Introduction à voix sur IP - ITU
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Transcript Introduction à voix sur IP - ITU
IIT101 – Introduction to Voice Over IP Technology
www.iitelecom.com
© Internation Institute of Telecommunications inc., 2000-2004
Logistics
Schedule : from 8h30 to 16h30
Break:
10h00 to 10h15
Meal:
11h45 to 13h00
Break:
14h45 to 15h00
Directions
Toilets
Telephones
Concessions
Instructions
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In the event of emergency
Food, drink, smoking
Course content
Objectives:
The participant will be able to:
– Describe the routing of a voice signal in an IP network;
– Define various compression and sampling methods;
– Define the protocols used with IP;
– Describe the architectures and the components of a VoIP
network;
– Describe various services and applications.
Contents:
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LU 1:
Traffic and networks
LU 2:
Signal processing
LU 3:
QoS Protocols
LU 4:
Network architectures and components
LU 5:
Implementation considerations
LU 6:
Applications
Round table
Your name
Your employer
Your roles and functions
Your expertise and your knowledge in telecommunications
OSI, LAN, IP, Voice Networks
Your expectation about the course
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LU 1: Traffic and networks
Training objectives:
The participant will be able to:
– Characterize each traffic type
– Characterize the network types
– Describe the operation and the components of a PSTN
– Describe the operation and the components and of an
IP network
– Identify the performance factors influencing a VoIP
network
Contents:
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Isochronous traffic and the PSTN
Networks and the IP protocol
The integration of Voice over IP
Activity 1.1 - Integration of VoIP
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Why integrate voice in an IP network?
Constraints
Challenges
Market trends
Why integrate voice over IP?
To reduce the costs?
To maximize bandwidth usage
To use a single network?
Telephony on Intranet and Internet
To create new commercial applications?
To integrate voice mail, email and fax
Call Centers
–
CTI (Computer Telephony Integration)
–
Telephony on Internet
Integration of voice-data-images?
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VoIP traffic on the network
Traffic: voice, video…
Isochronous traffic
– Fixed intervals
Throughput
– Constant (voice)
– Variable (video)
Short delays
The IP network:
Asynchronous mode of transfer
–
Process by which the data can be
transmitted at unspecified intervals.
–
Generate variable and unforeseeable
delays
–
T T T T T
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Designed for non isochronous traffic
The traffic adaptation to the network
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3 performance factors
Name three factors which will influence the ambulance delay :
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3 performance factors
Transpose these 3 factors in a network environment and relate them
to packet latencies at a LAN exit point
LAN
LAN
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Router
Router
Constraints / Solutions
Sampling
Delays
Delay variation/jitter
Echo
Compatibility/interworking
Inter local area networks (QoS)
With the PSTN
– Signaling and supervision
– Added value services
Information security
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Define packets
Signal compression
RTP/UDP
QoS
Priority
– RSVP
– ToS/Diffserv Field
– IPv6
Dimensioning/capacity
Equipment
Gateway, gatekeeper
What are the challenges?
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The signal must be transported via various network types
– A great number of technologies and standards are used
– Interworking and interconnections between various
technologies and equipment
The applications, the services, the technologies and the
standards evolve quickly
– Proprietary solutions versus standards
Maintain the quality of the voice signal
– Toll Quality or equivalent
To preserve the current telephone services
– Call transfer, call on hold, caller ID, etc.
Interworking between the traditional voice network and the IP
network
– Seamless services
– Addressing, classification
– Quality of service, MTBF, MTTR
– Security and confidentiality
Billing
Operations and management
Who are the stakeholders?
Users
(residential, commercial)
Internet service providers
Carriers
Equipment vendors
Technology
(platform, chip, software)
Industry standards
(ANSI, ITU, IEEE, IETF)
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Market trends (U.S.)
Million lines
14000
12000
10000
Overall market
8000
6000
4000
Traditional market
2000
2001 2002
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IP Telephony
2003
2004
2005
Source:
Phillips InfoTech 2002
2006
In the year 2000, some figures…
VoIP will account for 75 % of the world voice services in 2007. *
The number of IP Centrex lines should grow from 13,000 in 2001 to 10
million in 2008. The number of standard Centrex lines in 2001 was
estimated at 16,5 million. *
The market for IP PBX should reach 3,9 billion dollars in 2005, which
represents 20 % of the traditional PBX market. **
The companies will migrate their voice service from traditional networks
towards IP networks at a rate which will generate a world market for IP
PBX of 16,5 billion dollars from here to 2006. ***
90 % of the companies operating in multiple sites will migrate towards
IP systems for the transport of the voice in the next 5 years. ****
25 % of Internet users will adopt IP Telephony PC-TO-TELEPHONE
in 2006. ****
*
**
***
****
*****
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Frost & Sullivan
Synergy research
Allied Business Intelligence
Phillips Group
Ovum
In the year 2000, some figures… (cont’d)
Unified messaging applications will bloom in 2004 and 2005. The
expenditure related to these applications should reach 3.5 billion
dollars in 2005 with an annual growth rate of 32.1 % *
Call center systems will constitute nearly 30 % of the market for
VoIP systems from here to 2003 **
* TIA (Telecommunication Industry Association)
** IDC (International Dated Corporation)
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Questions?
?
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LU 2: Signal processing
Training objectives:
The participant will be able to:
– Identify and enumerate the protocols used in the transport of
voice in a IP network
– Enumerate the various sampling procedures and compression
standards
– Describe the various delay sources and describe their effect on
signal quality
– Describe the role and the operation of RTP and RTCP protocols
Contents:
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The protocol suite
Sampling and compression
Delay sources
Bandwidth usage
Delay variation (Jitter)
RTP/RTCP
Activity 2.1 The protocol suite
Laboratory activity
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IP Telephony experimentation
Discovering the protocol suite
Laboratory activity 2.1
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IP telephony experimentation;
Using a protocol analyzer;
Discovering the protocol suite.
Equipment configuration
PSTN
ZZZ-ZZZZ
YYY-YYYY
172.30.1.30
Call
Manager
DHCP &
DNS
POTS
9XXX-XXXX
1/0/0
FX/O
GW1
23XX
.1
YYY-YYYY -> 2300
FX/S
1/1/0
.6
2210
9XXX-XXXX
172.30.2.0/30
Ethernet 10
ETH0
Mbps
Frame-Relay 64
Kbps
172.30.2.4/30
GW2
.2
SER0
.1
.5
23XX
172.30.1.0/24
FX/S
23XX
1/1/0
2110
Trainer
2300
IP
Phone
2210
2110
IP
Phone
POTS
POTS
Phone extensions
Legend
Trainer IP phone:
IP phones manual configuration:
IP phones auto-configuration:
Dialer:
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2300
2301 à 2320
2330 à 2349
2350 à 2370
Ethernet
Frame-Relay
POTS
Laboratory activity2.1 (cont'd)
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Experimenting with IP telephony;
Using a protocol analyzer;
Discovering the protocol suite.
Activity 2.1/IP Telephony and protocol suite
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Your Observations
Discussion
The protocol suite
PCM
Ethernet Checksum (FCS) 4 bytes
Voice sample N O 1
compression
Voice sample N O 2
n bytes
Voice sample N O 3
sampling
packetization
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RTP Header
12 bytes
UDP Header
8 bytes
IP Header
20 bytes
Ethernet/LLC header
18 bytes
Activity 2.2 Sampling and compression
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Sampling
Delay
Echo
Compression
Bandwidth usage
Packets construction
Sampling
Continuous signal conversion into individual samples
sampling
Size of a sample
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Which will be the size (in bytes) of a 20ms voice signal sample when
using a 64 Kb/s PCM?
–
The packet size has a significant effect on the quality of service
– Delay
– Echo
– Delay variation
Packet size versus tolerable delay
Did you see the
storm yesterday?
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He cuts me
all the time
200 - 250 ms is tolerable
Overlaps if higher > 250ms
How is your
sister?
Delay Sources
Serialization time
Propagation time
Processing time
“Processing delay”
Time to sit in memory
“Queuing delay”
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Serialization time
Time necessary for the transmission of the first to the last bit of a
given packet.
Called transmission delay
Delay = L/C
C [ bps ]
10 10 01 10 11 00 01 0
L [ bits ]
Time (ms)
Packet
Length
Circuit speed (kbps)
(Bytes)
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56
128
256
1 536
60
8,6
3,8
1,9
0,3
500
71,4
31,3
15,6
2,6
1 500
214
93,8
46,9
7,8
4 000
571
250
125
20,8
Propagation time
T1
T
2
Delay = T 1 + T 2 [ms]
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Processing time
Synchronization
Sampling, compression
Packetization/depacketization
Decision of route choice (routing)
Multiplexing
…
Processor
Header
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Payload
Header
Payload
Queuing / Time to sit in memory
Header
Payload
Header
Header
Payload
Payload
Header
Memory
Queuing delay
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Payload
End-to-end Delays
Gateway
Gateway
IP
network
PBX
Propagation Delay
Queuing, Processing
and Serialization Delays
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PBX
Queuing, Processing
and Serialization Delays
What is echo?
Do I hear
echo ?
Hello!
Hybrid junction
Hybrid junction
PSTN
Hello!
One way delay;
if higher than 20 - 30 ms,
a person distinguishes the echo.
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How can we cancel echo?
What causes echo?
Residential sector
Telephone exchange
Reception
Local loop
Hybrid
Junction
Telephone
Transmission
2 wire to 4 wire conversion
Bad impedance matching
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Echo is always present
Echo varies according to the delay and the power of the return
signal.
Return
signal ,
reduction
of power
Echo is undetectable
(dB)
Echo poses a problem
Delay (ms)
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How to eliminate/minimize echo?
Decrease the return signal
–
–
Advantage:
• An easy solution
Disadvantages:
• Human intervention
• Reduction of the signal power for the person who speaks
Installation of echo cancellers
–
–
Advantage:
• Eliminate the return signal (echo)
Disadvantage:
• Costs
Reduction of the delays
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Voice compression
Objectives:
– To reduce bandwidth usage.
– To reduce packet size
Compression algorithms are used for video and voice.
Disadvantages:
– Reduction in voice quality
– Introduction of delay (echo)
Bandwidth
(kbit/s)
Not
acceptable
Toll
Quality
Quality
commercial
64
PCM (G.711)
Cellular
32
ADPCM 32 (G.726)
24
ADPCM 24 (G.726)
16
ADPCM 16 (G.726)
8
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LD-CELP 16 (G.728)
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0
CS-ACELP 8 (G.729)
MPC-MLQ 5,3 (G.723.1)
Quality
Mean Opinion Score (MOS)
Evaluation of the sound quality with various compression methods on a
scale of 1 (bad) to 5 (excellent)
A result of 4,0 is considered “Toll Quality”
Test conditions:
–
–
–
With background noise, several people discussing at the same time;
Individuals speaking various languages, men and women;
Etc.
4.1
4
3.9
3.8
3.7
MOS
3.6
3.5
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G.723.1
MPC-MLQ
5,3 kbit/s
G.729a
CS-ACELP
8 kbit/s
G.728
LD-CELP
16 kbit/s
G.729
CS-ACELP
8 kbit/s
40
G.726
ADPCM
32 kbit/s
G.711
PCM
64 kbit/s
3.3
Silence suppression
Automatically deactivated for fax/modem
Background noise generated at the destination
-31 dbm
Voice signal
power
Meter
Bandwidth
Economy
-54 dbm
Time
Voice Signal
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Silence
Voice Signal
Large DATA packets affect voice
Gateway
Gateway
WAN
56 kbit/s
Voice
1 500 bytes data
Voice
Data Packet
Serialization time:
PBX
Voice
Voice
1 500 bytes data
PBX
214 ms
Voice
1 500 bytes data
Voice
Voice packets
60 bytes every 20 ms
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Voice packets
60 bytes every 214 ms
Solution - Fragmentation
Gateway
Gateway
WAN
56 kbit/s
500
Voice
500 Voice
Data packet
Serialization time
PBX
Voice
Voice
1 500 bytes data
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60 bytes every 20 ms
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71,4 ms
PBX
Voice
1 500 bytes data
Voice
Voice packets
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500
Voice packets
60 bytes every 71,4 ms
Packet construction
The packetization
The sample
The sample
12 bytes
RTP
8 bytes
UDP
20 bytes
IP
LLC/802.3
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22 bytes
Header compression
Version
IHL
Type of Service
Identification
Time to Live
Length Total
Flags
Protocol
Fragment Offset
Header Checksum
Address Source
Address Destination
Options
V=2
P
Padding
Port Source
Destination Port
Length
Checksum
X
DC M
Pt
Number Sequence
Timestamp
Synchronization Source (SSRC) To identify
CRTP - Compressed Real-time Protocol - RFC2508
Used on link with low bandwidth
G.729: 20ms@8kb/s gives 20 bytes of information
40 bytes per packet: header IP 20; header UDP 8; header RTP 12
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The 40 bytes of header are compressed to 2-4 bytes
Quiz
Associate the various methods of coding and compression
to the bandwidth used:
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G.711 PCM
___
a) 32, 24, 16 kbit/s
G.726 ADPCM
___
b) 5,3, 6,4 kbit/s
G.728 LD-CELP
___
c) 16 kbit/s
G.729 CS-ACELP
___
d) 64 kbit/s
G.723.1 MPC-MLQ
___
e) 8 kbit/s
Quiz
Which of these statements is responsible for
echo?
a) Too great distances between two telephones
b) Delay between two telephones superior > 30 ms
c) Return Signal too high in the hybrid junction
d) Bad impedance matching in hybrid junction
e) All these answers
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Laboratory activity 2.2
Evaluation of voice quality with IP in relation with packet size.
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Activity 2.3 Delay variation, RTP/RTCP and UDP
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Delay variation causes
RTP and RTCP
The use of UDP
Delay variation/ Jitter
Gateway
Gateway
IP
network
Congestion
PBX
C
B
D2 = D1
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PBX
A
D1
C
B
D2 = D1
A
D1
Delay variation causes
Serialization time
Dynamic route
Variable number of routers
Variable link bandwidth
Propagation time
Source
Dynamic route
Variable link bandwidth
Variable distance
Processing time
Dynamic route
Variable number of routers
Queuing delay
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Dynamic route
A variable number of routers
Variable load on the network
Destination
3
3
3
2
2
3
1
2
2
1
3
1
1
1
Real Time Transport Protocol (RTP)
Provides end-to-end transport functions for the applications that
need real time video and audio.
– Identification of the compression type
– Packet sequencing
– Packet lost detection
– Synchronization
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RTP uses UDP for transport
RFC 1889, January 1996
January 1996 Netscape Live Media based on RTP
Microsoft announced that NetMeeting supported RTP
RTP does not offer
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network resource reservation
guarantees of delivery within the required delay
guarantees of quality of service
guarantees of packets delivery
RTP header description
1 1 1 1 1 1 1 1 1 1 2 2 2 2 2 2 2 2 2 2 3 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
4 bytes
V PX
DC
PT
M
SEQUENCE NUMBER
4 bytes
TIMESTAMP
4 bytes
SYNCHRONIZATION SOURCE (SSRC) IDENTIFIER
CONTRIBUTING SOURCE (CSRC) IDENTIFIERS (1 …)
Useful
payload
CONTRIBUTING SOURCE (CSRC) IDENTIFIERS (… 15)
… Voice Samples …
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V:
Version
P:
Padding
X:
Extension
CC:
CSRC count
PT :
Payload Type
Payload type
Type
Audio/Video
Clock (Hz)
2
G.721
A
8 000
4
G.723
A
8 000
7
LPC
A
8 000
9
G.722
A
8 000
15
G.728
A
8 000
26
JPEG
V
90 000
31
_ H.261
V
90 000
34
_ H.263
V
90 000
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Encoding
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Controlling delay variation/Jitter
router A
20ms
router B
Network
IP
20ms
C
B
A
10
30
50
RTP Timestamp
Inter-packet delay - 20ms
20ms
20ms
80ms
C
B
A
10
30
50
RTP Timestamp
Delay variation/Jitter
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20ms
C
B
A
10
30
50
RTP Timestamp
Delay variations elimination
Real-time Transport Control Protocol (RTCP)
Control protocol intended to work jointly with RTP
Provides information for a RTP session in progress.
– Number of packets received
– Number of packets lost
– Delay between each packet (Jitter)
– Timestamps for end-to-end delay calculation
Services offered
– Monitors the quality of service
– Controls congestion
– Source identification
– Inter-media synchronization (sound and image)
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RTCP header description
1
2
Ver
3
P
4
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7
8 bits
Reception report number
Packet Type
Length
58
6
RTCP control packet types
Value
200
Standard Reports
Sender Report (SR)
(Synchronization, quantity of bytes transmitted)
201
Receiver Report (RR)
(Packets receive/lost, Jitter, Timestamps)
202
Source Description (SDES)
(name, telephone number, email address)
203
Greeting (BYE)
204
Application Definition (APP)
(Future Use)
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TCP/UDP
TCP:
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UDP:
Logical connection
Without connection
Deliveries acknowledgement
Check for data errors
Checks for data errors
Retransmission of the lost or
erroneous segments
Sequence check
Flow control
Do not offer:
– delivery acknowledgement
– flow control
– retransmission
– sequence check
Used for the transport of applications
sensitive to errors but less sensitive
to delays
Used for the transport of real-time
applications sensitive to delays but
where errors are less important than
delays
Quiz
Associate the following protocols to the function:
IP
___
RTP ___
TCP ___
UDP ___
RTCP ___
RSVP ___
a) Bandwidth reservation and quality of service
b) Control the quality of service (time, packet received)
c) Sequencing, synchronization, detection of the lost packets
d) Packet routing on the network
e) Retransmission, flow control , delivery acknowledgement
f) Identification of connections without delivery acknowledgement and
any retransmission
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Real-time Protocol Transport (RTP)
Real-time Transport Protocol Control (RTCP)
Laboratory Activity 2.3
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Demonstration of RTP and RTCP protocols.
Questions?
?
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LU 3 - Quality of service (QoS)
Training objectives :
The participant will be able to describe the protocols used to
support QoS in IP networks, in order to offer a better voice quality
Contents:
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Congestion causes
QoS role
QoS Protocols
– IPv4 TOS Field
– DiffServ (Differentiated Services)
– RSVP (Resource Reservation Protocol)
– MPLS (Multi-Protocol Label Switching)
– IPv6
What is QoS?
Methods (i.e protocols, dimensioning, architectures…) used to
fulfill an application transmission requirements (delays/ errors
rate).
A service allowing to fulfill an application congestion requirements
without affecting its performance
A set of traffic parameters to be managed
– Bandwidth
– Delay
– Delay variation (Jitter)
– Packet Loss
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Can be associated to business priorities through an administration
tool containing rules
The base for SLAs between service operators and clients
QoS role
In the corporate network
To ensure the priority of Real time traffic (voice/video)
To ensure an appropriate level of service to corporate critical
applications
In the service provider network
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QoS allows service providers to offer services with SLA
How can we increase QoS in a network?
By Reducing the delay
Decrease the packet size
Increase the bandwidth
…
By compensating for delay variation
Use RTP
…
By reducing the number of lost packets
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Increase the capacity of the buffer
Increase the bandwidth
…
Priority establishment
Weighted Fair Queuing (WFQ)
Voice
Voice
Data
Voice
Data
Voice
Data
Classification By:
IP address
Protocol
Port/Socket
Transmission
QoS function
Buffer
etc
Router
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Voice
Voice
DiffServ (differentiated services)
Principle
Grants a particular treatment to packets requiring it
Assigns various classes of services to packets
Use of 6 bits in the IP header (IPv4 TOS fields and IPv6 Traffic Class)
Differentiated
Services
Voice
Email, Web
Browsing
ERP (Enterprise
Resource Planning)
E-Commerce
Gold
Classification
of traffic
Platinum Service
low delay
Delay and delivery
guaranteed
Silver
Guaranteed delivery
Bronze
Best Effort Delivery
Voice
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DS Field (DiffServ)
IPv4
Version ToS
Length 1 Byte Len ID Offset TTL Proto CS IP-ITS IP-DADated
0
1
2
3
4
5
6
7
DS Field
TOS Field
0
Bits
0
1
2
3
4
5
6
1
2
3
Type of
Service
RFC 1122
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RFC 1349
5
6
7
7
DSCP
IP
Precedence
4
CU
MBZ
Must
Be
Zero
Currently
Unused
Differentiated Services Codes
Points (DSCP) - RFC 2474
Packet Classification and Code Points
Per-Hop Behaviours (PHB)/DiffServ Codepoints (DSCP)
Expedited
Forwarding
EF
Assured
Forwarding
Rejection Priority
WEAK
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Rejection Priority
AVERAGE
Rejection Priority
STRONG
Classify 1 AF11
001 01 0
AF12 001 10 0
AF13 001 11 0
Classify 2 AF21
010 01 0
AF22 010 10 0
AF23 010 11 0
Classify 3 AF31
011 01 0
AF32 011 10 0
AF33 011 11 0
Classify 4 AF41
100 01 0
AF42 100 10 0
AF43 100 11 0
Best
Effort
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101110
000000
DiffServ Operating modes
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Classifier
– Sorts the packets in traffic classes (per hop behavior)
Example: all the VoIP packets between UDP ports 16384 and 16484
belong to the Premium Class
Marker
– Marks (or colors) the packets by assigning them a DSCP value
Example: the VoIP packet Premium Class will be marked with
DSCP value - 101110
Meter (Optional)
– Checks the conformity with the traffic profile and gives the non-conforming
and conforming packets to the Marker or the Shaper/Dropper for processing
Shaper/Dropper
– Accepts the traffic but with a lower bandwidth (a few packet are put in queue
to conform to traffic profiles) (regulating)
– Rejects the excess of packets in the event of congestion
Resource Reservation Protocol (RSVP)
Control protocol between two network
equipments
– station and router
– router and router
The receiving station specifies the quality
of service (QoS) necessary in the network
before receiving information.
The routers accept or refuse the request,
depending on the network status.
The routers give priority to the traffic having
the greatest quality of service specified.
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Priority Class
RTP, RTCP and RSVP in a multi-media session
Real time applications
Real Time Server
RTP and RTCP
UDP
Router
RTP
RTP
RTP
RSVP
RSVP
• Each media (voice, video, data) are transported in
a different RTP session, with its own RTCP packets
controlling the quality of the session.
• The routers communicate via RSVP to reserve
and control the bandwidth of each session
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RTCP
RSVP
RTP
RTCP
RSVP
RTP
Router
RTP, RTCP and RSVP
Applications
MPLS (Multi-Protocol Label Switching)
Solution developed by IETF (RFC 3036 and 3037)
To improve the performance of IP networks by introducing a
switching mechanism based on the packet label
To introduce traffic management (Traffic Engineering) by selecting
a route based on QoS and by managing the traffic
Combination of several proposals
IP Switching (Ipsilon/Nokia)
Tag Switching (Cisco)
IP Navigator (Cascade/Ascend/Lucent)
ARIS (IBM)
CSR (Toshiba)
IP Switching
Ipsilon 95/96
ARIS
IBM 94
Tag Switching
Cisco 96/97
MPLS
IETF 97/98
CSR
Toshiba 94
IP Navigator
Ascend/Lucent 96
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MPLS Terminology
Label
Label being used to identify the packet and its routing
LER (Label Edge Router)
A router at the edge of the MPLS network which assigns the first label
to the packets at the entry of the MPLS network and removes it at the
exit
LSR (Label Switching Router)
A router or an ATM switch which processes the packets according to
the label
Forwarding Equivalence Class (FEC)
Flow of IP packets transmitted using the same mechanism, processed
in the same manner and identified by the same label
IP
5
Switching Table
Port 1
Port 3
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In
Out
(port, label) (port, label)
Port 2
Port 4
IP
9
(1, 2)
(2, 7)
(1, 4)
(3, 7)
(1, 5)
(4, 9)
(2, 3)
(3, 2)
MPLS header
20 bits
MPLS header
Exp - Experimental (CoS)
S - Bottom of Stack
TTL - Time To Live
3 bits 1 bit
Label
8 bits
Exp S
TTL
IP Packet
MPLS Header
32-bits
Packetization
ATM Cell Header
GFC
VPI
…
VCI
Label Header
PPP Header
(Packet over SONET/SDH)
LAN MAC Header
PPP Header
Label Header
Layer 3 Header
MAC Header
Label Header
Layer 3 Header
IPv6 Flow Label Field Version
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TC
Flow Label
Label Header
…
MPLS Routing in networks
134.5.6.1
Table MPLS
In
Out
(2, 84)
(6, 3)
Label Edge
Router (LER)
134.5.1.5
2
134.5.1.5
6
Routing Table
Destination Next Hop
2
134.5/16
Routing Table
Destination Next Hop
134.5/16
(2, 84)
200.3.2/24
(3, 99)
3
200.3.2/24 200.3.2.1
1
2
MPLS Table
In
Out
(1, 99) (2, 56)
3
5
MPLS Table
In
Out
(3, 56)
(5, 3)
Label Switching
Router (LSR)
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134.5.6.1
200.3.2.1 200.3.2.7
IPv6 - Motivations
Motivations
IP address shortage
Context of Always-on (cell phone, Palm, ADSL, etc.)
NAT unidirectional nature for the VoIP applications
Functionalities
128 bits Addressing (greater address space)
– 340282366920938463463374607431768211455 IP addresses
– 4 million unique addresses per square meter of the earth's
surface
Simplified header
Header extension support for options
Integrated security
Better mobility
QoS support
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Questions?
?
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LU 4 - Architectures and network components
Training objectives :
The participant will be able to:
– Identify the various VoIP standards and
specifications
– Describe the components associated with various
architectures
– Describe the protocols associated with various
architectures
Contents:
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H.323
SIP (Session Initiation Protocol)
MGCP (Media Gateway Control Protocol) and
Megaco
Activity 4.1 H.323, components and architecture
The H.323 standard
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Network Components
H.323 Protocols
VoIP network architecture
ITU H.323
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To be able to transport real time voice and video data on a packet
switched network.
Allow the interworking between applications and equipment from
various manufacturers.
Defines the components and services used.
H.323 v1 approved in May 1996 by ITU
H.323 v2 approved in January 1998 by ITU
H.323 v3 approved in September 1999 by ITU
H.323 v4 approved in November 2000 by ITU
H.323 Specific Protocols
Audio
G.711
G.723.1
G.726
G.728
G.729
Video
H.261
H.263
Data
T.120
Control and management
of the calls
H.225
RAS
H.225
H.245
Q.931
of control
Signaling signaling
RTP/RTCP
Voice and Video
Compression
UDP
Standards
IP
Connection (IEEE 802.3)
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TCP
H.323 Specific Protocols (cont’d)
Audio
Video
G.711
G.723.1
G.726
G.728
G.729
H.261
H.263
Data
T.120
Control and management
of the calls
H.225
RAS
H.225
H.245
Q.931
of control
Signaling signaling
RTP/RTCP
TCP
UDP
Transport
IP
Protocols
Connection (IEEE 802.3)
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H.323 Specific Protocols (cont’d)
Audio
G.711
G.723.1
G.726
G.728
G.729
Video
H.261
H.263
Data
T.120
Control and management
of the calls
H.225
RAS
H.225
H.245
Q.931
of control
Signaling signaling
RTP/RTCP
UDP
Data
Transmission
Protocols
TCP
IP
Connection (IEEE 802.3)
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H.323 Specific Protocols (cont’d)
Audio
Video
G.711
G.723.1
G.726
G.728
G.729
H.261
H.263
Data
T.120
Control and management
of the calls
H.225
RAS
H.225
Q.931
Signaling
H.245
of control
signaling
RTP/RTCP
UDP
Signaling and
CallIPset-up
Protocols
Connection (IEEE 802.3)
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TCP
H.323 environment and components
IP telephone
Gatekeeper
H.323 terminal
MCU
Network
Gateway
IP
PSTN
Access Server
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PBX
H.323 terminal
remote access
Gatekeeper
Gatekeeper: the brain of the H.323 network.
Gateway
Terminal
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MCU
H.323 Gatekeeper Functions
Admission Control which determines if a terminal can receive or
initiate a call
Translation of the phone numbers by determining the
H.323 terminal address to establish the call
514-841-3250
172.31.16.254
E.164 Number
IP Addresses
Bandwidth Control
Billing
Zone Management (Terminals, Gateway, MCU)
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H.323 Gatekeeper Optional Functions
Signaling Control
– Establishes connection between two terminals or simply lets
the terminals communicate between themselves
Authorization of the calls
Management of the bandwidth
– Issue requests for additional bandwidth.
Management of the calls
– Determine if the terminal called is busy.
– Call transfer
Bandwidth reservation
– For the terminals unable to make the reservation
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H.323 zone
Gatekeeper 1
Zone 1
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Gatekeeper 2
Zone 2
H.323 terminals
Must support:
– H.225 Protocol (recording, admission control,
signaling and call set-up with a gatekeeper)
– H.245 Protocol (exchange of functionalities between
the terminals and creation of transfer channels)
– Audio compression G.711
Optional:
– Audio compression G.723, G.729
– Video compression H.261 must be used if video is
supported
– Multipoint multimedia conference T.120
– Multipoint Control Unit (MCU)
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Real-time Protocol (RTP) is used for audio and video
packets transmission.
Gateways
IP
network
Allows connectivity between a H.323 network and a nonH.323 network (e.g. PSTN)
Provides translation functions :
– Converts the transmission formats
– Converts the signaling protocols
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PSTN
GW
H.323 interworking
Zone
H.323
Terminal
H.323
MCU
H.323
Local area IP network
Gatekeeper
H.323
PSTN
Terminal
V.70
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Terminal
H.324
Gateway
H.323
Terminal
H.323
LAN
QoS
Terminal
voice
ISDN
(N-ISDN)
Terminal
H.322
Terminal
voice
Terminal
H.323
ATM
(B-ISDN)
Terminal
H.320
Terminal
H.321
VoIP network Structure
Gatekeeper
Gatekeeper
Gateway
Gateway
PSTN
Private
IP Network
Internet
Gateway
V
Telecommuter
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Gatekeeper
Questions?
?
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Activity 4.2 H.323, Terminal configuration
and RAS
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The H.323 standard
–
Signaling and control
–
H.225/RAS
–
Terminal configuration
Laboratory activity 4.2.1
IP telephone autoconfiguration.
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Laboratory activity 4.2.2
IP telephone configuration and its functionalities.
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H.225 (RAS) - Registration Admission Status
Audio
G.711
G.723.1
G.726
G.728
G.729
Video
H.261
H.263
Data
T.120
Control and management
of the calls
H.225
RAS
H.245
H.225
of control
Q.931
Signaling signaling
RTP/RTCP
TCP
UDP
IP
Connection (IEEE 802.3)
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Call set-up stages
Discovery
To which address must I call?
Registration
I am Bob, do I have the permission
to call?
Call Initialization
I would like to speak to Joe.
Call Negotiation
Here are my reception and
transmission capacity
Channel establishment for the
transfer
Joe was reached.
Exchange of information
Hello! How are you?
Call termination
Hello, see you soon!
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H.225 Messages (RAS)
Remote access terminal
Gatekeeper
IP network
PSTN
Access
Server
PSTN
Gateway
RAS Process
Seek for a gatekeeper
Registration to a gatekeeper
Call control admission
Localization of the called terminal
Change of bandwidth
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H.225 messages
Questions?
?
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Activity 4.3 - H.323, Signaling and Control
The H.323 standard
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Signaling and Control
–
H.225/Q.931
–
H.245
–
TCP/UDP ports Attribution
H.225 Signaling derived from Q.931 and H.245
Audio
G.711
G.723.1
G.726
G.728
G.729
Video
Data
H.261
H.263
T.120
Control and management
of the calls
H.225
RAS
H.245
H.225
of control
Q.931
Signaling signaling
RTP/RTCP
TCP
UDP
IP
Connection (IEEE 802.3)
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Laboratory activity 4.3
H.225 and H.245 protocol analysis
TCP/UDP ports attribution
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Laboratory activity 4.3 (cont'd)
H.225 and H.245 protocol analysis
TCP/UDP port attribution
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Laboratory activity 4.3 (cont'd)
H.225 and H.245 protocol analysis
TCP/UDP port attribution
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Laboratory activity 4.3
Packet
1
2
3
4
6
7
8
9
10
11
12
13
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IP address
IP address
TCP/UDP port
TCP/UDP port
Transport
Type of
source
destination
source
destination
protocol
message
Laboratory activity 4.3
Packet
14
15
16
17
18
19
20
21
22
23
24
25
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Source
Destination
TCP/UDP port
TCP/UDP port
Transported
Type of
IP address
IP address
source
destination
protocol
message
Laboratory activity 4.3
Packet
26
27
28
29
30
31
32
33
34
35
36
37
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IP address
IP address
TCP/UDP port
TCP/UDP port
Transported
Type of
source
destination
source
destination
protocol
message
Laboratory activity 4.3
Packet
IP address
IP address
TCP/UDP port
TCP/UDP port
Transported
Type of
source
destination
source
destination
protocol
message
n - 11
n - 10
n-9
n-8
n-7
n-6
n-5
n-4
n-3
n-2
n-1
n
n = last packet
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Laboratory activity 4.3 (cont'd)
H.225 and H.245 protocol analysis
TCP/UDP port attribution
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H.225 signaling
Audio
G.711
G.723.1
G.726
G.728
G.729
Video
H.261
H.263
Control and management
of the calls
Data
T.120
H.225
RAS
H.245
H.225
of control
Q.931
Signaling signaling
RTP/RTCP
TCP
UDP
IP
Connection (IEEE 802.3)
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H.225 signaling derived from Q.931
Terminal
remote access
Gatekeeper
IP network
PSTN
Initialization
Call set-up
Warning
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PSTN
Access
Server
Gateway
Connection (ports attribution for H.245)
Command for end of session
Release of connection
H.225 signaling
Signaling Model
Determine which protocols go trough the gatekeeper and which
ones pass directly between the two termination points.
Missing gatekeeper from the H.323 network
– exchanged directly between the terminals or gateways
Gatekeeper present in the H.323 network
– exchanged directly between the terminals or gateways (Direct
Call Signaling)
– exchanged between the terminals or gateways after having
passed trough the gatekeeper (Gatekeeper-Routed Call
Signaling)
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The method is chosen by the gatekeeper during the admission.
Direct call signaling
Terminal
Terminal
Signaling H.225 (Q.931)
GK
Address translation
Address translation
Admission control
Admission control
Bandwidth control
Bandwidth control
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Gateway
H.245 Signaling of control
Audio
G.711
G.723.1
G.726
G.728
G.729
Video
Data
H.261
H.263
T.120
Control and management
of the calls
H.225
RAS
H.245
H.225
of control
Q.931
Signaling signaling
RTP/RTCP
TCP
UDP
IP
Connection (IEEE 802.3)
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H.245 messages
Gatekeeper
Remote access terminal
LAN
Router
IP network
PSTN
Gateway
PSTN
Access
Server
Exchange of capacities
Opening and closing the logical channels used for the transfer of
information, voice, video and data.
– Attribution of the RTP and RTCP ports
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H.245 messages
Logical channels
Terminal 2 H.323
Terminal 1 H.323
0
5
6
7
8
Control H.245
Audio
Video
Each logical channel uses a different socket
(IP address IP + logical port = socket)
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0
1
2
3
4
Logics
Ports
Transport of information with RTP and RTCP
Audio
G.711
G.723.1
G.726
G.728
G.729
Video
H.261
H.263
Data
T.120
Control and management
of the calls
H.225
RAS
H.245
H.225
of control
Q.931
Signaling signaling
RTP/RTCP
TCP
UDP
IP
Connection (IEEE 802.3)
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H.323v2
New functionalities added to the Gatekeeper
– Security, authentification, encryption
– Establishment of faster call (Fast call setup)
QoS Improvement, thanks to RSVP
Additional services
– Call transfer
– Call forwarding
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Set-up of a H.323v1 call
Terminal 1 H.323
Gatekeeper
Terminal 2 H.323
RAS
H.225
H.245
Logical channel for flows of information
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Set-up of a H.323v2 call
Terminal 1 H.323
Gatekeeper
Terminal 2 H.323
RAS
H.225
Logical channel for flows of information
Faststart OLC: Faster establishment of call with direct
opening of the logical channel
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H.323v2: network management
The Gatekeeper can provide centralized management.
Station management
for the network
SNMP/
CMIP
Gatekeeper
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Gateway
MCU and others
H.323 v3
Signaling (H.225, H.245) using UDP rather than TCP
Address resolution for inter and intra domains
Addition of supplemental services
– Call on hold
– Call park and pickup
– Message Waiting Signaling
– Call waiting
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H.323 v4
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Possibility for the terminal to select a gateway
H.323 URL (h323:[email protected])
Identification of the caller ID
Quiz
Associate the following functions with the suitable
component:
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Support multimedia applications
Translation of the telephone number to IP address
Conversion of the transmission format
Admission control
Conversion of the signaling protocols
Support multimedia conferences
a) Terminal
b) Gateway
c) Gatekeeper
d) MCU
___
___
___
___
___
___
Quiz
Associate the following functions the suitable protocol:
Establish connection between two terminals
Control registration
Opening of the logical channels (video, voice)
Admission control
Functionalities exchange
Termination of connection
a) H.225 (RAS)
b) H.225 (Q.931)
c) H.245
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___
___
___
___
___
___
Questions?
?
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Activity 4.4 SIP – Session Initiation Protocol
SIP Standard
To describe the SIP standard, addressing and components
To explain the various stages carried out during the call set-up in a
SIP network
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Session Initiation Protocol (SIP)
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Signaling protocol for multimedia applications
Independent of sub layer protocols (TCP, UDP)
Standard developed by the IETF (MMUSIC working group) - RFC 2543
SIP works in various phases of the call
– Localization of the corresponding terminal
– Analyze recipient profile and resources
– Negotiation of the media type and of the communication parameters
– Availability of the correspondent
– Call set-up and call follow-up
SIP uses several existing protocols
– Message format (HTTP 1.1)
– Media negotiation (SDP - Session Description Protocol),
– Media (RTP)
– Name resolution and mobility (DNS and DHCP)
– Applications encoding (MIME)
SIP Specific Protocols
Audio
Video
G.711
G.729
G.723.1
…
H.261
H.263
Signaling
SIP
RTP/RTCP
TCP/UDP
IP
Physical
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SDP
SIP Addressing
SIP Addresses are identified by URL, in the form
user@host
user = name or telephone number
host = domain name or IP addresses
Examples
sip:[email protected]
sip:[email protected]
sip:[email protected]; user=phone
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SIP Components
User Agent
An end user application initiating, receiving and terminating a call
Proxy Server
An application server conveying the requests on behalf of the end
user application
The request is processed and sent to the destination (called person)
or to another server
Redirect Server
An application server determining the destination address (To:)
and returning it to the end user application
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SIP Components (cont'd)
Localization Server
Used by the Proxy Server and Redirect Server to obtain the location of
the called user (one or more addresses)
Registration Server
Accept registration requests from the client applications
Generally, the service is offered by the Proxy Server or Redirect Server
DNS Server
Used to locate the Proxy Server or Redirect Server
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SIP components and services
SIP Servers and services
Registrar
Redirect
Locate
Where this name is
or tel. number…
Location
Database
Proxy
SIP Server
Register
I am here
SIP User
Agents
Redirect
Here is the address
Proxy INVITES
I will call it
for you.
INVITE
I want to speak
with another agent.
SIP User
Agents
GW SIP
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SDP - Session Description Protocol
SDP defines the conversation parameters on the client application
(User Agent)
SDP transmits information required to establish a multimedia
session
SDP is similar to H.245 in H.323 functions
SDP contains the following parameters:
– Medium to be used (codec, sampling rate)
– Destination (IP address and port number)
– Session name
– Session duration
– Contact
– etc…
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Example: INVITE
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060
Call-ID: [email protected]
From: sip: [email protected]
To: sip:[email protected]
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v = (protocol version)
O = (owner/creator and session to identify)
C = (session information)
T = (time the session is active)
m = (media name and address transport)
SIP session set-up
Each end knows
the other one IP address
INVITE
Signaling
100 Trying
180 Ringing
200 OK
ACK
Logical opening of RTP channel
Logical opening of RTCP channel
Contents
Logical opening of RTP channel
Logical opening of RTCP channel
Bye
Signaling
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200 OK
Media (UDP)
SDP Messages in a SIP session
Marie
192.168.1.20
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060
Call-ID: [email protected]
From: sip: [email protected]
To: sip:[email protected]
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v=0
o=marie 3123 121231 IN IP4 192.190.132.20
c=IN IP4 192.190.132.20
m=audio 5004 RTP/AVP 0
Each end knows
the other one IP address
INVITE
100 Trying
180 Ringing
200 OK
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.190.132.20:5060
Call-ID: [email protected]
From: sip: [email protected]
To: sip:[email protected]
Cseq 1 ACK
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Pierre
192.168.1.31
ACK
SIP/2.0 200 OK
Call-ID: [email protected]
From: sip: [email protected]
To: sip:[email protected]
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v=0
o=pierre 5664 456456 IP IP4 192.190.132.31
c=IN IP4 192.190.132.31
m=audio5004 RTP/AVP 0
SIP message types
SIP is modeled on HTTP
Use same syntax and semantics as HTTP
– Request
Method (INVITE, ACK, BYE, etc.)
Header (Accept, Contact, etc.)
– Answer
Status code (200 OK, 180 Ringing, etc.)
Header (Content-type, Content-encoding, etc.)
SIP Methods
INVITE
SIP Answers
Initiate a call by inviting a user to take part in a session.
1xx - Informational Messages.
ACK
Confirm that the client received a final response
2xx - Successful Responses.
to a request INVITES.
3xx - Redirection Responses.
BYE
Indicate the end of the call.
CANCEL
Cancel a request.
4xx - Request Failure Responses.
5xx - Server Failure Responses.
6xx - Global Failure Responses.
REGISTER To register the User Agent.
OPTIONS Used to know the capacities of the server.
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SIP in Proxy mode
Location
Server
2
Pierre?
3
INVITE [email protected]
[email protected]
4 INVITE [email protected]
1 From: [email protected]
From: [email protected]
6 200 OK
5 200 OK
7 ACK
[email protected]
8 ACK
Established session
Proxy
Server
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[email protected]
SIP in Redirect mode
Location
server
2
Pierre?
3
[email protected]
INVITE [email protected]
1 From: [email protected]
4
302 Moved
Contact: [email protected]
Redirect
Server
5 ACK
[email protected]
6 INVITE [email protected]
From: [email protected]
7 200 OK
8 ACK
[email protected]
Established session
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SIP call example
Call forward busy from B to C
Proxy Server
UA A
INVITE
100 Trying
UA B
INVITE
486 Busy
ACK
INVITE
180 Ringing
180 Ringing
200 OK
200 OK
ACK
ACK
Established session
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UA C
SIP call example
Call transfer from A to C
Proxy Server
UA B
UA A
UA C
Established session
Bye (also C)
200 OK
Bye (also C)
200 OK
INVITE (req A)
INVITE (req A)
100 Trying
180 Ringing
180 Ringing
200 OK
200 OK
ACK
ACK
Established session
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SIP References
Columbia university Web site
http://www.cs.columbia.edu/sip/
IETF SIP working group
http://ietf.org/html.charters/sip-charter.html
SIP forum
http://www.sipforum.org
Ubiquity Information Center : SIP center
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http://www.sipcenter.com
Laboratory activity 4.4
Analysis of the SIP protocol.
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Activity 4.5
MGCP (Media Gateway Control Protocol) and
Megaco (MEdia GAteway COntrol)
To describe the MGCP/Megaco standard and its components
To explain the various stages carried out during the call setup in a MGCP/Megaco network
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To compare H.323, SIP and MGCP/Megaco standards
New voice network architecture
Separation of the three architecture levels
Open Service
Application Layer
TDM/
Switch Circuit
Standard Interface
Switching Network
Line
Concentration
DIGITAL Trunk
Subsystem
Open Call Control Layer
Administration
Maintenance
Billing
Call Control
Connection Control
Features
Common Channel
Signaling Complex
Standard Interface
Standard-Based
Packet Infrastructure Layer
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Protocol evolution
October 1999
November 2000
MGCP 1.0
(IETF)
MEGACO
(MGCP+)
(IETF)
SGCP
(IETF)
July 1998
Bellcore
Cisco
IPDC
MDCP
(IETF)
August 1998
Level 3
December 1998
Lucent
SGCP - Simple Gateway Control Protocol
MDCP - Media Device Control Protocol
MGCP - Media Gateway Control Protocol
MEGACO - Media Gateway Control
IPDC - IP Device Control
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MGCP (Media Gateway Control Protocol)
Defined by the IETF in document RFC 2705 (version 1.0), in October 1999
Version 0.1 (October 98) is the result of the fusion of SGCP 1.2 (Telcordia)
with IPDC (Level 3)
Media Gateways (MGs) are controlled by the Media Gateway Controllers
(MGCs) in an master/slave architecture
–
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Voice call set-up only (no multimedia)
MGCP architecture is divided into three layers
–
Application (optional): Applications Server
–
Call control : Media Gateway Controller, Call Agent
–
Connectivity: Media Gateways, Routers, LAN, Switches
Uses the session description protocol (SDP) to describe the media
capabilities— like SIP and MEGACO/H.248
Can be deployed in a network in combination with other architectures
(H.323, SIP)
MGCP structure and protocols
SS7
Network
SS7
Network
MGC
SCTP
MGC
MGCP
MGCP
Media
PSTN
Media
gateway
PSTN
Media
gateway
SCTP: Stream Control Transmission Protocol
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SCTP
H.323, SIP, ISUP
© IITelecom,
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MGCP Components
Media Gateway (MG)
Various types: residential, access, trunking
Adapts the content format coming from a network type to another
network type format
Must be able to convert the audio into full-duplex
Media Gateway Controller (MGC)
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Also called Call Agent or Softswitch
Provides a centralized control for the gateways
Responsible for call signaling (set-up, modify, terminate)
Uses UDP
MGCP Primitives
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NotificationRequest (RQNT)
– Inform the gateway to supervise specific events
Notify (NTFY)
– Inform the MGC when the required events take place
CreateConnection (CRCX)
– Create a connection towards an Endpoint inside the gateway
ModifyConnection (MDCX)
– Change the parameters associated with an already established
connection
DeleteConnection (DLCX)
– Remove an existing connection — an ACK returns the call statistics
AuditEndPoint (AUEP) and AuditConnection (AUCX)
– Check an endpoint status and any associated connection
RestartIngProgresss (RSIP)
– Inform the MGC that an endpoint (or a group of endpoints) is out-ofservice
MGCP call set-up
MGCP
MGC
STP
Q.931/SS7
RTP
SS7 Links
User A
Residential
gateway
IP Network
Voice circuits Switch
Trunking
gateway
User B
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Megaco/H.248
Defined by the IETF and ITU (RFC 3015) in November 2000
– The H.248 recommendation was published in February 2001
Called Megaco by the IETF and H.248 by the ITU
Megaco is a control protocol between the Media Gateway and the
Media Gateway Controller (same as MGCP)
Multimedia applications support
– MGCP supports only voice
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Megaco/H.248 structure
SS7/IP
Gateway
SS7
Network
Sigtran
Gateway Media
Controller
(Softswitch)
Megaco/
H.248
IP network
PSTN
Media
Gateway
Media
Gateway
IP telephone
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Switch
PSTN
MGCP vs. Megaco
MGCP is the first developed
– Some products already on the market
MGCP is simpler
– Support only voice
– Megaco developed for multimedia application
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MGCP is the “de facto” standard
Megaco is the upcoming standard
Summary of VoIP protocols
Current tendencies according to
Insight Research, Jan 2001
H.323 is still the most used
protocol
MGCP is accepted by the
Softswitch Manufacturers
SIP is increasingly popular;
industry sees much interest
there; Windows XP includes
SIP client (Messenger)
2001
2002
2003
2004
H.323
89%
73%
45%
35%
MGCP
10%
15%
20%
14%
Megaco is increasingly
accepted
Megaco
1%
8%
20%
28%
SIP and Megaco are chosen
by 3GPP
SIP
0%
5%
15%
24%
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3GPP: 3rd Generation Partnership Project
SIP vs. H.323 vs. MGCP/Megaco
Service
Call Transfer
Call Redirection
Call standby/on-hold
Conference
Click-to-dial
Call set-up
H.323v1
No
No
No
No
No
6-7 RT
Source
Transport
Encoding
Services
Complexity
SS7 Compatibility
Cost
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H323v2
Yes
Yes
Yes
Yes
Yes
3-4 RT
H.323v3
Yes
Yes
Yes
Yes
Yes
2.5 RT
H.323
ITU
Primarily TCP
Binary ASN.1
Telephony
SIP
low
low
low
SIP
Yes
Yes
Yes
Yes
Yes
1.5 RT
SIP
IETF
Primarily UDP
ASCII Text
Multimedia
H.323
high
low
high
MGCP/Megaco
high
high
moderated
Other Specifications Related To VoIP
PINT (PSTN and Internet Internetworking)
– Allow an IP user to have access to the PSTN network (example:
Click-to-dial)
SPIRITS (Service in the PSTN/IN Requesting Internet Service)
– Allow a user of the PSTN network to have access to IP services
(example: Internet Call Waiting)
ENUM (Telephone Number Mapping)
– Translation of the telephone number in URL or IP addresses
TRIP (Telephony Routing over IP)
– Routing of the VoIP calls
Sigtran (Signaling Transport)
– Transport of SS7 signaling on IP
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References
Megaco from the IETF
– http://www.ietf.org/html.charters/megaco-charter.html
MGCP 1.0 (RFC 2705)
– http://ftp. ietf.org/rfc/rfc2705.txt?number=2705
Documentation on Megaco
– ftp://ftp.isi.edu/in-notes/rfc3015.txt
Archives
– http://standards.nortelnetworks.com/archives/megaco.html
Softswitch Consortium
– http://www.softswitch.org
Mailing list
– [email protected]
subscribe megaco
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Questions?
?
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LU 5 – Implementation considerations
Training objectives:
The participant will be able to:
–
Identify the different critical points to consider when
considering a VoIP implementation
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Corporate network
Number of offices and geographical distribution
Voice transmission
– Number of stations
– Centrex, PBX, Keys System
Obsolescence
Financial amortization
– PSTN Links (bandwidth/costs)
– Inter branches Links (bandwidth/costs)
Data transmission network
– Structure and components
– Internet Links (bandwidth/costs)
– Inter offices Links (bandwidth/costs)
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Needs and required functionalities
Growth
IP network
Fax
External/internal
Call processing
Added value services
Call centers
–
Inbound
–
Outbound
Integration of voice and data
–
Messages Server
Mail
Softswitch
PSTN
1
V
Message centralization
IP network
–
Personal assistant
–
Internet call centers
4
2
3
Personal assistant
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PBX
Calendar
V
Potential solutions
Traditional voice circuit switch system
Voice over IP system
Hybrid configuration
– Site by site migration
– Partial migration/keep existing equipment
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Manufacturer choice
Decision factors
QoS
Reliability and robustness
Supported functionalities and applications
Security
Costs
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QoS
Network scaling
Sampling size
– Packet size
– Bandwidth optimization
Compression type
– Locally
– Tie lines
Priority mechanisms
– Locally
– Tie lines
Effects on other decision factors
– Bandwidth costs and use
– Data transmission quality and effectiveness
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Reliability and robustness
Redundancy
– Power supply unit
– Power over Ethernet
– CPU redundancy
– Fall back on the PSTN
Effect on the costs of the equipment
MTBF/MTTR
– Technology maturity
– Difference in technology
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Effect on the maintenance costs
CPU: Central Processing Unit
MTBF: Mean time Between Failure (average time between breakdowns)
MTTR: Mean time To Repair (Mean repair time)
Supported functionalities and applications
Traditional applications
Technology Maturity
– Call processing functionality
– Call center management
– …
VS.
New applications
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Multimedia/ voice & data integration
Personal assistant
Internet call centers
…
Security/confidentiality
Conventional network
Point-to-point
Circuit switch
–
–
Network security
Additional security necessary
for specific applications
• Encryption
IP network
Broadcast environment
Divided bandwidth
–
–
Security at risk
Security mechanisms
• External
o Firewall
o VPN
o Encryptions
• Internal
o Encryption
–
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Effects on
• costs
• User-friendliness
VPN: Virtual Private Network
© IITelecom,
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Solutions costs
Equipment Cost
Equipment required for the new solution implementation,
considering:
–
–
Protection of the investment and amortization
Upgrade of the existing equipment
• PBX, key systems
• Routers, switches
o QoS Support
o Interworking with the WAN (bandwidth, protocol…)
–
–
Redundancy
Security
Wiring cost
–
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1 cable for the telephone and the PC
• Effect on reliability
Solutions costs (cont’d)
Bandwidth cost
Tie lines / inter offices links
–
Dimensioning voice and data
• Voice QoS
• Effects on data QoS
• Access to the PSTN and long distance calls expenses
Maintenance costs
Service contract
Respect of the MTBF and MTTR
Data and voice network
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Internet use in our private network
Today
Best effort network
–
–
–
No control on the bandwidth
No control on the packet size
No QoS mechanism
Evolution towards different QoS on the Internet
Effects on network architecture and technology
Effects on the price
–
–
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Single tariff
Billing
LU 6 - Applications
Training objectives :
The participant will be able to:
–
Describe various services and applications
offered in VoIP network and to describe the
advantages
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Where can we use Voice over IP?
Today
– Tie line replacement between PBX
– Long distance call “Internet Telephony Provider Service”
– Off Premise Extension (OPX)
– Replacement of key systems by a router “Router Key System”
– IP telephone system for small companies (< 100 users)
Tomorrow
– Call centers accessible by Internet “Virtual call centers”
– Integration of voice and data applications
– Collaboration platform (fax, electronic mail, voice messages)
– Unified messaging
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Voice over IP on an Intranet
Montreal
LAN
Toronto
Router
Router
LAN
DS1
PBX
PBX
Gateway
PSTN
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Gateway
PSTN
Internet Telephony Service Provider (ITSP)
Private IP
network
Router
Router
PSTN
PSTN
Gateway
1-514
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ITSP Montreal
Gateway
ITSP Toronto
1-416
Internet Telephony Service Provider (ITSP)
Private IP
network
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IP-PBX
Main
office
Softswitch
PSTN
V
V
Network
IP
V
V
Regional
office
Telecommuter
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Only one network for voice, data & video
No geographical limit
Integration with other Web applications
Simplified mobility with DHCP
Solution based on the standards vs. solution based
on PBX manufacturer
Regional
office
Centrex IP
CO
Main
office
PSTN
Softswitch
Media
Gateway
V
Network
IP
V
Regional
office
V
Telecommuter
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Same advantages as the IP-PBX
No acquisition cost for the technology
Click-to-dial (CTD)
User Database
CTD (Profiles)
Application Server CTD
2. App checks the coordinates
Of the CTD user (IP addresse)
Softswitch
3. App requests
the CA to establish the call
between No xxx and yyy
1. CTD requires
to call No yyy
Web server
4. CA calls
xxx
5. When xxx answers,
CA calls yyy
Navigator
WWW
6. When yyy answers,
CA establishes the call
between xxx and yyy
Called No yyy
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CTD User
No xxx
Personal assistant
Softswitch
PSTN
1
V
Softphone
IP Network
4
2
IP telephone
3
Personal assistant
Composition of the call number
Forward call to the personal assistant
The personal assistant checks and follows the
rules laid down by the user
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Calendar
V
Call centers and Internet
Internet
Hello, can
I help you?
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Centralization of messages
(fax, email, voice message)
Fax
IP network
PBX
Messages
Server
email
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Questions?
?
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Appendix - Acronym list
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Acronym list
3GPP
Third Generation Partnership Project
CIR
Committed Information Rate
ABR
Available Bit Rate
CM
Cable Modem
ACF
Admission Confirm
CMIP
Common Management Information Protocol
ACM
Address Complete Message
CMTS
Cable Modem Termination System
ADPCM
Adaptive Differential Pulses Code Modulation
CO
Central Office
ADSM
Asymmetric Digital Subscriber Line
CoPS
Common open Policy Server
AF
Assured Forwarding
CoS
Class of Service
ALI
Automatic Location Identifier
CPL
Common Programming Language
ANI
Automatic Number Identifier
CPU
Central Processing Unit
ANM
Answer Message
CRTP
Compressed Real-Time Protocol
ANSI
American National Standard Institute
CS-ACELP
Conjugate Structure Adaptive Code Excited Linear Prediction
ARP
Address Resolution Protocol
CSRC
Contributing Source
ARQ
Admission Request
CTD
Click-To-Dial
ASCII
American Standard Code for Information Interchange
CTI
Computer Telephony Integration
ASN
Abstract Symbol Notation
dB
Decibel
ATM
Asynchronous Transfer Mode
dBm
Decibel relative to 1 milliwatt
BGP
Border Gateway Protocol
DHCP
Dynamic Host Configuration Protocol
bps
Bits Per Second
DiffServ
Differentiated Services
CA
Call Agent
DNS
Domain Name Server
CAC
Connection Admission Control
DOCSIS
Data Over Cable Interface Specifications
CAN
Campus Area Network
DS
Differentiated Services
CAR
Committed Access Rate
DSCP
Differentiated Services Code Point
CBWFQ
Class-Based Weighted Fair Queuing
EF
Expedited Forwarding
CGI
Common Gateway Interface
EGP
Exterior Gateway Protocol
CIC
Circuit Identification Code
EIGRP
Enhanced Interior Gateway Routing Protocol
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Acronym list (cont'd)
ERP
Enterprise Resource Planning
ISDN
Integrated Services Digital Network
ETSI
European Telecomm Standards Institute
ISUP
Integrated Services Digital Network User Part
FCS
Frame Check Sequence
ITSP
Internet Telephony Service Provider
FEC
Forward Equivalence Class
ITU
International Telecommunication Union
FRF
Frame Relay Forum
ITU-T
ITU - Telecom
FRTS
Frame Relay Traffic Shaping
JPEG
Joint Photographic Expert Group
FTP
File Transfer Protocol
kbps
Kilobits Per Second
GK
Gatekeeper
LAN
Local Area Network
GTS
Generic Traffic Shaping
LDP
Label Distribution Protocol
HFC
Hybrid Fiber Coax
LD-CELP
Low-Delay Code Excited Linear Prediction
HTTP
Hypertext Transfer Protocol
LER
Label Edge Router
Hz
Hertz
LIB
Label Information Base
IAM
Initial Address Message
LLC
Logical Link Control
ICMP
Internet Control Message Protocol
LLQ
Low Latency Queuing
IDC
International Dated Corporation
LS
Location Server
IEEE
Institute of Electrical and Electronic Engineers
LSR
Label Switch Router
IETF
Internet Engineering Task Force
MAC
Media Access Control
IGP
Interior Gateway Protocol
MAN
Metropolitan Area Network
IHL
Internet Header Length
MAP
Mobile Application Part
INAP
Intelligent Network Application Profile
MCR
Minimum Cell Rate
IntServ
Integrated Services
MCU
Multipoint Control Unit
IP
Internet Protocol
MDCP
Media Device Control Protocol
IPv4
IP version 4
Megaco
Media Gateway Control
IPv6
IP version 6
MG
Media Gateway
IPDC
IP Device Control
MGC
Media Gateway Controller
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Acronym list (cont'd)
MGCP
Media Gateway Control protocol
PCM
Pulse Code Modulation
MIB
Management Information Base
PHB
Per-Hop Behavior
MIME
Multipurpose Internet Mail Extension
POP3
Post Office Protocol version 3
MIPS
Million Instruction Per Second
POTS
Plain Old Telephone Service
MMUSIC
Multiparty Multimedia Session Control
PPP
Point-to-Point Protocol
MOS
Mean Opinion Score
PQ
Priority Queuing
MPLS
Multi Protocol Label Switching
PS
Proxy Server
MP-MLQ
Multipulse Multilevel Quantization
PSTN
Public Switched Telephone Network
ms
Millisecond
QoS
Quality of Service
MTBF
Mean Time Between Failure
RARP
Reverse ARP
MTP
Message Transfer Part
RAS
Registration Admission Status
MTTR
Mean Time To Repair
RFC
Request For Comment
NAT
Network Address Translation
RMON
Remote Monitoring
NCS
Network-based Control Signaling
RPC
Remote Procedure Call
NFS
Network File System
RS
Registration Server
NNTP
Network News Transfer Protocol
RSVP
Resource Reservation Protocol
OLC
Open Logical Channel
RTCP
Real-time Control Protocol
OMAP
Operational, Management and Admin Process
RTP
Real-time Protocol
OPX
Off Premise Extension
SCCP
Signalling Connection Control Part
OSPF
Open Shortest Path First
SCP
Signal Control Point
OSI
Open System Interconnection
SCTP
Stream Control Transmission Protocol
PAN
Personal Area Network
SDH
Synchronous Digital Hierarchy
PBH
Per-Hop Behavior
SDP
Session Description Protocol
PBX
Private Branch Exchange
SGCP
Simple Gateway Control Protocol
PCR
Peak Cell Rate
SIGTRAN
Signalling Transport
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Acronym list (cont'd)
SIP
Session Initiation Protocol
VPN
Virtual Private Network
SLA
Service Level Agreement
WAN
Wide Area Network
SLIP
Serial Line Internet Protocol
WFQ
Weighted Fair Queuing
SMTP
Simple Mail Transfer Protocol
WRED
Weighted Random Early Drop
SNMP
Simple Network Management Protocol
SONET
Synchronous Optical Network
For additional acronyms,
SS7
Signaling System 7
http://www.csrstds.com
SSP
Switching Service Point
SSRC
Synchronisation Source
STHML
Safe Hypertext Transfer Protocol
STP
Signal Transfer Point
SVC
Switched Virtual Circuit
TCAP
Transaction Capabilities Application Part
TCP
Transport Control Protocol
TDM
Time Division Multiplexing
TFTP
Trivial File Transfer Protocol
TIA
Telecommunication Industry Association
ToS
Type of Service
TTL
Time To Live
UA
User Agent
UDP
User Datagram Protocol
URL
Uniform Resource Locator
VAD
Voice Activation Detection
VBR-rt
Variable Bit Rate - Real-time
VoIP
Voice over IP
193
IIT
101
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