Transcript Slide 1

Voice over Internet Protocol (VoIP)
technology
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Part I - Re-cap of Basics
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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
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
Call
dials
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
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Connectionless
Best effort
Unreliable
Out of order delivery
B
Packet switching
In
this
network
we
shall
The
Therefore,
firstIPpacket
second
path
packet
takes
taken
takes
the
by
examine
how
packets
from
red
green
a packet
coloured
coloured
in an
path
path
IP
computer
‘A’ travels through
network changes
the
IP network
and reach
according
to conditions
computer
prevailing‘B’
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?
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Integration of voice and data
Universal presence of IP
Maturation of technologies
Bandwidth consolidation
The shift to data networks
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
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Voice signal
Digitization
Compression
Transmission
VoIP media stream
Sampling error
Sampling rate
Packet delivery in VoIP
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
Reception at ‘B’ – note the
Transmitted
Voicesignal
This
Compressed
signalisgenerated
digitized at ‘A’
packets reach ‘B’ unordered
A
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B
Voice over packet data flow
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Part III - VoIP protocols
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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
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
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VoIP signaling protocol
ITU standard and is a protocol suite
Takes a more telecommunications-oriented approach
90%+ of all Service Provider VoIP networks
H.323 components
 Terminal
 Video/audio/data client
 MCU (Media Control Unit)
 Conference control
 Content mixing
 Gateway
 Protocol translation
 Gatekeeper
 Address resolution
 Admission control
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H.323 call flow
Hello
Please enter your
Calling Card
Number and PIN
(1) User Dials
Access Number
Billing Server
(3) AAA query
(4)AAA response
PSTN
PSTN
(2) IVR prompt
PSTN
POP
(Country B)
VoIP
Network
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
(5) IVR prompt
PSTN
PSTN
(7) H.323 Call Setup
PSTN
POP
(Country B)
VoIP
Network
(8) PSTN Call
Setup
Other Carrier
PSTN
1st leg
Access call
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H.323 call flow
Hello
Billing Server
(11) Billing Start
VoIP
Network
PSTN
PSTN
(10) H.323 Call
Answered
PSTN
POP
(Country B)
(9) PSTN Call
Answered
Hello
Other Carrier
PSTN
1st leg
Access call
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2nd leg
IP Transport
3rd leg
Termination call
H.323 call flow
Hello
Goodbye
Billing Server
(12) Disconnect
(13) Billing Stop
VoIP
Network
PSTN
PSTN
(14) H.323 Call
Disconnect
PSTN
POP
(Country B)
(15)
Disconnect
Other Carrier
PSTN
1st leg
Access call
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2nd leg
IP Transport
3rd leg
Termination call
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
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Another VoIP signaling protocol
IETF RFC2543
Takes an Internet-oriented approach
A text-based protocol
SIP components
 Clients:
 User Agent Client (UAC) / User Agent Server (UAS)
 Originate & Terminate SIP requests
 Typically an endpoint will have both UAC & UAS, UAC for
originating requests, and UAS for terminating requests
 Servers:
 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
 Redirect Server - redirects callers to other servers
 Registrar Server - accept registration requests from
users, maintains user’s whereabouts at a Location
Server
 Location Server
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SIP service
Registrar
Redirect
Location
SIP
Servers/
Services
“Where is this
name/phone#?”
REGISTER
“Here I am”
3xx Redirection
“TAhey moved,
try this address”
SIP Proxy
Proxied INVITE
“I’ll handle it for
you”
INVITE
“I want to talk
to another UA
SIP User
Agents
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SIP User
Agents
SIP-GW
SIP methods
 Basic messages sent in the SIP environment
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REGISTER: UA registers with Registrar Server
INVITE: request from a UAC to initiate a session
ACK: confirms receipt of a final response to INVITE
BYE: sent by either side to end a call
CANCEL: sent to end a call not yet connected
OPTIONS: sent to query capabilities outside of SDP
 Answers to SIP messages
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1XX – information messages (100 – trying, 180 – ringing, 183 – progress)
2XX – successful request completion (200 – OK)
3XX – call forwarding
4XX – error
5XX – server error
6XX – global failure
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
Architecture
SIP is modular because it covers basic
call signaling, user location, and
registration. Other features are in other
separate orthogonal protocols
H.323 covers almost every service,
such as capability exchange,
conference control, basic signaling,
QoS, registration, service discovery,
and so on.
Complexity
adequate: HTTP-like protocol
high: ASN, use of several different
protocols (H.450, H.225.0, H.245)
Extensibility
the protocol is open to new protocol
features
ASN.1 vendor specific
'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
VER
1
P
3
X
8
CC
M
16
PTYPE
31
SEQUENCE NUMBER
TIMESTAMP
SYNCHRONIZATION SOURCE IDENTIFIER
CONTRIBUTING SOURCE ID
…...
VER : Version(2 bits)
P : Padding(1 bit)
CC : No. of contributing sources(4 bits)
X : Extension header(1 bit)
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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Length – Denotes the length of the packet contents following the common header.
Converged telephony network
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VoIP protocol stack
OSI Model
TCP/IP
Voice
Application / Presentation
RTP, RTCP
Session
TCP
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UDP
Transport
IP
Network
Ethernet, PPP, FR, ATM
Data Link
Physical
Physical
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
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Delay
Jitter
Packet Loss
Echo
Bandwidth
Reliability
Security
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
Transmission delay
Propagation delay
Coding delay
Jitter buffer delay
POTS
IP Network
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Threshold of Delay for VoIP is 150 ms
Decoding delay
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
Protection
Delay
Delay
Delay
out
Ideal case
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out
Delay too big
Risk of overflow
Ideal Jitter Buffer Size for VoIP is 60 ms
out
Delay too small
Risk of empty
Packet Loss
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Packet loss is caused by buffer/queue overflow within the network or by late
packet arrival at the receiver or by network failures
For real-time interactive applications like voice, this means the signal must be
output without those packets.
Packet Loss creates gaps in voice communications, which can result in
clicks, muting, or unintelligible speech.
What can be done to minimize lost packets?
– QoS classification to expedite voice packets
– Longer jitter buffer (trade off between delay and distortion)
– 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 are caused by coupling between transmit and receive
paths (“reflection”)
• The effect of the echo on the quality of speech depends upon
the magnitude of the echo and the delay at which it occurs.
• Echoes are more problematic in VoIP due to the higher delays
• Echo cancellation is critical to perceived voice quality
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Bandwidth
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Bandwidth is the raw data transmission
capacity of a network
Bandwidth required per VoIP call will
depend on encoding standard used,
header compression, and payload size
For VoIP, bandwidth requirements are
usually more constant e.g. G.711 VoIP
average bandwidth required is 100 kb/s
Bandwidth for voice services and
associated signaling must take priority over
that of best-effort Internet traffic
Bandwidth Reduction causes both Delay
and Packet Loss in VoIP
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Reliability
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Traditional phones are powered by phone lines
and continue to work during a power outage
VoIP hardware is subject outages because it is
powered by household electricity
VoIP service outages may be caused by failures
within the network
– Failover strategies are desirable for cases
when network devices malfunction or links
are broken e.g. redundant equipment / links
– IP recovery is slow because it uses protocol
to detect and reroute traffic around failures if
an alternate path exists
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Security
Multim
edia
Server
IP
Security
Threat/
Attack
A
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Part VI - Our demonstration
Proud to present the
following research
and development
work carried out inhouse by SLT VoIP
engineers
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Web based calls
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Use of soft phones in
telephony
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IP phones with PSTN
routable numbers
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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 prepaid 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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