IP Telephony (Voice over IP)
Instructor
Ai-Chun Pang, acpang@csie.ntu.edu.tw
Office Number: 417, New building of CSIE
Textbook
“Carrier Grade Voice over IP,” D. Collins, McGraw-Hill, Second Edition, 2003.
Requirements
Homework x 2 (Homework I and Homework II) 30%
Mid-term exam 15%
Final exam 20%
Term project (or Homework III) 20%
Oral presentation 15%
TAs (office number: 442, New building of CSIE)
黃思瑋 (gestalt@voip.csie.ntu.edu.tw)
林俊仁 (jrlin@voip.csie.ntu.edu.tw)
Course Outline
Introduction
Transporting Voice by Using IP - RTP (Real-Time Transport Protocol)/RTCP (RTP Control Protocol)
H.323
Session Initiation Protocol (SIP) and ENUM
VoIP over Network Address Translation (NAT)
Skype – Voice over Overlay Networks
Media Gateway Control and the Softswitch Architecture
VoIP and SS7
Quality of Service
Designing a Voice over IP Network Mobile All IP Network
4 IP Telephony
Next Generation Networks [1/2]
Internet Telecom & Wireless Communication
Reference: CCL/ITRI Wireless
LAN WLAN
FA CA
ProxySIP Server
Internet
MS
CSCF3GPP
3G UMTS
GGSN SGSN
T-SGW
MGW
MGW PSTN
MGCF3GPP
Next Generation Networks [2/2]
Internet Telecom & Wireless Communication
IP
MGCF CSCF
T-SGW MGW MGW
WLAN GPRS
CSCF SIP
Server
PSTN
Internet
Wireless App.
Server 3rd Parties App.
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
Interoperability
8 IP Telephony
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
Next generation Telcos
Access and bandwidth are better managed.
QoS solutions
Service-level agreements between providers
IP
A packet-based protocol
Routing on a packet-by-packet base
Packet transfer with no guarantees
May not be received 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
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, lower operating
expense, integration of voice and data applications, potentially lower bandwidth requirements, the
widespread availability of IP
12 IP Telephony
Lower Equipment Cost
PSTN switch
Proprietary – hardware, OS, applications
New software application development for third parties
High operation and management cost
Training, support, and feature development
Mainframe computer
The IP world
Standard mass-produced computer equipment
Application software is quite separate
A horizontal business model
More open and competition-friendly
Intelligent Network (IN)
does not match the openness and flexibility of IP solutions.
A few highly successful services
VoIP networks can interwork with Signaling System 7 (SS7) and take advantage of IN services build on SS7.
Voice/Data Integration
Click-to-talk application
Personal communication
E-commerce
Web collaboration
Shop on-line with a friend at another location
Video conferencing
Shared whiteboard session
With IP multicasting
IP-based PBX
IP-based call centers
IP-based voice mail
Far more feature-rich than the standard 12- button keypad
14 IP Telephony
Lower Bandwidth Requirements
PSTN
G.711 - 64 kbps
Human speech frequency < 4K Hz
The Nyquist Theorem: 8000 samples per second to fully capture the signal
8K * 8 bits
Sophisticated coders
32kbps, 16kbps, 8kbps, 6.3kbps, 5.3kbps
GSM – 13kbps
Save more bandwidth by silence suppression
Traditional telephony networks can use coders, too.
But it is more difficult.
VoIP – two ends of the call to negotiate the coding scheme
The fundamental architecture of VoIP systems lends itself to more transmission-efficient network designs.
Distributed (Bearer traffic can be routed more directly from source to destination.)
The Widespread Availability of IP
IP
LANs and WANs
Dial-up Internet access
IP applications even reside within hand-held computers and various wireless devices.
The ubiquitous presence
VoFR or VoATM
Only for the backbone of the carriers
16 IP Telephony
VoIP Challenges
VoIP must offer the same reliability and voice quality as traditional circuit-switched
telephony.
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.
Speech Quality [1/2]
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.
18 IP Telephony
Speech Quality [2/2]
Echo
High Delay ===> Echo is Critical
Packet Loss
Traditional retransmission cannot meet the real-time requirements
Call Set-up Time
Address Translation
Directory Access
Managing Access and Prioritizing Traffic
A single network for a wide range of
applications, including data, voice, and video
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 a huge effort.
20 IP Telephony
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
MOS values are still subjective in nature.
Network Reliability and Scalability
PSTN system fails
99.999% reliability
Today’s VoIP solutions
Redundancy and load sharing
A balance must be struck between network cost and network quality.
Finding the right balance is the responsibility of the network architect.
Scalable – easy to start on a small scale and then expand as traffic demand increases
22 IP Telephony
VoIP Implementations
IP-based PBX solutions
A single network
Enhanced services
VoIP Implementations
IP voice mail
One of the easiest applications
IP call centers
Use the caller ID
Automatic call distribution
Load the customer’s information on the agent’s desktop
Click to talk
24 IP Telephony
