IP Telephony
Instructor
Ai-Chun Pang, [email protected]
Office Number: 417
Textbook
“Carrier Grade Voice over IP,” D. Collins, McGraw-Hill,
Second Edition, 2003.
Requirements
Homework x 3 30%
One mid-term exam (5/14) 40%
One term project (proposal: 5/7) 30%
Presentation ([5/28], 6/11 and 6/18), Demo (6/18)
TAs (office number: 213)
黃宇傑, [email protected]
劉志孝, [email protected]
Course Outline
Introduction
Transporting Voice by Using IP (Real-time Transport
Protocol - RTP)
Speech-Coding Techniques
H.323
Session Initiation Protocol (SIP) and ENUM
Media Gateway Control and the Softswitch Architecture
VoIP and SS7
Quality of Service
Designing a Voice over IP Network
Mobile IPv4, IPv6 and Micro-mobility
Wireless All IP Network
Mobile Number Portability
Introduction
Chapter 1
Carrier Grade VoIP
Carrier grade and VoIP
mutually exclusive
A serious alternative for voice communications with enhanced
features
Carrier grade
The last time when it fails
99.999% reliability (high reliability)
Fully redundant, Self-healing
AT&T carries about 300 million voice calls a day (high capacity).
Highly scalable
Short call setup time, high speech quality
No perceptible echo, noticeable delay and annoying noises on the
line
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VoIP
Transport voice traffic using the Internet
Protocol (IP)
One of the greatest challenges to VoIP is
voice quality.
One of the keys to acceptable voice quality is
bandwidth.
Control and prioritize the access
Internet: best-effort transfer
VoIP != Internet telephony
The next generation Telcos
Access and bandwidth are better managed.
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IP
A packet-based protocol
Routing on a packet-by-packet base
Packet transfer with no guarantees
May not receive in order
May be lost or severely delayed
TCP/IP
Retransmission
Assemble the packets in order
Congestion control
Useful for file-transfers and e-mail
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Data and Voice
Data traffic
Asynchronous – can be delayed
Extremely error sensitive
Voice traffic
Synchronous – the stringent delay requirements
More tolerant for errors
IP is not for voice delivery.
VoIP must
Meet all the requirements for traditional telephony
Offer new and attractive capabilities at a lower cost
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Why VoIP?
Why carry voice?
Internet supports instant access to anything
However, voice services provide more revenues.
Voice is still the killer application.
Why use IP for voice?
Traditional telephony carriers use circuit switching
for carrying voice traffic.
Circuit-switching is not suitable for multimedia
communications.
IP: lower equipment cost, integration of voice and
data applications, potentially lower bandwidth
requirements, the widespread availability of IP
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Lower Equipment Cost
PSTN switch
Proprietary – hardware, OS, applications
High operation and management cost
Training, support and feature development cost
Mainframe computer
The IP world
Standard hardware and mass-produced
Application software is quite separate
A horizontal business model
More open and competition-friendly
IN
does not match the openness and flexibility of IP.
A few highly successful services
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Voice/Data Integration
Click-to-talk application
Personal communication
E-commerce
Web collaboration
Shop on-line with a fried at another location
Video conferencing
IP-based PBX
IP-based call centers
IP-based voice mail
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Lower Bandwidth Requirements
PSTN
G.711 - 64 kbps
Human speech frequency < 4K Hz
The Nyquist Theorem: 8000 samples per second
8K * 8 bits
Sophisticated coders
32kbps, 16kbps, 8kbps, 6.3kbps, 5.3kbps
GSM – 13kbps
Save more bandwidth by silence-detection
Traditional telephony networks can use coders,
too.
But it is more difficult.
VoIP – two ends of the call negotiate the coding
scheme
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The Widespread Availability of IP
IP
LANs and WANs
Dial-up Internet access
The ubiquitous presence
VoFR or VoATM
Only for the backbone of the carriers
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VoIP Challenges
VoIP must offer the same reliability and voice
quality as PSTN.
Mean Opinion Score (MOS)
5 (Excellent), 4 (Good), 3 (Fair), 2 (Poor), 1 (Bad)
International Telecommunication Union
Telecommunications Standardization Sector (ITU-
T) P.800
Toll quality means a MOS of 4.0 or better.
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Speech Quality
Must be as good as PSTN
Delay
The round-trip delay
Coding/Decoding + Buffering Time + Tx. Time
G.114 < 300 ms
Jitter
Delay variation
Different routes or queuing times
Adjusting to the jitter is difficult
Jitter buffers add delay
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Speech Quality
Echo
High Delay ===> Echo is Critical
Packet Loss
Traditional retransmission cannot meet the
real-time requirements
Call Set-up Time
Address Translation
Directory Access
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Managing Access and Prioritizing Traffic
A single network for a wide range of
applications
Call is admitted if sufficient resources are
available
Different types of traffic are handled in different
ways
If a network becomes heavily loaded, e-mail traffic
should feel the effects before synchronous traffic
(such as voice).
QoS has required huge efforts
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Speech-coding Techniques
In general, coding techniques are such that
speech quality degrades as bandwidth reduces.
The relationship is not linear.
G.711 64kbps 4.3
G.726 32kbps 4.0
G.723 (celp) 6.3kbps 3.8
G.728 16kbps 3.9
G.729 8kbps 4.0
GSM 13kbps 3.7
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Network Reliability and Scalability
PSTN system fails
99.999% reliability
Today’s VoIP solutions
Redundancy and load sharing
Scalable – easy to start on a small scale and then
expand as traffic demand increases
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VoIP Implementations
IP-based PBX solutions
A single network
Enhanced services
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VoIP Implementations
IP voice mail
One of the easiest
applications
IP call centers CTI
Server
Use the caller ID Internet
Automatic call distribution
Load the customer’s
information on the agent’s
desktop PBX/ACD ITG Web Server
Call Center
Click to talk
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VoIP Evolution
IP
IP Network
Network
VoIP
VoIP VoIP Terminal
Terminal Terminal Gateway PSTN
1: PC to PC 2: Phone to PC over IP
IP
Network PSTN
Gateway Gateway
Gateway Gateway
PSTN PSTN IP IP
Network Network
VoIP VoIP
Terminal Terminal
3: Phone to Phone over IP 4: PC to PC over PSTN
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