Transcript Lecture 1 - IT, Sligo

Introduction to VoIP
Lecture 1
Paul Flynn
VoIP Part 1 – Sept - Dec
1. Introduction
2. Transporting Voice over IP Network
3. Speech Coding Techniques
4. Signalling 1 – H323
5. Signalling 2 – SIP
6. Media Gateway Control
7. VoIP and SS7
8. ATM
9. QOS and Traffic management
10.Designing a VoIP Network
Network Components
SF
Switch
Switch
RTP
SJ
IXC
Switch
Switch
PBX
Switch
CO
CO - Central Office
Trunk - Switch-switch connection
Loop - Line from switch to phone
Tandem switch - provides switch-switch interconnection
IXC - interexchange carrier
PBX - Private branch exchange
3
The PSTN: Separate Voice
and Signaling Networks
•SSP: Service
Switching Point
(Telephone
Switch)
•STP: Signaling
Transfer Point
(Router)
Voice
SSP
STP
Trunk
SSP
SS7
Trunk
Signaling
(Packet)
SCP
(TDM)
Trunk
•SCP: Service
Control Point
(Database, Logic)
SSP
STP
4
Local Loop
Speaker
2 wire
Talker Echo
S
w
i
t
c
h
2 wire
Listener
s
w
i
t
c
h
2 wire
2 wire
• 2 wire from phone to switch
• Tip and Ring - derived from old
switchboard plugs
• 4 wire used at switch
• Conversion performed by hybrid
Local Loop (cont.)
Problems with Analog Transmission
Speaker
2 wire
Talker Echo
H
y
b
r
i
d
2 wire
Listener
H
y
b
r
i
d
2 wire
2 wire
• Several problems with analog
• Attenuation - loss of signal power
• Distortion - unequal loss at different frequencies
• Noise - induced into line which is amplified along with
signal by network components
• Echo - due to 2/4 wire conversion
• Physical impairments - bad lines, bridge taps, load coils
Digitizing Voice
• Assumption is that human speech information is
contained in the range of 300-3400 Hz
Filter & use signal below 4 kHz to prevent aliasing
Sample and quantize signal at 8kHz
encoder produces 64 kbit/sec stream of data
7
Waveform Coders (codec)
Voice ENCODER
Sampler
2 * Fmax
Samples/Sec
Low Pass
Filter
BW = Fmax
Clock
Quantizer
n Bits/Sample
2n Levels
Binary
Encoder
Voice DeCODER
Pulse
Detector
Binary to
Decimal
Decoder
Filter
BW = Fmax
Non-Linear vs. Linear Encoding
Companding (a-law vs -law)
Output
Input
Output
Input
Non- Linear Encoding
Closely Follows Human
Voice Characteristics
High Amplitude Signals Have
More Quantization Distortion
(Both a- & - Law)
Linear Encoding
Relatively Easy to Analyze,
Synthesize, and Regenerate
All Amplitudes Have Roughly
Equal Quantization Distortion
Linear Predictive Coding
Source Coding
10
20 ms
0001
0010
0011
0100
0101
0111
1000
1001
1010
1011
1100
1101
1110
1111
0001
0010
0011
0100
0101
0111
1000
1001
1010
1011
1100
1101
1110
1111
Actual Code
1001
1011
0001
0010
0011
0100
0101
0111
1000
1001
1010
1011
1100
1101
1110
1111
0001
0010
0011
0100
0101
0111
1000
1001
1010
1011
1100
1101
1110
1111
Predicted Code
10
Bandwidth Requirements
Voice Band Traffic
Encoding/
Compression
G.711 PCM
A-Law/u-Law
G.726 ADPCM
Result
Bit Rate
64 kbps (DS0)
16, 24, 32, 40 kbps
G.729 CS-ACELP
8 kbps
G.728 LD-CELP
16 kbps
G.723.1 CELP
6.3/5.3 kbps
Variable
Voice Quality
Anything Above an MOS of 4.0 Is “Toll” Quality
Compression Method
MOS Score
Delay
(msec)
64K PCM (G.711)
4.4
0.75
32K ADPCM (G.726)
4.2
1
16K LD-CELP (G.728)
4.2
3–5
8K CS-ACELP (G.729)
4.2
15
8K CS-ACELP (G.729a)
3.6
15
Voice Activity Detection
- 31 dbm
B/W Saved
Voice
Activity
(Power
Level)
Hang Timer
No Voice
Traffic Sent
SID Buffer
SID
- 54 dbm
Pink Noise
Voice “Spurt”
Time
Silence
Voice “Spurt”
Applications of Speech Coding







Telephony, PBX
Wireless/Cellular Telephony
Internet Telephony
Speech Storage (Automated call-centers)
High-Fidelity recordings/voice
Speech Analysis/Synthesis
Text-to-speech (machine generated speech)
Rensselaer Polytechnic Institute
14
Different Types of Signaling
(when you place a call)
• Supervisory - Determines state of line/trunk whether
on/off-hook
EM signal leads, loop open/closed
• Addressing - passes digit information for call routing
DTMF, MF, DNIS
• Informational - indicates call progress
Busy signal, dial tone, ring back
Summary Page
Local Loop
FXS/ FXO
Loopstart/
Gndstart
SF
Switch
Switch
RTP
SJ
IXC
Switch
Switch
PBX
Switch
CO
T1/ E1
DTMF/ MF
CAS/ CCS
16
Voice Transport
Protocols
17
Voice Transport Protocol
Overview
Encoder/
Decoder
IP
Cisco
Gateway
T1/E1
CAS/CCS
ATM, FR, HDLC
Cisco
Gateway
PBX
PSTN
18
Queuing
• Voice always given priority over data
• Real-time queue for voice and video
Data queue serviced only if nothing
in Real Time queue - (Exhaustive like priority queuing)
• Non-real time queue (Data)
WFQ by default
WFQ Disabled if Frame Relay Traffic Shaping Enabled
Fancy queuing disabled if voice-encap set on interface
Voice Over IP
20
Protocols Used
• H.225.0 for Connection and Status
– Q.931 ‘derived’ messages
– ‘RAS’ for Endpoint-GK signaling.
• H.245 for negotiating channel usage and
capabilities
• Media transport
– RTP/RTCP -- standard payloads
(RFC1889/1890)
– ‘native’ uni/multicast support
21
VoIP Camps
Conferencing
Industry
Netheads
“IP over
Everything”
Circuit switch
engineers
“We over IP”
“Convergence”
ITU standards
H.323
SIP
“Softswitch”
BICC
ISDN LAN
conferencing
I-multimedia
WWW
Call Agent
SIP & H.323
BISDN, AIN
H.xxx, SIP
IP
IP
IP
“any packet”
Rensselaer Polytechnic Institute
Our focus
22
IP SIP Phones and Adaptors
1
Are true Internet hosts
• Choice of
application
Analog phone adaptor
• Choice of server 2
• IP appliances
Implementations
3
• 3Com (3)
• Columbia
University
Palm
control
• MIC WorldCom (1)
• Mediatrix (1)
• Nortel (4)
44
5
• Siemens (5)
Rensselaer Polytechnic Institute
23
PSTN to IP Call
PBX
PSTN
External T1/CAS
1 Call 9397134
Gateway
Internal T1/CAS
(Ext:7130-7139)
2
Call 7134
Ethernet
Regular phone
(internal)
5
3
SIP server
sipc
Bob’s phone
Rensselaer Polytechnic Institute
24
SQL
database
sipd
4
7134 => bob
IP to PSTN Call
PBX
PSTN
External T1/CAS
5 Call 5551212
Gateway
(10.0.2.3)
Internal T1/CAS
4 Call 85551212
3
Ethernet
5551212
Regular phone
(internal, 7054)
1
Bob calls
5551212
SIP server
sipc
2
SQL
database
sipd
Use sip:[email protected]
Rensselaer Polytechnic Institute
25
End-to-End Delay
Sender
Receiver
Network
First Bit
Last Bit
Transmitted Received
A
Processing
Delay
A
Network
Transit Delay
t
Processing
Delay
End-to-End Delay
26
Fixed Delay Components
Propagation Delay
Serialization Delay—
Buffer to Serial Link
Processing Delay
• Propagation—six microseconds per kilometer
• Serialization
• Processing
Coding/compression/decompression/decoding
Packetization
Variable Delay Components
Queuing
Delay
Queuing
Delay
Queuing
Delay
Dejitter
Buffer
• Queuing delay
• Dejitter buffers
• Variable packet sizes
Delay Variation—“Jitter”
Sender
Receiver
Network
B
A
C
Sender Transmits
t
A
D1
B
D2 = D1
C
D3 = D2
Sink Receives
t
29
85
Network QoS Toolkit
30
Logical Connections
Call Leg 1
Call Leg 2
IP Cloud
Call Leg 3
Call Leg 4
31