VoIPVoIP

Lecture 8

Paul Flynn

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

Switch

Switch

Switch

CO

IXCSJ

SF

RTP

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)

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

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

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)

9

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

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)

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

15

Summary PageSummary Page

PBX Switch

Switch

Switch

Switch

Switch

CO

IXCSJ

SF

RTP

T1/ E1DTMF/ MFCAS/ CCS

Local LoopFXS/ FXOLoopstart/Gndstart

16

Voice Transport ProtocolsVoice Transport Protocols

17

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

19

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

Rensselaer Polytechnic Institute

21

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

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 Adaptors

1

3                 

Analog phone adaptor

Palmcontrol

2

54

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

Rensselaer Polytechnic Institute

24

IP to PSTN Call

Gateway(10.0.2.3)

3

SQLdatabase

2Use sip:85551212@10.0.2.3

Ethernet

SIP server

sipdsipc

1Bob calls 5551212

PSTN

External T1/CAS

Call 55512125

5551212

PBX

Internal T1/CASCall 85551212 4

Regular phone(internal, 7054)

25

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

28

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

29

Network QoS ToolkitNetwork QoS Toolkit

30

Logical ConnectionsLogical Connections

Call Leg 3

Call Leg 1

IP Cloud

Call Leg 2

Call Leg 4