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Introduction to VoIP. Lecture 1 Paul Flynn. VoIP Part 1 – Sept - Dec. Introduction Transporting Voice over IP Network Speech Coding Techniques Signalling 1 – H323 Signalling 2 – SIP Media Gateway Control VoIP and SS7 ATM QOS and Traffic management Designing a VoIP Network. SF. RTP. - PowerPoint PPT Presentation
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Introduction to VoIPIntroduction to VoIP
Lecture 1
Paul Flynn
VoIP Part 1 – Sept - DecVoIP Part 1 – Sept - Dec
1. Introduction2. Transporting Voice over IP Network3. Speech Coding Techniques4. Signalling 1 – H3235. Signalling 2 – SIP6. Media Gateway Control7. VoIP and SS78. ATM9. QOS and Traffic management10.Designing a VoIP Network
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Network ComponentsNetwork Components
CO - Central OfficeTrunk - Switch-switch connectionLoop - Line from switch to phoneTandem switch - provides switch-switch interconnectionIXC - interexchange carrierPBX - Private branch exchange
PBX Switch
Switch
Switch
Switch
Switch
CO
IXCSJ
SF
RTP
4
SSP
SSP
SSPSTP
STP
SCP
•SSP: Service Switching Point (Telephone Switch)
•STP: Signaling Transfer Point (Router)
•SCP: Service Control Point (Database, Logic)
Trunk
Signaling(Packet)
Trunk
Trunk
SS7Voice
The PSTN: Separate Voice The PSTN: Separate Voice and Signaling Networksand Signaling Networks
(TDM)
Local LoopLocal Loop
• 2 wire from phone to switch
• Tip and Ring - derived from old switchboard plugs
• 4 wire used at switch
• Conversion performed by hybrid
2 wire
2 wire
2 wire 2 wire
Switch
switch
Speaker Listener
Talker Echo
Local Loop (cont.)Local Loop (cont.)Problems with Analog TransmissionProblems with Analog Transmission
• 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
2 wire
2 wire
2 wire 2 wire
Hybrid
Hybrid
Speaker Listener
Talker Echo
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Digitizing VoiceDigitizing 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
Voice ENCODER
Low Pass FilterBW = Fmax
Low Pass FilterBW = Fmax
BinaryEncoderBinary
EncoderClockClock
Pulse Detector
Pulse Detector
Binary to Decimal Decoder
Binary to Decimal Decoder
FilterBW = Fmax
FilterBW = Fmax
Voice DeCODER
Sampler2 * Fmax Samples/Sec
Sampler2 * Fmax Samples/Sec
Quantizern Bits/Sample2n Levels
Quantizern Bits/Sample2n Levels
Waveform Coders (codec)Waveform Coders (codec)
Non- Linear Encoding
Closely Follows Human Voice Characteristics
High Amplitude Signals Have More Quantization Distortion
(Both a- & - Law)
Input
Output
Linear Encoding
Relatively Easy to Analyze, Synthesize, and Regenerate
All Amplitudes Have Roughly Equal Quantization Distortion
Input
Output
Non-Linear vs. Linear EncodingNon-Linear vs. Linear EncodingCompanding (a-law vs Companding (a-law vs -law)-law)
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00010010001101000101011110001001101010111100110111101111
00010010001101000101011110001001101010111100110111101111
Linear Predictive CodingLinear Predictive CodingSource CodingSource Coding
00010010001101000101011110001001101010111100110111101111
00010010001101000101011110001001101010111100110111101111
Actual Code Predicted Code
1001 1011
10
20 ms
Bandwidth RequirementsBandwidth Requirements
Voice Band Traffic
Encoding/Encoding/CompressionCompression
ResultResultBit RateBit Rate
G.711 PCMG.711 PCMA-Law/A-Law/uu-Law-Law
64 kbps (DS0)64 kbps (DS0)
G.726 ADPCMG.726 ADPCM 16, 24, 32, 40 kbps16, 24, 32, 40 kbps
G.729 CS-ACELPG.729 CS-ACELP 8 kbps8 kbps
G.728 LD-CELPG.728 LD-CELP 16 kbps16 kbps
G.723.1 CELPG.723.1 CELP 6.3/5.3 kbps6.3/5.3 kbpsVariableVariable
Voice QualityVoice Quality
Compression MethodCompression Method MOS ScoreMOS Score DelayDelay(msec)(msec)
64K PCM (G.711)64K PCM (G.711) 4.44.4 0.750.75
32K ADPCM (G.726)32K ADPCM (G.726) 4.24.2 11
16K LD-CELP (G.728)16K LD-CELP (G.728)
8K CS-ACELP (G.729)8K CS-ACELP (G.729) 4.24.2 1515
8K CS-ACELP (G.729a)8K CS-ACELP (G.729a) 1515
3–53–54.24.2
3.63.6
Anything Above an MOS of 4.0 Is “Toll” Quality
Voice Activity DetectionVoice Activity Detection
Voice “Spurt” Silence
Pink Noise
Time
Voice Activity(PowerLevel) SID Buffer SID
Hang Timer No Voice Traffic Sent
B/W Saved
- 54 dbm
- 31 dbm
Voice “Spurt”
Rensselaer Polytechnic Institute
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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)
Different Types of SignalingDifferent Types of Signaling(when you place a call)(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 routingDTMF, MF, DNIS
• Informational - indicates call progressBusy signal, dial tone, ring back
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Summary PageSummary Page
PBX Switch
Switch
Switch
Switch
Switch
CO
IXCSJ
SF
RTP
T1/ E1DTMF/ MFCAS/ CCS
Local LoopFXS/ FXOLoopstart/Gndstart
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Voice Transport ProtocolsVoice Transport Protocols
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Voice Transport ProtocolVoice Transport ProtocolOverviewOverview
PSTN
PBX
ATM, FR, HDLC
IP
CiscoGateway
CiscoGateway
T1/E1CAS/CCS
Encoder/Decoder
QueuingQueuing
• Voice always given priority over data
• Real-time queue for voice and videoData 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
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Voice Over IPVoice Over IP
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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
Rensselaer Polytechnic Institute
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VoIP Camps
ISDN LAN conferencin
g
IP
H.323
I-multimediaWWW
IP
SIP
Call AgentSIP & H.323
IP
“Softswitch” BISDN, AIN
H.xxx, SIP
“any packet”
BICC
Conferencing Industry
Netheads“IP over
Everything”
Circuit switch
engineers “We over
IP”
“Convergence” ITU
standards
Our focus
Rensselaer Polytechnic Institute
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Are true Internet hosts
• Choice of application
• Choice of server
• IP appliances
Implementations
• 3Com (3)
• Columbia University
• MIC WorldCom (1)
• Mediatrix (1)
• Nortel (4)
• Siemens (5)
4
IP SIP Phones and Adaptors
1
3
Analog phone adaptor
Palmcontrol
2
54
Rensselaer Polytechnic Institute
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PSTN to IP Call
PBXPSTN
External T1/CAS
Regular phone(internal)
Call 93971341
SIP server
sipd
Ethernet
3
SQLdatabase
4 7134 => bob
sipc
5
Bob’s phone
GatewayInternal T1/CAS(Ext:7130-7139)
Call 71342
Rensselaer Polytechnic Institute
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IP to PSTN Call
Gateway(10.0.2.3)
3
SQLdatabase
2Use sip:[email protected]
Ethernet
SIP server
sipdsipc
1Bob calls 5551212
PSTN
External T1/CAS
Call 55512125
5551212
PBX
Internal T1/CASCall 85551212 4
Regular phone(internal, 7054)
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End-to-End Delay
Sender Receiver
NetworkTransit Delay
t
AA AA
Network
Last BitReceived
First BitTransmitted
ProcessingDelay
ProcessingDelay
End-to-End Delay
Fixed Delay ComponentsFixed Delay Components
• Propagation—six microseconds per kilometer
• Serialization
• Processing
Coding/compression/decompression/decoding
Packetization
Processing Delay
Propagation Delay
Serialization Delay—Buffer to Serial Link
Variable Delay Components Variable Delay Components
• Queuing delay
• Dejitter buffers
• Variable packet sizes
DejitterBuffer
Queuing Delay
Queuing Delay
Queuing Delay
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Delay Variation—“Jitter”Delay Variation—“Jitter”
t
t
Sender Transmits
Sink Receives
AA BB CC
AA BB CC
D1 D2 = D1
Sender Receiver
D3 = D2D3 = D2
Network
85
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Network QoS ToolkitNetwork QoS Toolkit
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Logical ConnectionsLogical Connections
Call Leg 3
Call Leg 1
IP Cloud
Call Leg 2
Call Leg 4
