Converged Network Testing Challenges: PSTN Telephony, IP Telephony, Voice Quality, QOS, Test Methodologies

Analyze Assure Accelerate
Converged Network
Testing Challenges
PSTN Telephony, IP Telephony, Voice
Quality, QOS, Test Methodologies
April 2004
Introduction to PSTN Telephony, IP
Telephony Voice Quality, and QOS
• Types of Networks
• Standards
• OSI Reference Model
• Physical Layer Transmission Media and Interfaces
• Public Switched Telephone Network Terminology
(PSTN)
• Packet Switched Data Network (PSDN) Terminology
and Devices
• Voice Quality and Quality of Service (QoS)
Terminology and Measurements
• Converged Network Testing Challenges
Public Switched Telephone Network (PSTN)
• Circuit Switched Network

• Intended for Voice Transmission
• Data Transmission using Modems
• High Quality Connections
– Low Delay
– High Reliability
– Full duplex 56/64 kbps channels
• Basis for Quality
11101101010

IP Network
• Packet Switched Data Network (PSDN)

• Internet Infrastructure
• Protocols: SIP, H323, MGCP, Megaco
• Data path is shared
– Designed to carry data traffic 11101101010
– Path is “Connectionless” rather than “Dedicated”

– Unreliable packet delivery service
– Non-real-time applications
• Used for IP Telephony
Converged (Hybrid) PSTN and IP Telephony
Telephony is migrating toward IP Telephony

IP Telephony - Converged Network Testing
Example: High Level Call Setup and Termination Procedure
Dialed
On
Ring Configure
Configure
Notify
Digits
Hook
Off
Dial
Good
Hook
#
Back
Hello
Bye
Connect Voice
Off Good
On
Hook
Connect Ring Conversation
Hello
Disconnect
Bye
Hook
Signaling Path
Associated Standard Organizations and Forums

OSI Model Layers, Protocols and Interfaces
Layer 7 Protocol
Layer 7 Layer 7
7 Application 7/8 Interface
Layer 6 Protocol
Layer 6 Layer 6
6 Presentation
5/6 Interface
Layer 5 Protocol
Layer 5 Layer 5
5 Session
4/5 Interface
Layer 4 Protocol
4 Transport Layer 4 Layer 4
3/4 Interface
Layer 3 Protocol
3 Network Layer 3 Layer 3
2/3 Interface
2 Data Link Layer 2 Protocol
Layer 2 Layer 2
1/2 Interface
1 Physical Layer 1 Protocol
Layer 1 Layer 1
Node A Virtual Communications Node B

OSI Layers Physical Communications

Twisted Pair Cable
RJ11 – local analog

telephone service (6 Pin
Modular Jack)
Ethernet – Category 5
(CAT5) Cable with TIA568
(often referred to as RJ45,
8 Pin Modular Jack)
T1/E1 – Category 6 (CAT6)
Cable with RJ48C (often
referred to as RJ45, 8 Pin
Modular Jack)
• Twisted Pair
– Pair of wires that are insulated and twisted around each other to
reduce crosstalk or electromagnetic induction between pairs of wires
– Transmission media used for local telephone, Ethernet transmission
(CAT 5) and some T-Carrier systems (CAT 6 Cable)
Types of Connectors
Modular Jacks
Type Number Purpose Picture of

of Pins Plug
RJ11 6 Pins Standard Telco

Jack used in
Home or Office
RJ12 6 Pins Telco 6 or 8 Pin

3 Pairs of T&R Modular Jack
TIA568 Ethernet (CAT 5 (Receptacle)
Connection)
(Often referred 8 Pins
to as RJ45) 10/100/100
Base-T
RJ48C T1/E1 (CAT 6
Connection)
(Often referred 8 Pins
to as RJ45)

Types of Connectors
Type Number Purpose Picture of Plug

of Pins
Multi-Line Telco 25 Pairs
of Tip and Ring
RJ21 50 Pins
Multi-Line Telco
64 Pins 28 T1 or 20 E1
Typically TX pairs are on
one connector and RX
pairs are on another
connector

Coaxial Cable
Coax Cable with BNC connector

• Coaxial Cable
• Round cable with central carrier wire surrounded by fine copper
wire mesh and/or an extruded aluminum sleeve. Cable is covered
with an insulating PVC outer Jacket. Shielded and central wire are
separated by an insulating material such as PVC or Teflon
• High immunity to electrical interference (noise)
• Wide bandwidth (500 MHz)
• Transmission media used for Ethernet, E3 and T3-Carrier systems

Ethernet
• IEEE 802.3
– Carrier Sense Multiple Access with Collision Detection (CSMA/CD)
protocol
– CAT 5 (Category 5 twisted pair Cable) or coaxial cable
• Electrical
– T is for twisted pair
– 10BASE-T -- up to 10 Mbps
– Fast Ethernet or 100BASE-T -- up to 100 megabits per second.
Uses multiple pairs of CAT 5 cable.
– Gigabit Ethernet 1000Base-T -- 1000 megabits per second (1
gigabit or 1 billion bits per second). Uses multiple pairs of CAT 5
cable.
• Optical
– 1 Gigabit
– 10-Gigabit -- up to 10 billion bits per second
Plain Old Telephone Service (POTS)
• Standard Phone Service that you have at you

home or office
Tip Tip
Ring Ring
A POTS Telephone connects from your home or office to the telephone company
Central Office (CO) through a twisted pair of wires that are called Tip And Ring

Origin of Tip and Ring
The terms Tip and Ring originated

from the old Operator Patch Panel
the plug contained three section
Tip, Ring and the Sleeve
• Tip/Ring – two leads of telephone line that are used as

the analog connection to the telephone
– Tip -- end of patch panel plug
– Ring -- second contact of the patch panel plug
– Sleeve – third contact of the patch panel plug
• Sometimes used for special control function on PBXs

Analog Signal vs Digital Signal
Analog Signal – Signal that varies

continuously (continuous function)
over a specified range of values (such
as -10V to +10V)
Digital Signal – Signal that can only

assume discrete values (such as 0V
represents a digital value of 0 and
+5V represents a digital value of 1)

Pulse Code Modulation (PCM)
Analog to Digital (A/D) and Digital to Analog (D/A) Conversion

Analog Digital Quantized Analog
Signal
10111001011
A/D D/A
Tip Tip
Ring Ring
11101101010
D/A A/D
Quantized
Signal
CODEC (64kb/s) CODEC (64kb/s)
Analog Signal (Voice) is Converted to a Digital

Signal for Transmission over the Network

Pulse Code Modulation — PCM
• A Method for Digitizing Speech
• Samples Analog Signal 8000
Times/Second Amplitude
– Nyquist Theorem
– 0-4 KHz limitation .3
KHz
KHz
3.3 44
Frequency
• Each Sample is One of 256
Amplitudes
Amplitudes
– Need 8 bits for amplitude 8 bit samples
• PCM is 64,000 Bits/Second
– 8000 samples/second x 8
bits/sample
8000/Second

Analog To Digital Conversion
Input Signal Digital Output Signal

Encoding (Quantized)
• Input Signal is sampled at 8kb/s with 8 bits/sample (64kb/s) with a CODEC

(COder DECoder) or DAC (Digital to Analog Converter), which converts
Analog to Digital (A/D) and Digital to Analog (D/A)
• Digital Codes are created that represent the associated amplitude of the
signal
• Digital Codes are transmitted across the network
• Received signal (output) is a quantized approximation of the original input
signal

Companding
• Signal is companded to improve the Signal to Noise Ratio (SNR)
• u-Law compander is used in US, Japan, Taiwan
• A-Law compander is used in all other parts of the world

Voice Compression Techniques
• PCM
– No Compression
• ADPCM
• LD-CELP Compression:
• Reduces the required data rate for each
• CS-ACELP channel
• Increases the number of channels that
• MP-MLQ can be connected to a Channel Bank
• Increases the number of channels that
can be multiplexed on a carrier system

Voice Compression Techniques
Speech CODECs
Type Compression (CODEC) Rate Technique Application
Clear Channel 64kbps PCM PSTN, PSDN
G.711 56, 64kbps, A Law, u Law PCM PSTN, PSDN
G.722 48/56/64 kb/s ADPCM, 7 kHz PSDN
G.722.2, AMR-WB 23.85, 23.05, 19.85, 18.25, 15.85, 14.25, 12.65, ACELP Mobile, 3GPP
Adaptive Multi-Rate Wide Band 8.85 and 6.6 kb/s
G.723.1 5.3, 6.3kbps, Annex A CELP PSDN

G.726 16, 24, 32 kbps ADPCM PSDN
G.728 16kbps CELP PSDN
G.729 8, 13kbps, Annex A & B CELP PSDN
AMR, Adaptive Multi-Rate 4.75, 12.2 kbps AMR Mobile
IS-641 7.4 kbps
PDC-EFR 6.7 kbps EFR
GSMEFR 12.2kbps EFR GSM
Enhanced Full Rate
GSMFR 13.2kbps FR GSM

Mobile Full Rate
VMR-WB, Variable-Rate Multi-Mode Wideband up to 144 Kbps 3GPP2
EVRC, IS-127, Enhanced Variable Rate CODEC 8, 13 kb/s, Full rate (171 bits), ½ rate (80 bits), CELP
1/8 rate (16 bits).

Video Compression Techniques
Type Compression (CODEC) Rate Formats Application

H.261 p x 64Kbit/s (p is in the QCIF, CIF PSTN, PSDN
range 1-30).
H.263 20-30kbps and above QCIF, CIF PSTN, PSDN, Video
SQCIF, 4CIF Conferencing, Video
16CIF. SQCIF Telephony
H.264 Less than 1 Mb/s MPEG-4 AVC Internet Protocol-
based broadcast-
quality video
MPEG 2 4 Mbps or higher Progressive broadcast quality
coding video
IS-13818
MPEG4 Less than 1.15Mb/s MPEG-4 Digital television,

Interactive graphics
'ISO/IEC 14496' applications,
Interactive
multimedia

Types of Signals that Pass Through
the Network
Voice/Data Signal
Control Signaling
DTMF, Pulse Dialing
Often referred to as Off Hook/On Hook
Voice/Data Plane and
Control Plane Ringing
Busy Tone

Circuit-Switched Network
Call Routing and Signaling
Trunk Lines Control

Signaling
Physical
Subscriber Connection
Subscriber
Switches
Local Exchange Exchange Voice/Data
Loops (LE or CO) Signal
(LE or CO)
LE = Local Exchange
CO = Central Office
Exchange
(LE or CO)
Subscriber
Exchange
(LE or CO) Dedicated Physical Connection for
Voice/Data Plane and Signaling Plane
Subscriber
during Call (Non SS7)

Typical Access Network Model
Telephone Central
Office (CO)
Remotely Exchange
Located May
Customer Customer Street contain
Premises Premises Cabinet Channel
Bank
Local
Distribution
Point
Local Loop Local Loop Local
Switch
or
Digital Loop Carrier (DLC)
System
(T1/E1, T3/E3, Fractional
T1, Frame Relay, ISDN
PRA or OC1/OC3)
Customer Customer
Premises Premises

T1 – Digital Loop Carrier System
Central
Office (CO)
24 DS0
Channels 24 Lines
(64kb/s T1
each) (1.544Mb/s)
DSO
T1
DS0
Channel Bank Central Office

Remote Terminal Terminal
(RT) (COT)
May be remotely Local Exchange (LE)

located in Street Switch
Cabinet
USA, Taiwan and Japan use T1 or J1

Super Frame (SF)
D4 Framing
Robbed Bit Signaling

(RBS) (8th bit of B
Frames 6&12) A Frame 12
B Frame 6
A Frame 12
1 2 3 4 5 6 7 8 DS0 Time Slot = 8
Frame 6 Bits/Channel
FT
____ CH ______ CH Frame = 24
or CH CH
FS 1 2 6 24 Channels (193 bits)
FR FR ______ FR ______ FR FR
Super Frame (SF) = 12
1 2 6 11 12
Frames (2316 bits)
Multiframe

Extend Super Frame (ESF)
Robbed Bit Signaling D
(RBS) (8th bit of Frame 24
C
Frames 6, 12, 18, 24) • FAS -- Frame Alignment
B Frame 18 Sequence (FAS)
A Frame 12 – 6 bits, 2 kb/s
1 2 3 4 5 6 7 8 Frame 6 DS0 Time Slot = 8
Bits/Channel • CRC -- Cyclic
Redundancy Check
CRC – 6 bits, 2 kb/s
FDL • FDL -- facility data link
FAS CH CH ____ CH ______ CH Frame = 24 – 12 bits, 4 kb/s
1 2 6 24 Channels (193 bits)
FR FR ______ FR ______ FR FR
1 2 11 23 24 Extended Super Frame (ESF) = 24 Frames
Multiframe

SLC-96 (Subscriber Loop Carrier)
TR-TSY-000008 (BellCore)
Central Office
(CO)
Bypass Pair
96 DS0 5 T1
Channels (1.544Mb/s each)
(64kb/s DSO T1
each) CB1
T1
CB1
T1
CB1
T1
DS0
CB1
T1
TR-08 RT
4 Channel Banks
May be remotely Local Digital Switch

located in Street
Cabinet
5th T1 is for Protection Switching

SLC-96 Framing
Robbed Bit Signaling

(RBS) (8th bit of B
Frames 6&12) Frame 12
A
B Frame 6
A Frame 12
1 2 3 4 5 6 7 8 Frame 6 DS0 Time Slot = 8
Bits/Channel
FT
____ CH ______ CH Frame = 24
or CH CH
FS 1 2 6 24 Channels (193 bits)
FR FR ______ FR ______ FR FR
Super Frame (SF) = 12
1 2 6 11 12
Frames (2316 bits)
DS4 SF = 6 SFs
1 2 3 4 5 6
(72 Channels or 13896 bits)

T1 Hierarchy
Plesiochronous Digital Hierarchy (PDH)
Signal Level Trunk Digital Bit Equivalent Circuits

Level Rate
Voice T-1 DS3
DS0 - 64 kb/s 1 - -
DS1 T1 1.544 Mb/s 24 1 -
DS2 T2 6.312 Mb/s 96 4 -
DS3 T3 44.736 Mb/s 672 28 1

E1
Central
Office (CO)
30 DS0
Channels 30/31 Lines
(64kb/s E1
each) (2.048Mb/s)
DSO
E1
DS0
Signaling (DS0)
Central Office
Framing (DSO)
Terminal
(COT)
May be remotely Channel Bank
located in Street Remote Terminal Local Exchange (LE)
Cabinet (RT) Switch
Used in most countries of the world, and on all international links.

Not used in USA, Taiwan and Japan
E1 Framing
Time Slot 0 Time Slot 16 Time Slots 1 to 15, 17 to 31
Even Frames 0, 2,4,6,8,10,11,12,14 Frames 0
1 0 0 1 1 0 1 1 0 0 0 0 X Y X X
FAS MAS
8 Bits/Time Slot Channel Data

Odd Frames 1,3,5,7,9,11,13,15 Frames 1 to15
*CAS or CCS
I I A N N N N N A B C D A B C D 1 2 3 4 5 6 7 8
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31
32 Frames/Time Slots
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
16 Frames / MultiFrames
*CCS May be in other Time Slots

Plesiochronous Digital Hierarchy (PDH)– E1
Signal Carrier Digital Bit Equivalent Circuits

Level System Rate Voice E-1 E3
DS0 - 64 kb/s 1 - -
E1 E1 2.048 Mb/s 30 1 -
E2 E2 8.448 Mb/s 120 4 -
E3 E3 34.368 Mb/s 480 16 1
- G.747 44.736 Mb/s 630 21 -
E4 E4 139.26Mb/s 1920 64 16
E5 E5 565.148 Mb/s 7680 256 64
Data Bit Rates are higher than expected based on the number of voice channels, because
overhead bits (called Justification Bits) that are needed when synchronizing with other
tributaries. Justification Bits are need because the timing on one multiplexer level is
different than the timing on another multiplexer level.

Line Codes
8 Zeros in a row
1 1 0 0 0 0 0 0 0 0 1 1
AMI
BPV BPV
B8ZS
(T1)
BPV BPV
HDB3
(E1)
AMI – Alternate Mark Inversion
B8ZS – Bipolar with 8-Zero Suppression
HDB3 – High Density Bipolar Three
BPV – Bipolar Violation
ISDN S/T and U Interfaces
ISDN Terminal
Telephone
NT1 Company
Terminator Central
S/T Interface U Interface Office (CO)
• Network Termination 1 (NT1)

– Connects to the Subscriber Line
– OSI Layer 1 (Physical and Electrical Termination)
• S/T Interface -- part on the ISDN modem to which all other ISDN devices connect in order to
communicate over ISDN.
– An NT1 must be placed between the ISDN device and the wall jack to terminate the line.
– The point between the NT1 and the wall jack is the U interface. Many ISDN modems and adapters have a
U interface option that builds the NT1 into their hardware. These devices can be connected directly into
the ISDN wall jack. (Note that
– In Europe, the ISDN wall jack is an S/T interface point rather than a U interface point
• U Interface -- Connects NT1 terminated ISDN devices to the telephone company's ISDN
network
– In North America most ISDN devices have a U interface option, which means they have the NT1
termination device built into their hardware and can be connect directly to the ISDN wall jack

ISDN
TA PSTN
PSTN
BRI
2B+D ISDN
ISDN
NT
E1 PRI T1 PRI PSDN

30B+ 2D 23B + D PSDN
PBX
Fax

T1 PRI
One ISDN T1 PRI Frame
1.544 Mb/s
Central
F 1 2 3 4 5 ------ 21 22 23 D Office (CO)
24 Lines
B Channel DS0
ISDN PRA
23 B Channels (T1)
(DS0 64kb/s 1.544 Mb/s
each)
B Channel DS0
D Channel
Central Office
Terminal
Channel Bank (COT)
Remote Terminal
(RT) Local Exchange (LE)
Switch
May be remotely
located in Street
Cabinet

E1 PRI
One ISDN E1 PRI Frame
2.048 Mb/s
Central
F 1 2 3 4 5 ------ 29 30 D D Office (CO)
30 Lines
B Channel DS0
30 B Channels ISDN PRA

(E1)
(DS0 64kb/s 2.048 Mb/s
each)
B Channel DS0
D Channel DS0
D Channel DS0
Central Office
Terminal
Channel Bank (COT)
Remote Terminal
Switch
May be remotely
located in Street
Cabinet

OSI Model for ISDN
OSI Layers ISDN Layers

7 Application
6 Presentation
Call Setup
5 Session
Q.931 (DSS1)
4 Transport
D Channel
3 Network Q.921 (LAPD)
2 Data Link 2B1Q

I.430
1 Physical

ISDN Connection Establishment
and Termination (Q.931 Operations)
Off
On Hook
Stop
Release
Ring
Dial Setup
Setup
Call Release
Connect
Disconnect
RingSetup
Info
BackACK
NT
Complete
Back
Tone Connect
Alerting
ACK
Disconnect
Release
Proceeding Disconnect
Setup
Connect
Connect
Alerting
ACK
Complete Release
Connect
Ringing
NT
Alerting
Calling Exchange Exchange Calling

Terminal Termination Termination Terminal
Voice
On
Dial
Off Hook
#
Hook

Q931 Protocols
• T1 PRI • ETSI
– Lucent – NET5 Switch
• 4ESS
• 5ESS
– Nortel
• DMS100
• DMS250
– US National
• AT&T NI1
• AT&T NI2
FAS and NFAS
• Facility Associated Signaling (FAS)

– FAS is a form of ISDN signaling where each ISDN-BRI or PRI
is assigned its own D- channel
• Non Facility Associated Signaling (NFAS) –
– A single D channel controls multiple primary rate interfaces
(PRIs). The NFAS option extends D-channel control to B-
channels not resident on the same interface. In other words,
one PRI will contain the D channel and 23 B channels, while
the other member PRIs will contain 24 B channels each.
– A single D-channel can control up to 20 interfaces (a maximum
of 479 B-channels, or a maximum of 478 with one B-channel
as a backup).

ISDN BRI
Central
Office (CO)
2 Lines
B Channel DS0 ISDN BRA

2 B Channels
192 kb/s
(DS0 64kb/s
each)
B Channel DS0
U
D Channel D Channel interface
16 kb/s
Central Office
Terminal
Channel Bank (COT)
Remote Terminal
Switch
May be remotely
located in Street
Cabinet

Types of Control Signaling (POTS)
• On Hook. • Dial Tone

• Off Hook • Second Dial Tone
• DTMF Dialing • Busy Tone
(Touchtone)
• Congestion Tone
• Pulse Dialing (Rotary (Trunk Busy)
Dialing)
• Howling Tone
• Ring Indication. • Metering Pulse.
• Ring Back Tone

Standard Telephone Signaling
Loop Start
Standard Telephones use Loop Start
Signaling between the Telephone and the CO Central Office (CO)
When the phone is taken off-hook (lift
handset) the Tip and ring leads are terminated
into a resistance which seizes the phone line
and draws dial tone.
Closes when
Handset is
lifted
Tip
Switch Hook Tip

48VDC Ring
Ring
Loop Start
Draws Dial tone Types of In-Band Control Signals
or Answers Call On Hook
Off Hook
DTMF Dialing
Pulse Dialing
Ringing
Busy Local Exchange (LE)
Trunk Busy Switch
Howling Tone

Private Branch Exchange (PBX)
PBXs use Loop Start, Telephone Company
Ground Start or E&M Central Office (CO)
Signaling between
the PBX and the CO
Analog
Lines
Analog Analog
Trunk Line
Card Card
Line Digital
Cards
Switch
Digital Digital
Trunk Line
Card Card
T1/E1
PRA
PBX or PABX
Local Digital Switch
Located at
Customer
Premises

Ground Start
Subscriber Exchange or Switch

T
switch S2
ground
hook
detector
current
limiter
current
detector
ring
speech detector
circuitry current
detector
R current
S1 limiter

E&M Signaling
Side A Side B
S1
M Current
–48 V
Detect
S2
S3
Current E
–48 V
Detect
side signal S1 S2 S3
A on hook open closed –
A off hook closed open –
B on hook – – open
B off hook – – closed
• Type 1 E&M Signaling

E&M Signaling
Earth (E) and Magnet (M)
Wink Start
Outgoing Call
On Hook Off Hook On Hook
CPE Side (Loop Open) (Loop Closed) Dial Digits Connect with Called Party (Loop Open)
and Conversation
On Hook Wink Ring Back On Hook

CO Side (Loop Open) Tone (Loop Open)
Immediate Start
Outgoing Call
On Hook Off Hook Dial On Hook
CPE Side (Loop Open) (Loop Closed) Digits Connect with Called Party (Loop Open)
and Conversation
(PBX)
On Hook Ring Back On Hook
CO Side (Loop Open) Tone (Loop Open)

Types of Signaling
In-Band Signaling
Robbed Bit Signaling D
(RBS) (8th bit of Frame 24
C
Frames 6, 12, 18, 24) Frame 18
B
A Frame 12
1 2 3 4 5 6 7 8
Frame 6
• Uses bandwidth of the voice channel to setup and maintain call, which
limits the available bandwidth (56kb/s). Robbed Bit Signaling (RBS)
• Used in
– D4 (SF)
– ESF
– SLC-96
– TR-TSY-000008
– GR-303
• Signaling within the Voice channel is also referred to as In-Band Signaling
(e.g. DTMF Dialing, Pulse Dialing, Busy, Ringing, etc.)

In-Band Signaling
Robbed Bit Signaling Codes (SF, ESF)
*Direction State A B C D
ESF
SF
Receive Ringing 0 0 0 0
Receive Off Hook w/Answer Supervision – 0 0/1
SF Framing
Receive Off Hook w/Answer Supervision – 0 1 0 0
ESF Framing
Receive Off Hook 0 1 0 1
Receive On Hook 0 1 0 1
Transmit On Hook 0 1 0 1
Transmit Off Hook/Loop Closed 1 1 1 1
*CPE's perspective (FXS).
Meter Pulse
period duration
Originate dial
pulse pulse pulse
Terminate
• High-frequency signal (out of voice band)

- 12-kHz and 16-kHz meter pulses sent periodically
during a call
- exchange side sends
- subscriber side detects
• Used to determine cost of call on payphone
- for areas outside North America

Caller ID – Timing
t1 t2
4085411010
ring caller ID ring

• Tones sent in-band while subscriber is on hook
– - between first and second ring (N. America)
– - optional before or after first ring (ROW)
– - FSK, 1200 baud
– - DTMF
• Framing Types
– Type 1 -- SDMF (FSK encoded)
• Calling Line ID
– Type 2 -- MDMF (FSK encoded)
• date and time
• - calling line ID
• calling name
Types of Signaling
Out-of-Band Signaling
F 1 2 3 4 5 ------ 29 30 D D
Control Signaling sent

in separate D Channel
• Uses a separate channel to setup and maintain call,

which results in full bandwidth (64kb/s) available for
voice or data transmission.
• Clear Channel – full bandwidth of channel (64kb/s) is
available since control signals are sent in a separate
channel

MF Signaling
MF Trunk MF Trunk
Party A MF Trunk #92
#1221
949-123-1212 #55
Party B
CO A 408-777-7992
CO BD
CO C
Class 5 Office
Class 5 Office
Class 4 Office
CO B
Class 4 Office
MF Control
PBX Signaling
Physical
Connection
Voice/Data
Signal
• MF R1 –15 tone pair combinations
• MF R2 – 30 tone pair combinations
• MF R1.5 – Russian telephone network

GR-303 or V5
Remote Digital Terminal (RDT) Subscriber
Loop Carrier (SLC-96) Access Network
(AN) Integrated Access Device (IAD)
MTA (Cable) Local Exchange
Foreign Exchange Station (FXS) (LE) Switch
GR-303
1 or more
or V5
DSO
Lines of
DS0 T1/E1,
Remotely located in Street T3/E3, HFC,
Cabinet, Apartment Complex, Fiber,
Customer Premises or Enterprise Cable, xDSL Central Office
Terminal (COT)
• GR303 – Telcordia/Bellcore Integrated Digital Loop Carrier System

Generic Requirements, Objectives and Interface (T1)
• V5 -- Electrical, physical, procedural and protocol requirements
connection between Local Exchange (LE) and Access Network (AN)
– V5.1 -- Static multiplexer based interface (E1)
– V5.2 -- Dynamic concentrator based interface (E1
OSI Model and V5 Layers
OSI Layers V5 Layers

7 Application
6 Presentation
5 Session Call Setup

G.964, G.965
4 Transport
D Channel (LAPD)
3 Network Q.921
2 Data Link
G.703
1 Physical

Channel Associated Signaling – CAS
• A means of conveying signaling information. Each

channel has its own associated means of transmitting
the signaling information, which is normally the same
channel as that used for the channel's traffic. Examples
include analog and T1 loop start circuits.
• The signaling information relates to the line signaling,
such as off hook and ringing.

Common Channel Signaling – CCS
• The signaling (to establish and tear down calls) for more than one
channel is carried in a data link that is separate from the traffic for
that channel.
• ISDN
– NFAS (Non Facility Associated Signaling) refers to the transport of
Common Channel Signaling (CCS) over a different circuit than the
circuit used to transport the traffic (voice or other data).
– When NFAS (Non-Facility Associated Signaling) is implemented on a
group of ISDN circuits, the signaling for all channels within the group
is carried on the D-channel of one of the circuits; the other circuits in
the group carry B-channel traffic on all 24 timeslots
• SS7 is an NFAS type of channel signaling

SS7 Signaling on PSTN
SS7 Links
Trunk Lines
SS7
SS7 Local Loops
Network
Network
End Office
Switch To CLEC or
Point of Long
Distance
Presence Carrier
Tandem
Switch
End Office End Office
Switch Switch

Signaling Link Architecture
D B A
D D A
B B
C C C
A
D B A
E
A A E A A Access (A)
F Bridge (B)
Cross (C)
Diagonal (D)
Extended (E)
Fully associated (F)
SS7 Routing and Routesets (Point Code)
245.100.000
Route A Route
245.111.111 245.222.222
C
1 2
Route Route
Routesets (Point Code) B
245 . 111 . 111 245.110.000
Network Cluster Member
Identifier Number Number
Origination Destination Number of Routes Direct (Associated) Route Indirect (Quasi) Route
SSP 1 STP A 2 SSP 1 / STP A SSP1 / STP B / STP A
SSP 1 STP B 2 SSP 1 / STP B SSP1 / STP A / STP B
SSP 1 SSP 2 2 SSP 1 / STP A / SSP 2 SSP 1 / STPB / SSP 2
STP A SSP 1 2 STP A / SSP 1 STP A / STP B / SSP 1

Architecture – 3 Major Devices
SS7
• Service Switch Point (SSP)
Voice
• Signal Transfer Point (STP)

Databases
Voice Mail
• Service Control Point (SCP)
– Common Channel Signaling Switching Office
Call Detail
Reporting
(CCSSO)
Electronic
Yellow Pages

SS7 Call Setup with ISUP
Address
Initial Address
Answer
Release
Release Complete
Message
Complete
Message
Message
Message
SS7 Links
IAM
Voice
REL ACM
ANM
RLC
Dial
OffNumber
Hook
Off Hook

Example: Packet Switched Data Network (PSDN)
Class 5
Switch PC and Modem
PSTN
Application Service Broker
Server
Fax
Call Agent / Trunking POTS
Server Gateway Phones
Trunking Application
Gateway Server
Signaling Service Broker
Media IP Network Gateway Signaling
Server Bandwidth Gateway
Manager Call Agent / Media
Server Server
r
LEC u te
IP Network
Ro Application
Edge Router ge Server
Ed Bandwidth
DSLAM
Access Ed ge
Ro Manager LEC
or DLC
Concentrator Ed ut
er
CMTS Service Broker er Edge Router
ge
ut
Ro
xDSL, Access Ro
Cable Gateway Call Agent / ut
ge
Server er Access
Ed
Subscriber CPE
Gateway
Gateway Bridge/Router Fax
IP Network
IP
PBX PC Bandwidth Media PC
Manager Server
POTS POTS
Phones IXC PC and Modem Phones
Sip or IP
IP PC Phones
Phone SIP or H323 SIP-T or BICC
PC
RTP MCGP or H.248
Analyze Assure Accelerate SS7, V5, PRI
Datagram
Packets take Address E
different paths
Address A
Message (Called
Datagram) Addressed to
Address E
Address D
• Packets may arrive at

destination at different times
Address B and must be put back in order.
• Packets may be impaired as a
Address C result of network impairments
(i.e., jitter, delay, Packet loss)
Virtual Circuit
Packets take the
same path Address E
Address A
Address D
• Packet/Cells arrive at destination

at the same time and in order
• Packets are multiplex through the
Address B
network
• Permanent Virtual Circuit
Address C • Switched Virtual Circuit

User Agent (UA)
Application
Server
(UA)
MGC SIP MGC

(UA) (UA)
IP Soft
IPNetwork
Network Phone
(UA)
• A User Agent (UA) is a logical entity in a SIP network

• Initiates or responds to SIP requests
• Most elements in a SIP network exhibits a UA behavior
• UA-Client initiate SIP requests
• UA-Server responding to SIP request

Call Agent
Call
Agent
SIP
Ethernet
MGCP
Megaco
(H.248)
PSTN Media
PSDN
Analog, Gateway PSDN
Ethernet
BRA, RTP
E1/T1
• Participates in signaling and device control flows originating, terminating or forwarding

messages
• Service logic for supplementary services, e.g. Caller ID, Call Waiting
• Associated Protocols -- SIP, SIP-T, H.323, BICC, H.248, MGCP/NCS, SS7, AIN, ISDN
• Produce details of each call to support billing and reconciliation
• Also known as Media Gateway Controllers (MGC), Softswitches and Call Controllers.

SIP Server/Client
SIP
Proxy/Registrar
Server
SIP
(Call Signaling) SIP
(Call Signaling)
IP
IPNetwork
Network
SIP Client SIP Client SIP

(SIP Phone or RTP Phone or
Terminal) Terminal
• Example: Terminates the SIP signaling and converts it to H.248 or MGCP to set up a call to the
correct subscriber
• SIP Server
– equivalent function to a Call Agent
– primary roles are to route and forward SIP requests, enforce policy (for example call admission control)
and maintain call details records
• SIP Client
– originates or terminates SIP signaling rather than forwarding it to a SIP Phone or other CPE device

Application Server
Application
Server
SIP SIP
Subscriber
RTP
User Media
Agent Server
• Example: Voice mail or conference calling facilities

• Located in the service provider’s network
• Provides the service logic execution for one or more applications
or services that are not directly hosted on the Call Agent
• A Call Agent may be used routes calls to the appropriate
application server when a service is invoked that the Call Agent
cannot itself support

Media Server
Application
Server
SIP or
SIP or
H323
H323
Subscriber
RTP
User Media
Agent Server
• Located in the service provider’s network.

• May be called Announcement Server
• Control protocol is typically H.248 (Megaco) or MGCP, under the control of the call agent or
application server
• Functions
– playing announcements
– mixing – providing support for 3-way calling etc
– codec transcoding and voice activity detection
– tone detection and generation
– interactive voice response (IVR) processing
– fax processing

Media Gateway
Call
Agent
SIP
Ethernet
MGCP
Megaco
(H.248)
PSTN Media
PSDN
Analog, Gateway PSDN
Ethernet
BRA, RTP
E1/T1
• A media gateway provides translation of protocols for call setup and release, conversion of
media formats between different networks:
– Transfer of information between H.323 and SIP networks on an IP Network
– Translation between transmission formats and communication signals and procedures (e.g.
between IP and PSTN)
• Passes call signaling not applicable to the media gateway through to the network endpoint
(e.g. supplemental services such as call forwarding)
• Performs call setup and clearing on both sides
• Translates between encoding formats

Signaling Gateway
PSTN
PSTN PSDN
SS7 PSDN
SIGTRAN
Signaling
Gateway

• Acts as a gateway between the call agent signaling and the SS7-based
PSTN
• Signaling gateway is between different packetbased carrier domains.
• Provides signaling translation, for example between SIP and SS7 or
simply signaling transport conversion e.g. SS7 over IP to SS7 over TDM
• SIGTRAN provide PSTN signaling transport over IP

Trunking Gateway
MGC
Sigtran
SIP/H.323
MGCP/
SS7 Megaco
(H.248)
IP
PSTN Network
PSTN
E1/T1/E3/T3 Ethernet
/PRI Trunking RTP
Gateway
• Located in the service provider’s network and as a gateway between the carrier IP network and
the TDM (Time Division Multiplexing)-based PSTN
• Provides trans-coding from the packet based voice, VoIP onto a TDM network.
• Typically controlled by a Call Agent / Media Gateway Controller through a device control
protocol such as H.248 (Megaco) or MGCP
• In the case of SS7 the signaling gateway function is generally integrated into the MGC function
• In the case of PRI the signaling gateway function is integrated in Tunking Gateway

Alternative Trunking Gateway
MGC
SIP/H.323
MGCP/
Megaco
(H.248)
IP
PSTN Network
PSTN
E1/T1/E3/T3 Ethernet
/PRI Trunking RTP
Gateway

Access Gateway
Call
Agent
SIP/H323
MGCP
Megaco
(H.248)
PSTN PSDN
Analog, PSDN
Ethernet
BRA,
RTP
E1/T1
Access
Gateway

• Provides support for POTS phones
• Typically under the control of the Call Agent / Media Gateway Controller
through a device control protocol such as H.248 (Megaco) or MGCP

Access Concentrator
Call
Remotely located in Agent
Apartment Complex,
Customer Premises or
Enterprise (IAD,
MGCP
Residential Gateway,
Megaco
MTA
(H.248)
Subscriber Access
Gateway Concentrator /
Access
T1/E1, T3/E, Gateway
HFC, Fiber,
Cable, xDSL
Located in Local Exchange (LE)

(DSLAM, CMTS)
• Terminates the service provider end of the WAN links used over the “last mile”
• Examples:
– DSLAM in a DSL network, this is a DSLAM
– CMTS in a cable network, MGCP would be the NCS variant
• May also include the Access Gateway functionality
– DLC that combines DSLAM capability with direct POTS termination

Subscriber Gateway
Remotely located in Apartment Complex, Call

Customer Premises or Enterprise (IAD, Agent
Residential Gateway, MTA
MGCP
Megaco
(H.248)
Access
Concentrator
T1/E1, T3/E,
HFC, Fiber,
Cable, xDSL
Subscriber
Gateway Local Exchange (LE) Switch
• Located at the customer premises

• Terminates the WAN (Wide Area Network) link (DSL, T1, fixed wireless, cable etc) at the
customer premises
• Provides both voice ports and data connectivity
• Uses a device control protocol, such as H.248 (Megaco) or MGCP/NCS, under the control of
the Call Agent
• Functionally similar to the Access Gateway but typically supports many fewer voice ports.
• Also known as IADs, Residential Gateways, or MTAs (in a cable network)

Edge Router
Call
Agent
Edge Router
Edge Router
Edge Router
CPE Other
Router PSDN
PSDN Networks
SIP
Terminal
• Controls CoS (Call of Service) and QoS (Quality of Service) using TOS (Type of
Sevice) bits, DIFSERV, RSVP
– ToS bits specify how the network should make trade-offs between throughput, delay,
reliability, and cost.
• Routes IP traffic onto the carrier backbone network
• Provides many other functions and
• May be combined with the Access Concentrator

Bridge/Router
Access
CPE Concentrator
Modem T1/E1, T3/E,
HFC, Fiber,
Bridge/Router Cable, xDSL
• Located at the customer premises

• Terminates the WAN (Wide Area Network) link (DSL, T1, fixed wireless,
cable etc) at the customer premises
• Difference between bridge/router and Subscriber Gateway
– Bridge/router does not provide any native voice support
• Voice services for example SIP phones, can be bridged/routed via this
device

IP Packet – IPV4
Ver IHL Type of Service Total Length

M D Fragment Offset
Identification F F
Time to Live Protocol Header Checksum
Source Address
Destination Address
Options Padding
Data/User Payload

IP Addressing – IPV4
32 Bits
Source Address Field

and
Destination Address Field
NETID HOSTID
W . X . Y . Z
IP Address Decimal: 192 . 168 . 1 . 50
IP Address Binary: 11000000 . 10101000 . 00000001 . 0110010
Subnet Mask Decimal: 255 . 255 . 255 . 0
Subnet Mask Binary: 11111111 . 11111111. 1111111 . 00000000

IP Addressing – IPV4
Class Range of Subnet Mask Number of Number of

Address Networks Hosts per
Network
A 001.x.y.z 255.0.0.0 126 16M
126.x.y.z
B 128.x.y.z 255.255.0.0 16k 66k
191.x.y.z
C 192.x.y.z 255.255.255.0 2M 254
223.x.y.z
3 Classes of IP Addressing
• Each class has a specified range of Network
Address, Subnet Masks
• Class C allow for the largest number of Network
Addresses

Gateway / Router
Gateway / Router
Windows Network Properties Settings
192.168.110.1 192.168.120.1
19
0.0 2.
1 16
8.1 8.
6 12
2.1 0.
0
19
A Gateway/Router
routes the Data
(Packet) from
Source Host Source Terminal to Destination Host
IP Address: 192.168.110.20 the Destination IP Address: 192.168.120.70
Subnet Mask: 255.255.255.0 Terminal
Default Gateway: 192.168.110.1

Transmission Control Protocol (TCP)
Packet Structure
Source Port Destination Port
Sequence Number
Acknowledgement Number
Data Control
Reserved Window
Offset Flags
Header Checksum Urgent Pointer
Options Padding
Data/User Payload
• Connection-oriented protocol
– Guarantees both delivery and sequential
• Buffered Transmission of Data
– Data is buffered at both the sender and receiver
TCP Session
Host A Host B
SYN SYN ACK

SEND
ACK
ACK ACK
DATA
ACK IP ACK
FIN DATA
ACK
FIN IPNetwork
Network

User Datagram Protocol (UDP)
Source Port Destination Port
Length Checksum
Data/User Payload
Packet Structure
User Datagram Protocol (UDP)
• Used to exchange messages (called a datagram) between
computers in the network that use Internet protocol (IP)
• UDP does not provide the service of dividing a message into
packets (datagrams) and reassembling it at the other end
• UDP doesn't provide sequencing of the packets
• The application program must be able to make sure that the entire
message has arrived and is in the right order
• Layer 4 (Transport Layer) of the OSI Reference Model
UDP Session
Host A Host B
UDP DATA IP
IPNetwork
Network

RTP/RTCP
Real-Time Transport Protocol (RTP)

• Provides end-to-end delivery services of real-time Audio (G.711, G.723.1,
G.728, etc.) and Video (H.261, H.263),
• Data is transported via the user datagram protocol (UDP).
• RTP provides payload-type identification, sequence numbering, time
stamping, and delivery monitoring.
• UDP provides multiplexing and checksum services.
• RTP can be used with other transport protocols.
Real-Time Transport Control Protocol (RTCP)
• Counterpart of RTP that provides control services
• Primary function of RTCP is to provide feedback on the quality of the data
distribution – RTP-XR
• Carries transport-level identifier for an RTP source
– Used by receivers to synchronize audio and video.

H.323, SIP, MGCP, Megaco/H.248
• H.323
– IP communications protocol for real-time voice and video over IP.
– Includes core protocol and gatekeeper toolkits.
– International Telecommunications Union (ITU) recommendation for audio, video, and
data communications across IP-based networks.
• SIP (Session Initiation Protocol)
– Signaling protocol for establishing real-time calls and conferences over IP networks.
– SIP is an IETF (Internet Engineering Task Force) Protocol.
• MGCP (Media Gateway Control Protocol)
– A complementary IETF protocol to H.323 and SIP
– Defines the communication procedures for a Media Gateway Controller to provide
instructions and to gather information from Media Gateways
• Megaco/H.248 (Media Gateway Control)
– Similar to MGCP, jointly defined by the IETF and ITU-T SG-16
– Gradually replacing MGCP

H.323
• ITU-T standard
• Set of standards that defines real-time multimedia communications
for packet-based networks (IP telephony) and Local Area
Networks (LANs) that do not provide a guaranteed Quality of
Service (QoS)
– H.32x that provides multimedia communication services over a variety
of networks.
• Audio (G.711, G.723.1, G.728, etc.), Video (H.261, H.263), and
Data communications across IP-based networks
• Used Softswitches, Call Agents, Gatekeepers, Central Office
Switches, Gateways (Trunking, Residential and Access), Network
Access Servers, Cable Modems, PBXs, IP Phones, Soft Phones
and IADs
• Directs the call to its destination
• Terminates a calls from another gateways and sends them to the
PSTN
• Example: Netmeeting

H.323
Audio/Video
Terminal Control and Management
Applications
Voice Codec
G.711, 723, 729, etc. H.225 H.225 H.245
RTCP
H.261, H.263
(Real Time RAS Call (Logical
Channel
RTP Control Protocol)
(Registration,
Authentication Signaling Signaling)
(Real Time Protocol) & Status)
UDP TCP
(User Datagram Protocol) (Transmission Control Protocol)
IP (Internet Protocol)
Data Link Layer

Physical Layer
H.323 With Direct Endpoint Signaling
My phone number is 1481.

My IP address is 192.168.1.20 Gatekeeper
RCF
ACF
IP
IP IP
IP Connect
Alerting
Proceeding
Voice
RRQ
ARQ
SETUP
IP
IP RTP/UDP
Advantage
Tracks Available Bandwidth
Provides Call Accounting
H.323 Zone
• H.323 components
– terminals (End-Points)
– gateways
– gatekeeper -- provides call-control services
– multipoint control units (MCUs)
• H.323 Inter-works with Other Multimedia Networks
Session Initiated Protocol (SIP)
• IETF Standard RFC 2543 approved 1999
– Newly defined standard – RFC 3261
• Addressing and Naming
– Similar to e-mail addressing
– johndoe@spirentcom.com, johndoe@192.168.1.10,
8185551212@spirentcom.com
• SIP Messages
– Text based protocol.
– Similar to HTTP.
– Can be used together with Encryption
• Session Initiation Protocol for Telephones (SIP-T) – Protocol for
interconnecting the PSTN with Packet Switched Data Networks
• Example: Broadcasting talks, meetings and seminars with
multimedia content, broadcasts of public events, Internet
Messager (IM), Webex

Internet Messenger uses SIP Protocol

SIP
Audio/Video
Terminal Control and Management
Applications
Voice Codec
G.711, 723, 729, etc.
SIP SDP SAP
H.261, H.263 RTCP (Session (Session (Session
(Real Time
RTP Control Protocol)
Initiation Description Announcement
Protocol) Protocol) Protocol)
(Real Time Protocol)
UDP TCP
(User Datagram Protocol) (Transmission Control Protocol)
IP (Internet Protocol)
Data Link Layer

Physical Layer
SIP Commands
• INVITE: Start a session.
• ACK: Confirms response to an invite.
• BYE: Used by the UAC to the server indicating that it

wishes to release the call.
• CANCEL: Cancels a pending request.
• OPTIONS: Requests information about capabilities,

but does not set up a connection.
• REGISTER: Informs the SIP server about a users location.

SIP VoIP Network
SIP/TCP or UDP
IP
IP
VOICE
Invite
BYE
ACK 200 OK
Ringing
RTP/UDP

VoIP with SIP Server
SIP Server
SIP/TCP or UDP
Trying
IP IP
200 OK
Ringing
BYE
ACK
VOICE
Invite
IP RTP/UDP

Session Initiated Protocol for Telephones
(SIP -T)
SIP-T
POTS MGC MGC

ISUP ISUP
SIP-T SIP-T
H.248/Megaco H.248/Megaco
SIP Server
PSTN
PSTN PSTN
PSTN
RTP/RTCP
Trunking
IP Gateway
Trunking IPNetwork
Network
Gateway
POTS
• IETF RFC 3372

• Session Initiation Protocol for Telephones (SIP-T) – Protocol for
interconnecting the PSTN with Packet Switched Data Networks

Media Gateway Control Protocol (MGCP)
• Complementary protocol to H.323 and SIP

• Controls media gateways from external call control elements in
decomposed gateway architectures
• IETF RFC 3435
• Although widely used it did not become a standard
• Converts the information carried on circuit-switched public
switched telephone networks (PSTN) to data that can be carried in
packets over an internet protocol (IP) network.
• Media Gateway Control (MGC)
– Interactions with the service logic leading to establishment of a call.
• Media Gate way (MG)
– Adapts media streams as packaged at the network boundaries

Media Gateway Control Protocol
MGC
Config Proxy
Call Server SIP
Agent
GK
H.323
Accounting
MGCP
RTP
SS7
IP Network
PCM PCM
RTP PSTN
PSTN
Media Gateway Media Gateway

Media Gateway Control (H.248/MEGACO)
• New RFC 3525 - The name “Megaco” has been changed to

“Gateway Control Protocol” (GCP)
• Similar to MGCP
• Newest industry standard protocol for decomposed multimedia
gateways jointly defined by the IETF and ITU-T
• IETF RFC 3015
• Jointly developed by the IETF and ITU-T Study Group 16.
• ITU-T Recommendation H.248.
• The protocol used between elements of a physically decomposed
multimedia gateway, i.e. a Media Gateway and a Media Gateway
Controller.
• Intended to replace MGCP as an extended and more standardized
protocol in distributed network architectures
• Exists in two formats: ASCII Text (full and abbreviated) and Binary

Media Gateway Control Protocol (MEGACO/H.248)
Call Signaling
Agent Gateway
SS7
SS7
PSTN
PSTN
MEGACO/H.248
Media Gateway
Multimedia Terminals
(End Points) Media Gateway
Megaco Commands
• End Point Configuration – Configure end point or

multi-points with gateway (e.g., ulaw, Alaw)
• Notification Request – Call Agent notifies Gateway of
some specific event (e.g., telephone on/off hook, fax,
modem, DTMF)
• Create Connection – Sent to Gateway to establish
connection
• Modify Connection – Modifies how Gateway handles
the current connection (e.g., change coding scheme)
• Delete Connection – Indication to disconnect call

Megaco Communications Flow
Call Agent
Signaling Signaling
Notification
EPConfig
Config
Gateway EP Gateway
Req
SS7 STP SS7 STP
PSTN CO Create Modify

Dialed
Switch 1 Connect Connect Ring PSTN CO
Connection Switch 1
Digits IP Voice
Network Off Hook
Off
Dial
Hook
#

Parameters and Impairments that Effect
Voice Quality
• Network Architecture • Jitter
• Types of Access Links • Echo
• QoS controlled Edge Routing • Out of order packets
• Voice coding algorithm • Circuit noise
• A/D and D/A Conversion • External noise
• MTU Size • Packet Loss (Frame Loss)
• Signaling protocol mismatches • One Way Delay (Latency)
• Network faults • Variable Delays (Jitter)
• Link Failure • Background Traffic
(Congestion, Bandwidth,
• Time Drift Utilization, Network Load,
• Route Flapping Load Sharing)
ITU G.113 defines the transmission impairments and their impact on voice quality

IP Network

Voice Quality Prediction using E-Model
• ITU G.107 -- The E-Model, A Computational Model for use in

Transmission Planning
• TIA TSB116 – Voice Quality Recommendations for IP Telephony
• The model predicts the voice quality based on the network
configuration and performance metrics
R = Ro - Is - Id - Ie + A
R Transmission rating factor
Ro Basic signal-to-noise ratio
Is All simultaneous impairments
Id All delays
Ie Equipment Impairment factor
A Expectation factor

Predicting Voice Quality using
Emodel R Scores
P.861 P.862
PAMS PSQM/PSQM+ PESQ
5 0 4.5
1 6.5 -0.5

Speech Compression Impairment
G.711
Best Quality

Effect of Frame Loss on Voice Quality
• Sources of frame loss

– Network congestion
– Frames arrive too late
– Out-of-sequence frames
– Frame errors
• Frame loss will degrade voice quality
• Relationship between voice quality and frame loss depends on the
coding algorithm
– Different coding algorithms have different ways to deal with
frame erasure
– For example, G.711 has an optional feature for Packet Loss
Concealment (PLC)

le Factor
Effect of Packet Loss, PLC and Codecs
G.711 with Packet Loss

Concealment has lower
le Factor
– The bigger the le number, the higher the impairment (0 means

no impairment)
– PLC and VAD implementation techniques effect le
VAD – Voice Activity Detection
No VAD
VAD
Data is intentionally not sent during times of Silence

Impact Of Packet Size (G.729)
20 Bytes Normal size for VoIP applications

= 20 ms Speech
40 Bytes
= 40 ms Speech
80 Bytes
= 80 ms Speech
• Remember: Keep Packets Small for Best

Results

Sources of Fixed Delays
• Algorithmic delay
• Serialization delay
• Propagation delay
• Component delay
Serialization
delay
Algorithmic
and component
delays Propagation
delay

End to End Delay Sources

Network Operating With Constant Delay

Effect of Delay on quality
< 25ms Echo Cancellation Required
S TN > 150 ms (with echo

P cancellation): acceptable
150-400 ms:
acceptable if
delay expected
< 400 ms
unacceptable for
most applications
Echo Impairment on IP network

Effect of Echo Level on Voice Quality
TLER – Talker Echo Return Loss

Sources of Variable Delays (Jitter)
• Variable processing delay

– A busy router or switch will take longer to look up the routing
(address) table
• Queuing delay
– Network congestion

Jitter
Source Destination
t1
t2 t1+∆ t1 on-time
t3 t2+∆ t2 early
t3+∆ t3
late
No jitter Jitter
∆ t1= ∆ t2 = ∆ t3

Jitter
Delay
Time

Removal of Jitter
• Voice gateways typically implement a jitter buffer (playout

buffer) in order to compensate for the jitter
• The jitter buffer adds to the end-to-end delay
Gateway with
jitter buffer
IP Network

Elastic Buffer
• Smoothes Out Voice
Voice Signal
Voice Holding Tank
50ms
Constant Stream

Voice Quality Testing Methods
• Active (Intrusive) Testing

– Sends, Receives and compares Wave Files to measure voice quality
– MOS
– PSQM, PSQM+
– PESQ
• Passive Testing
– R-Value
• Measures Voice Quality on RTP Packets
• Based on E-model
• Japan – J-MOS
– 3SQM (ITU-T recommendation)
• Measures Voice Quality of Voice traffic based on Audio Siginal
• Provides an estimate of PSQM

Mechanisms for Assuring QOS
QoS Class Applications (Examples) Node Mechanisms Network Techniques
(Y.1541)
0 Real-Time, loss sensitive, Constrained Routing and
Strict QoS. Guaranteed no
Jitter sensitive, high Distance
over subscription on links.
interaction (VoIP, VTC)
1 Real-Time, Jitter sensitive, Separate Queue with Less constrained Routing
interactive (VoIP, VTC). preferential servicing, and Distances
Traffic grooming
2 Transaction Data, Highly Constrained Routing and
Interactive, (Signaling) Separate Queue, Drop Distance
3 Transaction Data, priority Less constrained Routing
Interactive and Distances
4 Low Loss Only (Short Long Queue, Drop priority Any route/path
Transactions, Bulk
Data, Video
Streaming)
5 Traditional Applications of Separate Queue (lowest Any route/path
Default IP Networks priority)
• Class of Service (COS) ITU-T Y.1541 defines the 5 classes of service and their application
• Type of Services (TOS)
• TOS and COS are both elements with in an IP Packet
• DIFSER and RSVP provide mechanisms to improve QOS

IP Telephony Test Challenges
• Quality of Service
– How effective are the QoS
mechanisms?
• Voice Traffic
• Data Traffic
• Voice and Data Traffic
• Metrics
– How to measure and interpret
results?
• Packet loss
• Latency
• Jitter
• Call setup time Packets out of
order
• Connection speed
• Throughput
• Message response time
11101101010

Human Scoring (MOS)
• Mean Opinion Score

– specified by ITU-T P.830
– range of results 1 to 5
– 4 or greater is good
• Panel of human listeners
• Problems
– expensive to set up
– cumbersome maintenance
– inconsistent results
Sample MOS Test Results
MOS Scores are calculated from PSQM Scores
Number of calls
Number of calls
G.711 uLaw G.723.1 (6300 bps) uLaw
Comparison of PSQM Scores for G.711Alaw and G.723.1 (6300 bps)

PSQM (ITU-T P.861)
PSQM (Perceptual Speech Quality Measurement)

• Developed by KPN Research in the Netherlands
• Approved by ITU-T, August 1996
Chosen over competing methods
• Intrusive voice quality measurement technique
– Range is 0 to 6.5: where 0 is good, 6.5 is awful, 4 is
intelligible
– Not processor intensive

Sample PSQM, PSQM+ Test Results
Number of calls
Number of calls
G.711 A Law G.723.1 (6300 bps)
Comparison of PSQM Scores for G.711Alaw and G.723.1 (6300 bps)

PESQ (ITU-T P.862)
Perceptual Evaluation of Speech Quality

• Joint development
– KPN (Netherlands) and BT (UK)
• Approved by ITU-T, February 2001
– Accepted as practical replacement, for PSQM and PAMS, for
characterizing voice quality
• Intrusive voice quality measurement technique
– range of results –0.5 to 4.5
– greater than 3.5 is good
• Problems
– algorithm is more complex
– the hardware will process fewer channels than PSQM
• Licensed through Psytechnics

PESQ (ITU-T P.862) Benefits
• Additional diagnostic information to help

identify quality problems
• Accurate at predicting subjective quality (MOS)
in a wide range of applications
• Models and compensates for filtering that takes
place in the telephone handset and in the
network.
• Uses a powerful technique, based on PAMS, to
identify and account for delay changes.

Sample PESQ Test Results
Number of calls
G.711 A Law G.723.1 (6300 bps)
Comparison of PESQ Scores for G.711Alaw and G.723.1 (6300 bps)

New MOS Scores based on R-Factor
Calculations
• MOS-LQ – Estimated Listening Quality MOS
– Close correlation with subjective MOS listening test data
– Does not take into account impairments that cause
conversational quality problems, such as delay, and
– Similar subjective MOS scores from Absolute Category Rating
(ACR) listening tests
• MOS-CQ – Estimated Conversational Quality MOS
– Close correlation with subjective MOS data obtained from
conversational tests
– Incorporates factors, such as delay and recency, which affect
conversational quality, and listening quality.
• MOS-PQ – Estimated P.862 Score
– Close correlation with calculated, non-transformed standard
PESQ scores
Converged Network
Testing Challenges
Integrated IP Telephony Test Migration
System
IP Telephony Testing
requires a solution to
test the migration
from legacy to
converged networks
Interworking of legacy and converged networks

Key Test Scalability, performance, interoperability and conformance
Areas
Voice quality
All essential PSTN and IP Telephony media types, protocols, and

interfaces in a single system
Unique Integrated, real-time results for all performance, conformance
Attributes and functional tests
Extreme capacity with voice quality metrics on 100% of channels

Test Issues
Network Equipment Manufactures
Services
Deployment
Performance
Analysis
Industry Manufacturing
standards Quality Assurance
Performance
Functionality & Testing
Conformance Testing
• Characterize system
before trial
• Validate system
scalability
• Identify capacity limits
Analyze Assure Accelerate • Measure call performance
• Automate regression
Test Issues
Service Providers
Service
Assurance
Initial Deployment
Network Certification
Widespread
Network
Pilot Networks
Deployment
• Facilitate vendor selection

• Identify performance ceilings
• Enable accurate capacity
planning
• End-to-end service assurance
testing
• Improve operational performance
• Improve customer
IP Telephony Test Metrics are required
• Migration Trend
– towards IP Network convergence
– from IPv4 to IPv6
– towards “Wireless” replacing
“Wired”
– to Network centric services
Migration can’t happen without IP Telephony test metrics

Test Issues
• VoIP only test tools

are not adequate.
• A combination of
VoIP and PSTN is a
key ingredient for
facilitating the C2P
transition

• Feature validation • Optimize Circuit Utilization

• Functional testing – What is the performance of the
IP network?
• Application testing
• Voice Traffic Analysis
• Protocol conformance
– Generate a mixture of voice and
• Interoperability testing data frames
• Quality of Service – Must be able to simulate a large
number of sessions to test
• Performance scalability
• Troubleshooting – Monitor performance on a per-
flow basis
• Problem resolution

Voice Quality Testing Challenges
• Active (Intrusive) Testing

– Sends, Receives and compares Wave Files to measure voice quality
– MOS
– PSQM, PSQM+
– PESQ
• Passive Testing
– R-Value
• Measures Voice Quality on RTP Packets
• Based on E-model
• Japan – J-MOS
– 3SQM (ITU-T recommendation)
• Measures Voice Quality of Voice traffic based on Audio Siginal
• Provides an estimate of PSQM

• Quality of Service
– How effective are the QoS
mechanisms?
• Voice Traffic
• Data Traffic
• Voice and Data Traffic
• Metrics
– How to measure and interpret
results?
• Packet loss
• Latency
• Jitter
• Call setup time Packets out of
order
• Connection speed
• Throughput
• Message response time
11101101010

In-house Testing tools
• Lacks credibility with SPs

• Lacks depth of expertise
• Insufficient scalability
• Wastes valuable internal
resources

Other Types of Testing
• Protocol Conformance and Torture Testing
– Analyze VoIP call flow (SIP, H.323, MGCP, Megaco) and conformance to
IETF and ETSI Standards
– Evaluate VoIP Signaling protocols
– Negative testing
• IP Telephony Protocol Analysis
– Monitor and analysis protocol flow diagrams
• IP Network Impairment Emulator
– Emulate IP Network Impairments such as: Delay, Jitter, Packet loss
– Placed in series between Tester and network equipment under test
– Test Examples:
• Ability of equipment to handle impairments
• Evaluate jitter buffer and packet loss concealment algorithms

Demonstration
Signaling Path
Hardware setup to run a test with a Media
Gateway (SIP or H323 to PSTN)
ABACUS 5000
Media Gateway

Example: System Configuration for
a Trunking Gateway (SIP to PSTN)
SIPV2 T1 1/0
408-555-1001 415-555-1001
192.168.1.192 ESF
ICG3 255.255.255.0 B8ZS PCG3
Trunking PRI-Group
G.711 ulaw
Gateway 1 to 23 +1D
T1
FastEthernet0/0 RJ48C
100 Pin-to-Pin
Jumpers Cable
RJ45 NCTE
IP Address
192.168.1.193
255.255.255.0
ABACUS 5000
SC3
Cross-Over
Cable SC3 IP address: 10.30.14.214
IP mask: 255.255.254.0
PC IP address: 10.30.14.10 Gateway IP: 10.30.14.1
IP mask: 255.255.254.0
Gateway IP: 10.30.14.1

SIP or H323 to PSTN (T1 PRI) Call Generation and Testing
1. Channel Status Monitoring

2. Audio Monitoring
3. Monitoring Variances
4. Voice Quality Measurements
– PESQ, PSQM+, MOS
5. Monitoring Call Statistics
6. Compare Transmit and Received Wave files
7. Protocol Monitoring
8. Errors vs Time
9. System Events
10.Testing with Multiple CODECs (G.711ulaw and G.729A/B)
– Affect on Voice Quality Measurements
11.Automatic Report Generation
Spirent Solution
Abacus Product Family
The Future of IP Telephony Testing
Abacus 5000
• IP Telephony Migration platform
• Fully Integrated IP and PSTN Telephony test system
• Cost-effective IP Telephony test system to test low-to-
medium density network element performance
• Focus on new testing methodologies
Abacus2TM
• An Integrated IP and PSTN Telephony test system
• Complete IP/PSTN Telephony test system to test
medium-to-high density network element performance
Abacus1TM
• Scalable and modular PSTN test system
• PSTN traffic generation and switching to test low-to-
medium density network element performance

Abacus Product Family at a Glance
Abacus Abacus 5000 Abacus2TM Abacus1TM
Product IP Telephony Test High-density PSTN PSTN Test System
Family System VoIP Test System
Test Functional, Decodes, Functional, Decodes, Performance with Voice
Methodology Performance, Scalability, Performance, Scalability, Quality
Interoperability, Interoperability, and
Conformance and Voice Voice Quality tests
Quality tests
Payload Media Tones, WAV, Video, Fax Tones, WAV, Video, Tones, WAV, Video,
Fax/Data Modem Fax/Data Modem
Protocols SIP, H.323, MGCP, SIP, H.323, MGCP, CAS, ISDN PRA & BRA,
Megaco/H.248 Megaco/H.248 V5.1, V5.2, GR-303, MF
R1/R2, SS7, SS5, SLC-96
CAS, ISDN PRA, V5.1, V5.2, CAS, ISDN PRA, V5.1,
GR-303, MF R1/R1.5/R2, V5.2, GR-303, MF R1/R2,
SS7, NFAS SS7, NFAS
Interfaces 10/100/1000 Base-T Ethernet 10/100 Base-T Ethernet
T1, E1, T3, E3 T1, E1, T3, E3 T1, E1
Analog (FXO) Two wire monitor Analog (E&M, FXO &

insertion points FXS)

Abacus 5000 Convergence Test
& Measurement
• Convergence Test & Measurement (CTM-5000)
– Scriptable VoIP call flow (SIP, H.323, MGCP, Megaco)
• Complete configurability of VoIP Signaling protocols
• Hundreds of automated test suites for Conformance & torture testing
included for use as-is or to be modified for negative testing
– IP Telephony Protocol Analysis
• Simplifies complexity via visual analysis of flow diagrams
– IP Network Impairment Emulator
• Creates Delay, Jitter, Packet loss anywhere in network
• Placed in series between Tester and network equipment under test
so that test results are not skewed
• Test Examples:
– Evaluate jitter buffer and packet loss concealment algorithms
– Ability of equipment to handle impairments
Spirent provides the only solution to test the migration from legacy to converged networks


Converged Network Testing Challenges: PSTN Telephony, IP Telephony, Voice Quality, QOS, Test Methodologies

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Analyze Assure Accelerate

• Circuit Switched Network

Analyze Assure Accelerate

• Packet Switched Data Network (PSDN)

– Path is “Connectionless” rather than “Dedicated”

Telephony is migrating toward IP Telephony

Analyze Assure Accelerate

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Node A Virtual Communications Node B

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RJ11 – local analog

Type Number Purpose Picture of

RJ11 6 Pins Standard Telco

RJ12 6 Pins Telco 6 or 8 Pin

Analyze Assure Accelerate

Type Number Purpose Picture of Plug

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Coax Cable with BNC connector

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• Standard Phone Service that you have at you

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The terms Tip and Ring originated

• Tip/Ring – two leads of telephone line that are used as

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Analog Signal – Signal that varies

Digital Signal – Signal that can only

Analyze Assure Accelerate

Pulse Code Modulation (PCM)

Analog Signal (Voice) is Converted to a Digital

Analyze Assure Accelerate

Analyze Assure Accelerate

Input Signal Digital Output Signal

• Input Signal is sampled at 8kb/s with 8 bits/sample (64kb/s) with a CODEC

Analyze Assure Accelerate

Analyze Assure Accelerate

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G.723.1 5.3, 6.3kbps, Annex A CELP PSDN

GSMFR 13.2kbps FR GSM

VMR-WB, Variable-Rate Multi-Mode Wideband up to 144 Kbps 3GPP2

Analyze Assure Accelerate

Type Compression (CODEC) Rate Formats Application

MPEG4 Less than 1.15Mb/s MPEG-4 Digital television,

Analyze Assure Accelerate

Analyze Assure Accelerate

Trunk Lines Control

Analyze Assure Accelerate

Analyze Assure Accelerate

Channel Bank Central Office

May be remotely Local Exchange (LE)

USA, Taiwan and Japan use T1 or J1

Robbed Bit Signaling

Analyze Assure Accelerate

Analyze Assure Accelerate

May be remotely Local Digital Switch

5th T1 is for Protection Switching

Robbed Bit Signaling

Analyze Assure Accelerate

Signal Level Trunk Digital Bit Equivalent Circuits

DS1 T1 1.544 Mb/s 24 1 -

DS2 T2 6.312 Mb/s 96 4 -

DS3 T3 44.736 Mb/s 672 28 1

Analyze Assure Accelerate