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VoIP VoIP Lecture 8 Paul Flynn

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VoIP. Lecture 8 Paul Flynn. SF. RTP. SJ. IXC. CO. Network Components. 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. - PowerPoint PPT Presentation

TRANSCRIPT

Page 1: VoIP

VoIPVoIP

Lecture 8

Paul Flynn

Page 2: VoIP

2

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

SwitchSwitch

Switch

CO

IXCSJ

SF

RTP

Page 3: VoIP

3

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)

Page 4: VoIP

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 wireSwitch

switch

Speaker Listener

Talker Echo

Page 5: VoIP

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 wireHybrid

Hybrid

Speaker Listener

Talker Echo

Page 6: VoIP

6

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

Page 7: VoIP

Voice ENCODER

Low Pass FilterBW = Fmax

BinaryEncoderClock

Pulse Detector

Binary to Decimal Decoder

FilterBW = Fmax

Voice DeCODER

Sampler2 * Fmax Samples/Sec

Quantizern Bits/Sample2n Levels

Waveform Coders (codec)Waveform Coders (codec)

Page 8: VoIP

Non- Linear EncodingClosely Follows Human

Voice CharacteristicsHigh Amplitude Signals Have More Quantization Distortion

(Both a- & - Law)

Input

Output

Linear EncodingRelatively 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)

Page 9: VoIP

9

00010010001101000101011110001001101010111100110111101111

00010010001101000101011110001001101010111100110111101111

Linear Predictive CodingLinear Predictive CodingSource CodingSource Coding

00010010001101000101011110001001101010111100110111101111

00010010001101000101011110001001101010111100110111101111Actual Code Predicted Code

1001 1011

10

20 ms

Page 10: VoIP

Bandwidth RequirementsBandwidth Requirements

Voice Band TrafficEncoding/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

Page 11: VoIP

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

Page 12: VoIP

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”

Page 13: VoIP

Rensselaer Polytechnic Institute

13

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)

Page 14: VoIP

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

Page 15: VoIP

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Summary PageSummary Page

PBX Switch

Switch

SwitchSwitch

Switch

CO

IXCSJ

SF

RTP

T1/ E1DTMF/ MFCAS/ CCS

Local LoopFXS/ FXOLoopstart/Gndstart

Page 16: VoIP

16

Voice Transport ProtocolsVoice Transport Protocols

Page 17: VoIP

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Voice Transport ProtocolVoice Transport ProtocolOverviewOverview

PSTN

PBX

ATM, FR, HDLC

IP

CiscoGateway

CiscoGateway

T1/E1CAS/CCS

Encoder/Decoder

Page 18: VoIP

QueuingQueuing

• 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

Page 19: VoIP

19

Page 20: VoIP

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

Page 21: VoIP

Rensselaer Polytechnic Institute

21

VoIP Camps

ISDN LAN conferencin

gIP

H.323I-multimedia

WWW

IP

SIPCall Agent

SIP & 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

Page 22: VoIP

Rensselaer Polytechnic Institute

22

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 Adaptors1

3                 

Analog phone adaptor

Palmcontrol

2

54

Page 23: VoIP

Rensselaer Polytechnic Institute

23

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

Page 24: VoIP

Rensselaer Polytechnic Institute

24

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/CASCall 55512125

5551212

PBX

Internal T1/CASCall 85551212 4

Regular phone(internal, 7054)

Page 25: VoIP

25

End-to-End Delay

Sender Receiver

NetworkTransit Delay

t

A A

Network

Last BitReceived

First BitTransmitted

ProcessingDelay

ProcessingDelay

End-to-End Delay

Page 26: VoIP

Fixed Delay ComponentsFixed Delay Components

• Propagation—six microseconds per kilometer• Serialization • Processing

Coding/compression/decompression/decodingPacketization

Processing Delay

Propagation DelaySerialization Delay—Buffer to Serial Link

Page 27: VoIP

Variable Delay Components Variable Delay Components

• Queuing delay• Dejitter buffers• Variable packet sizes

DejitterBuffer

Queuing Delay

Queuing Delay

Queuing Delay

Page 28: VoIP

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Delay Variation—“Jitter”Delay Variation—“Jitter”

t

t

Sender Transmits

Sink Receives

A B C

A B CD1 D2 = D1

Sender Receiver

D3 = D2

Network

85

Page 29: VoIP

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Network QoS ToolkitNetwork QoS Toolkit

Page 30: VoIP

30

Logical ConnectionsLogical Connections

Call Leg 3

Call Leg 1

IP Cloud

Call Leg 2

Call Leg 4