We send 8 Digits to customer
We expect to receive 10 Digits from the customer
Any Long distance , preselection or other previous carrier settings must be removed
from the PABX prior to testing
Ensure maintainer is set for correct amount of channels (ISDN10,20 or 30)
To Identify an AA handset dial 18008011920 from handset

TROUBLESHOOTING

Symptom Possible Cause Solution

No Dial Tone
Inbound and or out bound calls
fail
All inbound calls Fail
Inbound calls from AAPT netw,
inbound calls from other carriers
fail
Inbound calls work but outbound
fail
Poor quality Line drop
outs/ongoing issues
Connection Issue
Problem with the IAD
Service not configured or
activated
Connection Issue
Problem with the IAD
Circuits still turned down
Connection Issue
Pabx not programmed
Service not confirgured correctly
on switch
Port not completed by losing
carrier
Old carrier codes still
programmed on PABX
Service not configured correctly
on switch
Connection Issue
Circuit issues
Check all Physical connections
Check IAD Powered on or
working
Call SCC
Check all physical connections
Check IAD powered ON &
working
Request SCC activate circuits
Check all connections
Check maintainer has
programmed PABX
Call SCC
Call Losing carrier
Check PABX for dialing Prefixes
Call SCC
Check all connections
Raise fault with SCC/NCC
 ISDN via FIBRE – KRONE BLOCK (no IAD)

• If delivered by fibre there will be 2 krone blocks with the Line isolating units between them
• On top of the Krone block will be lablelled
• The bottom krone block will have pair 1 & 2 shorted out
• The pabx maintainer will be required to install CAT-5 cable to the bottom of krone block,
remove the short and cable back to the PABX krone Frame.

Krone Blocks have 10 Pairs

CAT5 Cable

Cat5 cable has 8 Pins

Pair # Wire Pin #

White/Blue 5
1-White/Blue
Blue/White 4
White/Orange 1
2-Wht./Orange
Orange White 2
 White/Green 3
3-White/Green
Green/White 6
4-White/Brown White/Brown 7
Brown/White 8

FUTURE OF VOICE

What is SIP - Session Initiation Protocol?

SIP, short for Session Initiation Protocol is an IP telephony signaling protocol used to establish, modify
and terminate VOIP telephone calls. SIP was developed by the IETF and published as RFC 3261

SIP describes the communication needed to establish a phone call. The details are then further
described in the SDP protocol.

SIP has taken the VOIP world by storm. The protocol resembles the HTTP protocol, is text based, and
very open and flexible. It has therefore largely replaced the H323 standard

What is a SIP trunk

A SIP Trunk provides a Voice over Internet Protocol (VoIP) connection from a Soft Switch or Application
Server to an IP enabled PBX or SIP Gateway. In this design a SIP Trunk is similar in functionality to
that of the traditional and widely used ISDN PRI (TDM) connection.

Voice Engineering is proposing to offer Phase 1 SIP trunking using the Nortel CS2K

The CS2K provides the call capacity,

call control and billingfunctions
Nortel CS2K

IP MG15K

TheFirewall provides TheBCP providesnetwork topology

Firewall BCP hidingand network security for the
network security for the
SIP Signallingand RTP RTP (voice)
RTP
SIP

The IP PBX provides call routing and

IPPBX
phoneserviceat thesite.

Users

How will the SIP trunk be delivered to the Customer

There are a number of elements involved in delivering a SIP trunk to a customer, in brief they are:

 The physical circuit (AAPT network access) will be delivered utilizing the IP/MPLS network via an
Ethernet Access tail. This access will be dedicated for voice only. Ethernet Trunk will not be part of
the initial deployment however it does need to be considered as part of the design to identify any
design specifics to avoid potential re-engineering of the service at a later date. An Ethernet trunk
will provide a single network access for voice and data services, therefore providing benefits of a
single access performing multiple functions.
 The Nortel CS2K will provide call control and billing capabilities along with other functions required
to support the SIP Trunking product. The CS2K has two platforms that can deliver SIP services,
they are:

 Session Server Lines (SSL) – The SSL is the preferred platform to be used for SIP Trunking. It is
designed to offer SIP services with the same functionality to that of TDM ISDN services. The SSL
hardware is installed but not used today. Session Server Trunks (SST) – currently used to provide
VoIP CTS services. The SST will only be considered for customers requiring greater than 500
concurrent calls. These customers will be required to submit via the “Specials” process which is in
line with current VoIP CTS guidelines.

ISDN OVER ETHERNET

Facing the network, the HN404 interfaces with up to four bonded SHDSL copper pairs delivered
from a Hatteras HN4000 to form an E.SHDSL connection. The HN4000 bonds these pairs to create
a maximum access rate of 22816kbps. From the HN4000, there is an uplink connection into the
SDH network, which connects into the Alcatel 7450 ESS7, and then IP terminated at the Alcatel
7750 SR7 in a VPRN. Below is a high level diagram of the new production network. The Ethernet
Access is where the HN404’s role is performed.

In figure above, VLAN 400 carries voice traffic back to the ESS 7450 and the management to the
One Access400 is via VLAN 4094. Both these VLANs are passed transparently through the HN404.
The management VLAN extends back to the ES 7450 and the e-NTU management VLAN.

Bandwidth Requirements and Rate Reach considerations.

To transmit ISDN 30 requires an aggregate bandwidth of ~2.7 Mbps. The HN404 can take 4 pairs, the
bandwidth required per pair is 2.7/4 = 675 Kbps. This equates to a maximum reach of 2.8 Kms of
0.4mm PE Cable or about 2 Kms radial distance from the nearest Telstra exchange where the Hatteras
HN4000 is located. If a HN408 CPE is used instead of the HN404, then this distance goes up to 5 Kms.

The new production network is connected back to legacy Powertel MPLS VoIP network in which CS2K
Softswitch is connected. CS2K works with One400 to provide the ISDN PRI voice service to the
customer. The high level network interconnect diagram is as below:

ONE400 CPE Design

Same as the current VoIP ISDN10/20 solution, an One400 CPE (together with an HN404) will be
delivered to the customer premises. An E1 interface from the One400 (as for the X163 service
replacement) is delivered for customer’s PBX connection.
The same ISDN10/20 PRI service and functionalities will be delivered to the customer over the One400,
and we will deliver 30 voice channels (aka ISDN30) with this solution.
The current One400 configuration for ISDN10/20 needs to be updated in order to deliver the ISDN30
service. The modification is to change the uplink connection from using G.SHDSL to Fast Ethernet.
Fast Ethernet interface is expected to connect to HN404. A dot1q trunk is configured to trunk the
Management traffic (VLAN 4094) and Voice traffic (VLAN 400) back to the HN404.
The One400 configuration for ISDN30 will need to be updated as:

Cable Distance Considerations

The required bandwidth for the ISDN 30 over Midband Ethernet is 2.8 Mbps. When spread over 4 pairs,
this equates to 700 kbps per pair. Comparing this to the deployment class limits, the minimum
deployment class for the ULLs is 9c giving a maximum cable distance of 2.8 Kms of 0.4mm PIUT type
cable or approx 2 Kms radial.

Class Technologymin rate max ratedB Reach Freq

9a SHDSL 192 576UnlimitedUnlimited 96
9c SHDSL 192 776 30.2 2800 132
9d SHDSL 192 1160 32.1 2700 196
9e SHDSL 192 1544 33.9 2600 259
9h SHDSL 192 1800 31.8 2000 300
9f SHDSL 192 2056 29.2 2000 344
9g SHDSL 192 2312 29.3 1900 388
9i E-SHDSL 192 3496 27.8 1700 439
9j E-SHDSL 192 3840 27.4 1600 481
9k E-SHDSL 192 4096 26.6 1500 513
9l E-SHDSL 192 4352 25.6 1400 545
9m E-SHDSL 192 4608 24.6 1300 577
9n E-SHDSL 192 4864 25.2 1300 609
9o E-SHDSL 192 5120 23.8 1200 641
9p E-SHDSL 192 5376 23.9 1200 649
9q E-SHDSL 192 5696 22.7 1100 713