Voice over Internet Protocol (VoIP)

technology
Presentation prepared by
VoIP unit
Sri Lanka Telecom
E-mail: voip@slt.lk
For Infortel Exhibition
11th to 15th October 2006
BMICH

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Part I - Re-cap of Basics

• What is a protocol?
• Telephony
• Circuit switching
• Important technical terms
• Public switched telephone network (PSTN)
• Internet Protocol (IP) suite
• Internet Protocol networks
• Packet switching
• What is VoIP?
• What is the need for VoIP?
• Growth opportunity for VoIP

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 What is a Protocol?

• A protocol is a special set of rules that end

points in a telecommunication connection use
when they communicate

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Telephony
• Telephony is “communicating at a distance”
• It is “circuit switched”, i.e., there is dedicated channel for
exchange of voice and signaling throughout the
conversation
• Reliable delivery
• End to end

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 Circuit switching
B

‘B’
‘A’rings
Calldials
established
‘B’
End to end path setup

A
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Important technical terms

• Media – refers data / audio / video

• Gateway – a network element that interconnect
two disparate networks such as PSTN and IP
networks
• Signaling – controls that govern how a media
stream is set up, maintained, and gracefully
discontinued
• TDM – Time Division Multiplexing (used in
telecom networks)

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 Public Switched Public Switched Telephone
Network (PSTN)

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Internet Protocol (IP) Suite

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 Internet Protocol (IP) networks

• Use Internet Protocol for communication of data

across “packet switched” network
• Characteristics of IP are
– Connectionless
– Best effort
– Unreliable
– Out of order delivery

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Packet switching

In
The this
firstIPpacket
Therefore,
second network
path
packet we
takes
takenshall
takes
the by
examine
red
green
a packet
coloured how
coloured
in an packets
path
path
IP from
computer ‘A’ travels through
network changes
the IP network
according and reach
to conditions
computer ‘B’
prevailing in the network
at a particular time (eg:
congestion, failure etc)

A
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 What is VoIP?

• VoIP is the transmission of voice traffic in

packets using IP as the transport protocol
• It is the merger of telephony and IP worlds
together

IP network

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What is the need for VoIP?

• Integration of voice and data

• Universal presence of IP
• Maturation of technologies
• Bandwidth consolidation
• The shift to data networks

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 Growth opportunity for VoIP

By 2007, international
VoIP expected to grow
to 127B, representing
54% of all international
traffic, including TDM
Traffic (IDC IP
Telephony Market, 2002)

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Part II - Voice processing in VoIP

• Voice signal
• Digitization
• Compression
• Transmission
• VoIP media stream
• Sampling error
• Sampling rate
• Packet delivery in VoIP

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 Voice signal

The transducer present inside the

The human voice (analog in
mouth piece converts this analog
nature) impacts the diaphragm of
sound signal to a voltage signal
the mouth piece of handset of the
similar in shape, amplitude and
telephone.
timing as shown in figure

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Digitization

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 Compression

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Transmit

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 VoIP media stream

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Sampling error

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 Sampling rate

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Packet delivery in VoIP B

Reception at ‘B’ – note the

Transmitted
Voicesignal
This
Compressed
signalisgenerated
digitized at ‘A’
packets reach ‘B’ unordered

A
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 Voice over packet data flow

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Part III - VoIP protocols

• Main types of VoIP protocols

• Diagrammatic representation of VoIP protocols
• H.323
• MGCP / Megaco (H.248)
• SIP
• SIP vs H.323
• VoIP signaling protocol standards compared
• RTP
• RTCP
• Converged telephony network
• VoIP protocol stack

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 Main types of VoIP protocols

• Call control / signaling

• H.323 by ITU-T
• SIP (Session Initiation Protocol) by IETF
• Call control / signaling, Gateway control
• MGCP (Media Gateway Control Protocol)
• Megaco/H.248
• Bearer (carries media)
• RTP (Real-Time Protocol)
• RTCP (Real Time Control Protocol)

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Diagrammatic representation of VoIP

protocols

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 H.323

• VoIP signaling protocol

• ITU standard and is a protocol suite
• Takes a more telecommunications-oriented approach
• 90%+ of all Service Provider VoIP networks

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H.323 components

™ Terminal
ƒ Video/audio/data client
™ MCU (Media Control Unit)
ƒ Conference control
ƒ Content mixing
™ Gateway
ƒ Protocol translation
™ Gatekeeper
ƒ Address resolution
ƒ Admission control

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 H.323 call flow
Hello
Please enter your
Calling Card
Number and PIN Billing Server
(1) User Dials
Access Number (3) AAA query

(4)AAA response
PSTN
PSTN

(2) IVR prompt VoIP

Network

PSTN POP
(Country B) Other Carrier

PSTN

1st leg
Access call
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H.323 call flow

Hello
Two Stage
Dialling
Billing Server
(6) User Dials
Destination Number

(7) H.323 Call Setup PSTN

PSTN
(5) IVR prompt (8) PSTN Call
VoIP
Setup
Network

PSTN POP
(Country B) Other Carrier

PSTN

1st leg
Access call
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 H.323 call flow
Hello

Billing Server
(11) Billing Start

(10) H.323 Call PSTN

PSTN
Answered
Hello
VoIP (9) PSTN Call
Answered
Network

PSTN POP
(Country B) Other Carrier

PSTN

1st leg 2nd leg 3rd leg

Access call IP Transport Termination call
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H.323 call flow

Hello
Goodbye
Billing Server
(13) Billing Stop
(12) Disconnect

(14) H.323 Call PSTN

PSTN
Disconnect
(15)
VoIP Disconnect
Network

PSTN POP
(Country B) Other Carrier

PSTN

1st leg 2nd leg 3rd leg

Access call IP Transport Termination call
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 MGCP / Megaco (H.248)

• Protocols that have been defined for communication

between media gateway controllers and media
gateways. Commonly used are
– Media Gateway Control Protocol (MGCP)
– H.248 (ITU-T) or MEGACO (IETF)

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SIP

• Another VoIP signaling protocol

• IETF RFC2543
• Takes an Internet-oriented approach
• A text-based protocol

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 SIP components

™ Clients:
ƒ User Agent Client (UAC) / User Agent Server (UAS)
ƒ Originate & Terminate SIP requests
™ Typically an endpoint will have both UAC & UAS, UAC for
originating requests, and UAS for terminating requests
™ Servers:
ƒ Proxy Server - relays call signaling, i.e. acts as both
client and server, operates in a transactional manner,
i.e., it keeps no session state
ƒ Redirect Server - redirects callers to other servers
ƒ Registrar Server - accept registration requests from
users, maintains user’s whereabouts at a Location
Server
ƒ Location Server
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SIP service

SIP
Servers/
Registrar Redirect Location Services

“Where is this
name/phone#?”

3xx Redirection
REGISTER “TAhey moved,
try this address” SIP Proxy
“Here I am”

Proxied INVITE
“I’ll handle it for
INVITE
you”
“I want to talk
to another UA

SIP User
Agents SIP User
Agents
SIP-GW

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 SIP methods

™ Basic messages sent in the SIP environment

¾ REGISTER: UA registers with Registrar Server

¾ INVITE: request from a UAC to initiate a session
¾ ACK: confirms receipt of a final response to INVITE
¾ BYE: sent by either side to end a call
¾ CANCEL: sent to end a call not yet connected
¾ OPTIONS: sent to query capabilities outside of SDP

™ Answers to SIP messages

¾ 1XX – information messages (100 – trying, 180 – ringing, 183 – progress)
¾ 2XX – successful request completion (200 – OK)
¾ 3XX – call forwarding
¾ 4XX – error
¾ 5XX – server error
¾ 6XX – global failure

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Basic SIP call flow

SIP UA1 SIP UA2

INVITE w/ SDP for Media Negotiation

100 Trying

180/183 Ringing w/ SDP for Media Negotiation

200 OK

ACK

MEDIA

BYE

200 OK

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 SIP registration process

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SIP operation in proxy mode

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 SIP operation in redirect mode

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SIP vs H.323

SIP H.323
Encoding textual binary

H.323 covers almost every service,

SIP is modular because it covers basic
such as capability exchange,
call signaling, user location, and
Architecture registration. Other features are in other
conference control, basic signaling,
QoS, registration, service discovery,
separate orthogonal protocols
and so on.

high: ASN, use of several different

Complexity adequate: HTTP-like protocol
protocols (H.450, H.225.0, H.245)

ASN.1 vendor specific

the protocol is open to new protocol
Extensibility features
'nonstandardParam' at predefined
positions only

Use in 3gpp yes no

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 VoIP signaling protocol standards compared

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RTP

• The challenge for the designers of RTP, was to build a mechanism

for robust, real-time media delivery above an unreliable transport
layer (UDP).
• RTP was developed by the Audio/Video Transport working group of
the Internet Engineering Task Force (IETF). RTP is defined by the
IETF proposed standard RFC 1889 published in January 1996. It
has been adopted by the International Telecommunication Union
(ITU) as part of the H.323 series recommendations, and by several
other standards organizations.
• In the TCP/IP model it is hard to say in which layer RTP is in. On the
one hand, it looks as an application layer protocol since it runs in
user space and is linked to the application program. On the other
hand, it is a generic, application independent protocol that just
provides transport facilities, so it looks like a transport protocol. The
best description would be that RTP is a transport protocol
implemented in the application layer.
• Designed to carry a wide variety of data (voice, audio, video)

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 RTP message format
0 1 3 8 16 31

VER P X CC M PTYPE SEQUENCE NUMBER

TIMESTAMP
SYNCHRONIZATION SOURCE IDENTIFIER
CONTRIBUTING SOURCE ID
…...

VER : Version(2 bits) P : Padding(1 bit) X : Extension header(1 bit)

CC : No. of contributing sources(4 bits) M : Periodic Marker (1 bit)
PTYPE : Payload Type(7 bits)
SEQUENCE NUMBER : Sequence no. of message(16 bits) - Is used to identify packets, and to
provide an indication to the receiver of packets are being lost or delivered out of order.
TIMESTAMP : Timestamp of message(32 bits) - Denotes the sampling instant for the first octet of
media data in a packet, and it is used to schedule playout of the media data.
Synchronization source identifier (SSRC): This is chosen by the participants at random when they
join the session.
Contributing source identifier (CSRC) : This is chosen corresponding to the SSRC of the participant
who contributed to the packet

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RTP Encapsulation

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 RTCP

• RTCP provides out-of-band communication (such as

periodic reporting of information such as reception
quality feedback, participant identification, and
synchronization between media streams) between the
endpoints.
• RTCP allows senders and receivers to transmit a series
of reports to one another.
• Although data packets are typically sent every few
milliseconds, the control protocol operates on the scale
of seconds.
• RTCP messages are encapsulated in UDP datagrams.
• UDP port number used is one greater than the port
number of the associated data stream in RTP.

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RTCP message format

V P IC PT Length

Format-specific information

Padding if P=1

V – Version(2 bits) - Current version is 2.

P- Padding(1 bit) – If set indicates indicate that the packet has been padded.
IC – Item count – Indicates the number of items included in the packet.
PT - Packet type – Identifies the type of information carried in the packet (five standard packet types).
Type 200: Sender report – senders periodically send these messages to provide an absolute
timestamp
Type 201: Receiver report – receivers periodically send these messages informing the sender
on the condition of reception
Type 202: Source description message – provide general information about the user who owns
and controls the source
Type 203: Bye message – is used by sender to end a stream
Type 204: Application specific message – allow applications to define their own message type
(eg: subtitles)

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 Converged telephony network

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VoIP protocol stack

TCP/IP OSI Model

Voice Application / Presentation

RTP, RTCP Session

TCP UDP Transport

IP Network

Ethernet, PPP, FR, ATM Data Link

Physical Physical

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 Part IV - VoIP architectures

• Centralized architecture
• Distributed architecture

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Centralized architecture

• Intelligence is in the network and endpoints are

relatively dumb
• Centralizes management, provisioning and call
control
• Similar to PSTN
• Critics claim it stifles innovation of endpoint
features
e.g. MGCP / Megaco / H.248

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 Distributed architecture

• Network intelligence distributed between

• Endpoints and call-control devices
™ Endpoints – IP phones, VoIP G/W, PCs
™ Call control – gatekeepers (H.323) Proxy or redirect servers
(SIP)
• Flexible, easy to add new services
• More complex
e.g. H.323, SIP

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Part V - Performance issues in VoIP

• Delay
• Jitter
• Packet Loss
• Echo
• Bandwidth
• Reliability
• Security

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 Delay

• Average time a packet takes to make its way

through a network end to end
• Major components include Propagation delay &
Processing delay
• Packets exceeding a set delay are dropped
Queuing delay

Propagation delay
Transmission delay
Coding delay
Jitter buffer delay

POTS

IP Network Decoding delay

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Threshold of Delay for VoIP is 150 ms

Jitter

• Jitter is variation in packet arrival time

• Due to the nature of packet networks, packets can travel
from a source to a destination using different paths
resulting in different travel delay
• Speech samples have to be played back at regular
intervals (sampling rate). Otherwise, a severe
degradation in the speech quality can take place
• A delay jitter buffer is used to reorder the packets and
absorb the delay jitter caused by the network.
• The larger the buffer the better is the protection from
delay jitter. However, this will result in larger delays

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 Jitter buffer

in in in

Jitter
Protec-
tion Delay

Delay
Delay

out out out

Delay too big Delay too small

Ideal case
Risk of overflow Risk of empty

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Packet Loss

•• Packet
Packet loss
loss is
is caused
caused by by buffer/queue
buffer/queue overflow
overflow within
within the
the network
network oror by
by late
late
packet
packet arrival
arrival at
at the
the receiver
receiver oror by
by network
network failures
failures
•• For
For real-time
real-time interactive
interactive applications
applications like
like voice,
voice, this
this means
means thethe signal
signal must
must be
be
output
output without
without those
those packets.
packets.
•• Packet
Packet Loss
Loss creates
creates gaps
gaps inin voice
voice communications,
communications, which
which can
can result
result in
in
clicks,
clicks, muting,
muting, oror unintelligible
unintelligible speech.
speech.
•• What
What cancan be
be done
done toto minimize
minimize lost
lost packets?
packets?
–– QoS
QoS classification
classification to to expedite
expedite voice
voice packets
packets
–– Longer
Longer jitter
jitter buffer
buffer (trade
(trade off
off between
between delay
delay and
and distortion)
distortion)
–– Call admission control to prevent congestion
Call admission control to prevent congestion

Maximum Tolerable Packet Loss is 3%

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 Packet Loss (contd.)

• We can make voice transmission robust to small amounts of packet

loss by using Packet Loss Concealment (PLC) algorithms
• These are algorithms that smooth over the gaps in the speech
• Some codecs have a built-in PLC feature, while external PLC is
added to other codecs
• Lost packets are handled by one of the following PLC approaches:
– Replacing lost packet by a silence packet (no speech)
– Repeating the previous packet
– Skipping the lost packet
– Inserting a noise packet with the proper energy level & spectrum
– Most vocoders have internal packet concealment techniques that
optimize the speech quality
• PLC can help for short losses, not effective for long bursts (> 3 or so
packets - 40-60 ms of speech )

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Echo and echo control

Reflection

•• Echoes
Echoes areare caused
caused by by coupling
coupling between
between transmit
transmit and
and receive
receive
paths
paths (“reflection”)
(“reflection”)
•• The
The effect
effect of
of the
the echo
echo onon the
the quality
quality of
of speech
speech depends
depends upon
upon
the
the magnitude
magnitude of of the
the echo
echo and
and the
the delay
delay at
at which
which itit occurs.
occurs.
•• Echoes
Echoes areare more
more problematic
problematic in in VoIP
VoIP due
due toto the
the higher
higher delays
delays
•• Echo
Echo cancellation
cancellation is is critical
critical to
to perceived
perceived voice
voice quality
quality

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 Bandwidth

•• Bandwidth
Bandwidth is is the
the raw
raw data
data transmission
transmission
capacity
capacity of
of aa network
network
•• Bandwidth
Bandwidth required
required per
per VoIP
VoIP call
call will
will
depend
depend onon encoding
encoding standard
standard used,
used,
header
header compression,
compression, and and payload
payload size
size
•• For
For VoIP,
VoIP, bandwidth
bandwidth requirements
requirements are are
usually
usually more
more constant
constant e.g.
e.g. G.711
G.711 VoIP
VoIP Strict Priority
average Absolutely goes through,
average bandwidth
bandwidth required
required isis 100
100 kb/s
kb/s Can starve other apps!
•• Bandwidth for voice services
Bandwidth for voice services and and 8 Prot. Control
associated
associated signaling
signaling must
must take
take priority
priority over
over 7 Voice
that
that of
of best-effort
best-effort Internet
Internet traffic
traffic 6
5 Broadcast Video
4 File Transfer
Bandwidth Reduction causes both Delay 3
and Packet Loss in VoIP 2
1 Web Surfing

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Reliability

•• Traditional
Traditional phones
phones areare powered
powered by by phone
phone lines
lines
and
and continue
continue toto work
work during
during aa power
power outage
outage
•• VoIP
VoIP hardware
hardware isis subject
subject outages
outages because
because itit isis
powered
powered by by household
household electricity
electricity
•• VoIP
VoIP service
service outages
outages maymay be
be caused
caused byby failures
failures
within
within the
the network
network
–– Failover
Failover strategies
strategies are
are desirable
desirable for
for cases
cases
when
when network
network devices
devices malfunction
malfunction oror links
links
are
are broken
broken e.g.
e.g. redundant
redundant equipment
equipment // links
links
–– IP
IP recovery
recovery is
is slow
slow because
because itit uses
uses protocol
protocol
to
to detect
detect and
and reroute
reroute traffic
traffic around
around failures
failures ifif
an
an alternate
alternate path
path exists
exists

X X
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 Security

Multim
edia
Server

IP

Security
Threat/
Attack

A B

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Part VI - Our demonstration

Proud to present the
following research
and development
work carried out in-
house by SLT VoIP
engineers

• Web based calls

• Use of soft phones in
telephony
• IP phones with PSTN
routable numbers
• SMS call back

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 Offering the virtual number service to Sri
Lankan people residing overseas

The demonstrations on display highlights

that SLT VoIP platform is capable of
offering this value added service. The
customer gains the advantage of
possessing a telephone service from Sri
Lanka while overseas, and call his / her
relatives at rates applicable to SLT local
phone charges.
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Part VII – Current VoIP services offered

• International call originations from Sri Lanka to

A-Z countries worldwide through MAXTALK pre-
paid card – available at teleshops. The face
values of such cards are LKR 200/- and LKR
400/-.
• International call terminations to Sri Lanka
through local VoIP wholesale partners.

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