voip vs pstn

A little about VoiceA little about Voice► Sound is a disturbance of mechanical energy that Sound is a disturbance of mechanical energy that

propagates as a wavepropagates as a wave

► Sound waves, like other waves, are caracterized by:Sound waves, like other waves, are caracterized by: FrequencyFrequency: Represents the number of periods in a : Represents the number of periods in a

second and is measured in second and is measured in hertz hertz (Hz) or (Hz) or cycles per cycles per secondsecondHuman hearing frequency range: 20Hz to 20kHz Human hearing frequency range: 20Hz to 20kHz (audio) (audio)

AmplitudeAmplitude: The measure of displacement of the : The measure of displacement of the air pressure wave from its meanair pressure wave from its mean

WavelengthWavelength: The distance between repeating : The distance between repeating units of the propagation waveunits of the propagation wave

A little about VoiceA little about Voice► A voice frequency is one of the frequencies, A voice frequency is one of the frequencies,

within part of the audio range, that is used within part of the audio range, that is used for the transmission of human speechfor the transmission of human speech

► Usable (inteligente) voice frequency ranges Usable (inteligente) voice frequency ranges from approximately 300 Hz to 3400 Hz and from approximately 300 Hz to 3400 Hz and it is called voice bandit is called voice band

► Voice chanel has a range of 0 – 4kHz. Area Voice chanel has a range of 0 – 4kHz. Area between 3.4 and 4 kHz is used for system between 3.4 and 4 kHz is used for system controlcontrol

DigitalizationDigitalization

► Why voice digitalization?Why voice digitalization?Ensures better qualityEnsures better qualityProvides higher capacityProvides higher capacitySupports longer transmission distanceSupports longer transmission distanceThink as of discrete electrical impulses Think as of discrete electrical impulses

► Different voice digitalization techniques Different voice digitalization techniques existexistIn this lecture we will talk about Pulse In this lecture we will talk about Pulse

Code Modulation Code Modulation

Before we continue - Before we continue - TerminologyTerminology

► PSTN: Public Switch Telephone NetworkPSTN: Public Switch Telephone Network

► POTS: Plain Old Telephone ServicePOTS: Plain Old Telephone Service

► LEC: Local Exchange CarrierLEC: Local Exchange Carrier

► PBX: Private Brunch eXchangePBX: Private Brunch eXchange

► Bandwidth: Line capacity (in KHz or Kbits/s)Bandwidth: Line capacity (in KHz or Kbits/s)

► Voice Circuit: The bandwidth used by a voice Voice Circuit: The bandwidth used by a voice communicationcommunication

► IN Services (or Class Services): 800 Number, IN Services (or Class Services): 800 Number, LNP (Local Number Portability), Call Forward, LNP (Local Number Portability), Call Forward, etc.etc.

PSTN Overview - HistoryPSTN Overview - History► Graham Bell connected 2 rudimentary phones Graham Bell connected 2 rudimentary phones

(Carbon Membrane, Battery and a Magnet ) with (Carbon Membrane, Battery and a Magnet ) with an electrical cord in 1876 - this was a direct an electrical cord in 1876 - this was a direct connection between 2 phones with no dialingconnection between 2 phones with no dialing

► Design evolved from one-way to a bi-directional Design evolved from one-way to a bi-directional voice transmission voice transmission

► First improvement: connect every phone to an First improvement: connect every phone to an operator to switch callsoperator to switch calls

► Second improvement: Dialing and Mechanical Second improvement: Dialing and Mechanical Switches (later on: Electronic Switches) Switches (later on: Electronic Switches)

PSTN architecturePSTN architecture

►Three components:Three components: Customer premises equipment (CPE)Customer premises equipment (CPE)

► Telephone set, private branch exchanges Telephone set, private branch exchanges (PBX)(PBX)

The transmission facilitiesThe transmission facilities►Trunks and subscriber linesTrunks and subscriber lines

The switching systemThe switching system►Central offices (CO), tandemsCentral offices (CO), tandems

PSTN ArchitecturePSTN Architecture

► A pair of copper wires (Local Loop) runs A pair of copper wires (Local Loop) runs between a subscriber home and a local between a subscriber home and a local Central Office (CO or Class 5 switch)Central Office (CO or Class 5 switch)

► COs connect to their local Tandem Switch (or COs connect to their local Tandem Switch (or Class 4 Switch)Class 4 Switch)

► Local Tandem Switches connect to higher Local Tandem Switches connect to higher Layer Tandem SwitchesLayer Tandem Switches

► Switches connect through TrunksSwitches connect through Trunks

► Sam portions of the PSTN use as many as five Sam portions of the PSTN use as many as five levels of hierarchylevels of hierarchy

PSTN Hierarchical LayoutPSTN Hierarchical Layout

CO/Class 5 Switch

CO/Class 5 Switch

Class 4 Switch

Class 4 Switch

Class 4 Switch

Trunk

TrunkTrunk

TrunkTrunk

► Trunk is the link between Trunk is the link between switchesswitches Usually copper cables Usually copper cables

between local switchesbetween local switches Optical Fiber between Optical Fiber between

higher level switcheshigher level switches Trunks carry digital Trunks carry digital

voicevoice

► Local Loop is the link Local Loop is the link between a subscriber and between a subscriber and the Central Officethe Central Office Voice stream is usually Voice stream is usually

analoganalog The capacity (or The capacity (or

bandwidth) of the line bandwidth) of the line is limited to 4KHz is limited to 4KHz (64KBits/s) (64KBits/s)

ISDN (digital voice, out ISDN (digital voice, out of band signaling) did of band signaling) did not really catch upnot really catch up

Local LoopLocal Loop

Area

Call setup and releaseCall setup and release►A call requires a communications A call requires a communications

circuit between two subscribers.circuit between two subscribers.►The setup and release of connection The setup and release of connection

is triggered by signals.is triggered by signals.

Basic call setupBasic call setup

IAM: Initial Address IAM: Initial Address

MessageMessage

ACM: Address ACM: Address Complete Complete

MessageMessage

ANM: Answer ANM: Answer MessageMessage

Basic call releaseBasic call release

REL: Release REL: Release MessageMessage

RLC: Release RLC: Release CompleteComplete

MessageMessage

Pulse Code ModulationPulse Code Modulation► The advantages of digital communication The advantages of digital communication

systems (cf. analogue communication)systems (cf. analogue communication) Easier to store as a pattern of 1's and 0'sEasier to store as a pattern of 1's and 0's

► Increased Immunity Increased Immunity

non-linearitiesnon-linearities Easier to process in computers and Easier to process in computers and digital signal digital signal

processorsprocessors Can be coded for security and error correction Can be coded for security and error correction

purposespurposes Several digital signals can easily be interleaved Several digital signals can easily be interleaved

(multiplexed) and transmitted on one channel(multiplexed) and transmitted on one channel Noisy digital signals can be regenerated more Noisy digital signals can be regenerated more

effectively than analogue signals can be amplified. effectively than analogue signals can be amplified.

0000

1111

1110

1101

1100

1011

1010

1001

0001

0010

0011

0100

0101

0110

0111

A brief aside about ADCs A brief aside about ADCs

0000 0110 0111 0011 1100 1001 1011

Numbers passed from ADC to computer to represent analogue voltage

► ADCs are used to convert an analogue input voltage into a number ADCs are used to convert an analogue input voltage into a number that can be interpreted as a physical parameter by a computer.that can be interpreted as a physical parameter by a computer.

Resolution=1 part in 2n

SamplingSampling► The input signal is sampled prior to digitisation and an The input signal is sampled prior to digitisation and an

approximation to the input is reconstructed by the digital-to-approximation to the input is reconstructed by the digital-to-analogue converter:analogue converter:

Sampling Digitisation code, modulate

Transmission•Wire/optical fibre•Aerial/free-space

input

FilteringDigital-to-analogue

conversionDemodulate, Decode

output

Sampling an analogue Sampling an analogue signalsignal

► Prior to digitisation, signals must be sampledPrior to digitisation, signals must be sampled With a frequency With a frequency ffss=2=2BB=1/=1/TT

► ADC converts the height of each pulse into binary representationADC converts the height of each pulse into binary representation► Sampling involves the multiplication of the signal by a train of Sampling involves the multiplication of the signal by a train of

sampling pulsessampling pulses

ExamplesExamples

►For the compact disc (Audio CD) the For the compact disc (Audio CD) the maximum signal frequency is 20 kHz maximum signal frequency is 20 kHz and the sampling rate is 44.1 kHz. and the sampling rate is 44.1 kHz. The Nyquist Sampling Rate is 40 kHzThe Nyquist Sampling Rate is 40 kHz Hence the guard band is 4.1 kHz wide. Hence the guard band is 4.1 kHz wide.

► In the telephone system (see Section In the telephone system (see Section 5.8), the speech signal has a bandwidth 5.8), the speech signal has a bandwidth up to 3.4 kHz and a sampling rate of 8 up to 3.4 kHz and a sampling rate of 8 kHz, kHz, The Nyquist Sampling Rate is 6.8 kHzThe Nyquist Sampling Rate is 6.8 kHz Hence the guard band is 1.2 kHz wide.Hence the guard band is 1.2 kHz wide.

Signaling ConceptsSignaling Concepts► Signaling is the generation, transmission, and Signaling is the generation, transmission, and

reception of information needed to direct and control reception of information needed to direct and control the setup and disconnect of a callthe setup and disconnect of a call

► Two groups of signaling methodsTwo groups of signaling methods User-to-network signaling – end user/PSTN User-to-network signaling – end user/PSTN

signaling – Dual Tone Multi Frequency, ISDN (BRI signaling – Dual Tone Multi Frequency, ISDN (BRI and PRI)and PRI)

Network-to-network signaling – Network-to-network signaling – intercommunication between PSTN switchesintercommunication between PSTN switches

► Signaling: on hook, off hook, digits collectionSignaling: on hook, off hook, digits collection

► In-band SignalingIn-band Signaling

► Out of Band SignalingOut of Band Signaling

PSTN Signaling PSTN Signaling ► Network-to-network/Signaling between Network-to-network/Signaling between

trunks:trunks:Old Switches: in-band is not flexible, hard Old Switches: in-band is not flexible, hard

to deploy IN servicesto deploy IN servicesModern Switch use out of band Signaling. Modern Switch use out of band Signaling.

SS7 is the most popularSS7 is the most popularSS7:SS7:

► Very flexible Very flexible ► Saves bandwidth for voice Saves bandwidth for voice ► Delegates IN Services to special nodes Delegates IN Services to special nodes

in the network (SCPs)in the network (SCPs)

Signaling System No. 7 (SS7)Signaling System No. 7 (SS7)

►SS7 isSS7 is a global standard for a global standard for telecommunications telecommunications

defined defined by ITU-Tby ITU-T..►SS7 follows ISO 7 layer architecture.SS7 follows ISO 7 layer architecture.

SS7 Protocol Stack SS7 Protocol Stack

► ISUP:For call control,it ISUP:For call control,it defines the protocol defines the protocol and procedures used and procedures used to set-up, manage and to set-up, manage and release trunk circuits.release trunk circuits. Ex: Call setup or releaseEx: Call setup or release

Encapsulation and Encapsulation and translationtranslation

►Encapsulation: Some of SS7 ISUP Encapsulation: Some of SS7 ISUP messages are encapsulated into the SIP messages are encapsulated into the SIP message body in order that information message body in order that information necessary for services is not discarded in necessary for services is not discarded in the SIP request.the SIP request.

►Translation: Some critical SS7 ISUP Translation: Some critical SS7 ISUP messages are translated into the messages are translated into the corresponding SIP headers in order to corresponding SIP headers in order to determine how the SIP request will be determine how the SIP request will be routed. routed.

Enterprise TelephonyEnterprise Telephony► Private companies have different needs than Private companies have different needs than

residential users: more services, short dialingresidential users: more services, short dialing

► Options:Options:Small Business Lines (SBL): higher motley Small Business Lines (SBL): higher motley

fees, limited capabilitiesfees, limited capabilitiesCentrex Lines: Local PSTN Provider manages Centrex Lines: Local PSTN Provider manages

ET. Costly and not flexible but offers more ET. Costly and not flexible but offers more features (transfer calls, calls on hold)features (transfer calls, calls on hold)

VPNs: Private network where telephone VPNs: Private network where telephone company manages a private dialing plancompany manages a private dialing plan

Acquire own Switch or PBX Flexibility to add, Acquire own Switch or PBX Flexibility to add, move, numbers. Key Systems are small PBXsmove, numbers. Key Systems are small PBXs

Enterprise Telephony – Enterprise Telephony – PBXPBX

► Business circuit switched telephony system Business circuit switched telephony system with business featureswith business featuresCall HoldCall HoldThree way callingThree way callingCall transferCall transferForwardingForwarding

PBX usually provides a programming PBX usually provides a programming interface: CTI (Computer Telephony interface: CTI (Computer Telephony Integration) to support additional Integration) to support additional applications: Call Centers, Conferencing, etc.applications: Call Centers, Conferencing, etc.

Enterprise Telephony - Enterprise Telephony - PBX ExamplePBX Example

PBX

Key System

PSTN

Main Office

Remote Office

T1

DS3

An Introduction to An Introduction to Voice over IPVoice over IP

What is VoIP?What is VoIP?

► VoIP (VoIP (VVoice oice oover ver IInternet nternet PProtocol), rotocol), sometimes referred to as Internet telephony, sometimes referred to as Internet telephony, is a method of digitizing voice, is a method of digitizing voice, encapsulating the digitized voice into encapsulating the digitized voice into packets and transmitting those packets over packets and transmitting those packets over a packet switched IP network.a packet switched IP network.

Voice over IP - the basics Voice over IP - the basics

►Most implementations use H.323 Most implementations use H.323 protocolprotocol Same protocol that is used for IP video.Same protocol that is used for IP video. Uses TCP for call setupUses TCP for call setup Traffic is actually carried on RTP (Real Traffic is actually carried on RTP (Real

Time Protocol) which runs on top of UDP.Time Protocol) which runs on top of UDP.

VoIP models: IP CentrexVoIP models: IP Centrex

LANIP

enterprise or residence

serviceprovider

VoIP ProtocolsVoIP Protocols

►H.323 Multimedia StandardH.323 Multimedia Standard H.225 RAS - Registration, Admission, H.225 RAS - Registration, Admission,

StatusStatus Q.931 - Call Signaling (Setup & Q.931 - Call Signaling (Setup &

Termination)Termination) H.245 - Call Control (Preferences, Flow H.245 - Call Control (Preferences, Flow

Control, etc.)Control, etc.) Lots of G.7XX CODECS for audioLots of G.7XX CODECS for audio

►SIP – Session Initialization ProtocolSIP – Session Initialization Protocol Covered in next presentationCovered in next presentation

Here’s how it stacks up:Here’s how it stacks up:H.323H.323 Multimedia ProtocolMultimedia Protocol

H.225H.225 Call setup & Control – RAS Call setup & Control – RAS (Q.931)(Q.931)

H.235H.235 Security & AuthenticationSecurity & Authentication

H.245H.245 Call negotiation, capability Call negotiation, capability exchangeexchange

H.450H.450 Other supplemental ServicesOther supplemental Services

H.246H.246 Circuit Switched Network Circuit Switched Network Interop.Interop.

H.332H.332 ConferencingConferencing

H.26XH.26X Video CODECSVideo CODECS

H.7XXH.7XX Audio CODECSAudio CODECS

How they fit in: The ISO How they fit in: The ISO ModelModel

ISO Model LayerISO Model Layer Protocol or Protocol or StandardStandard

PresentationPresentation Applications / Applications / CODECSCODECS

SessionSession H.323 & SIPH.323 & SIP

TransportTransport RTP / UDP / TCPRTP / UDP / TCP

NetworkNetwork IP – Non QOSIP – Non QOS

Data LinkData Link ATM, FR, PPP, ATM, FR, PPP, EthernetEthernet

H.323H.323

►Definition: a multimedia standard that Definition: a multimedia standard that provides a foundation to transport provides a foundation to transport voice, video and data communications voice, video and data communications in an IP based non-QOS network.in an IP based non-QOS network.

►H.323 ZoneH.323 Zone Collection of terminals, gateways, MCUs Collection of terminals, gateways, MCUs

registered with a single gatekeeper. registered with a single gatekeeper.

H.323 EntitiesH.323 Entities

►Terminals (LAN Endpoints)Terminals (LAN Endpoints)►Gateways (Optional but really useful)Gateways (Optional but really useful)►Gatekeepers (Also optional)Gatekeepers (Also optional)►MCUsMCUs

H.323 EquipmentH.323 Equipment

► GatewayGateway Device that connects H.323 voice network to non-Device that connects H.323 voice network to non-

H.323 voice network (SIP or PSTN) H.323 voice network (SIP or PSTN) Allows H.323 terminals to communication with Allows H.323 terminals to communication with

non-H.323 terminalsnon-H.323 terminals► GatekeeperGatekeeper

Provides address translation (H.323 & E.164 to IP)Provides address translation (H.323 & E.164 to IP) Admission control for H.323 terminals and Admission control for H.323 terminals and

gatewaysgateways Manage bandwidth allocation Manage bandwidth allocation Other optional servicesOther optional services


►MCU (multipoint control unit)MCU (multipoint control unit) MC – multipoint controllerMC – multipoint controller

►Routes call and control signaling to ensure Routes call and control signaling to ensure endpoint compatibilityendpoint compatibility

MP – multipoint processorMP – multipoint processor►Switches, mixes and processes vice and video Switches, mixes and processes vice and video

streams to conferencing equipmentstreams to conferencing equipment


►TerminalTerminal An endpoint that supports 2-way An endpoint that supports 2-way

streaming with another H.323 terminal or streaming with another H.323 terminal or gatewaygateway

Originates and terminates callsOriginates and terminates calls Includes videoconferencing stations, hard Includes videoconferencing stations, hard

phones, & soft phonesphones, & soft phones

Call Setup using H.225 RASCall Setup using H.225 RAS

►Registration, Admission and Status (RAS), Registration, Admission and Status (RAS), is responsible for registration, admission, is responsible for registration, admission, and disengaging procedures between and disengaging procedures between H.323 Gatekeeper and Gateway. H.323 Gatekeeper and Gateway.

►Discovery: GRQ, GCF, GCRDiscovery: GRQ, GCF, GCR Unicast Discovery using UDP port 1718. Unicast Discovery using UDP port 1718.

Endpoint knows GK IP & register directlyEndpoint knows GK IP & register directly Multicast using UPD multicast address Multicast using UPD multicast address

224.0.1.41 – non static, less admin overhead224.0.1.41 – non static, less admin overhead

Call Setup using H.225 RASCall Setup using H.225 RAS

►Registration by terminals, Gateways & Registration by terminals, Gateways & MCUs using H.323 ID or E.164 addressMCUs using H.323 ID or E.164 address RRQ Registration RequestRRQ Registration Request RCF Registration ConfirmRCF Registration Confirm RRJ Registration RejectRRJ Registration Reject URQ Un-registration RequestURQ Un-registration Request URF Un-registration ConfirmURF Un-registration Confirm URJ Un-registration ConfirmURJ Un-registration Confirm

H.323 – H.225 RAS MessagesH.323 – H.225 RAS Messages

►LRQ – location requestLRQ – location request Gatekeeper A requests contact Gatekeeper A requests contact

information from directory information from directory gatekeeper. gatekeeper.

►LCF – location confirmLCF – location confirm Gatekeeper B returns IP address of Gatekeeper B returns IP address of

destination gateway to gatekeeper A.destination gateway to gatekeeper A.

Signaling using Q.931 Signaling using Q.931 messagesmessages

►Q.931 is a signaling protocol used to Q.931 is a signaling protocol used to setup, manage, and terminate H.323 setup, manage, and terminate H.323 connections between endpoints.connections between endpoints. ARQ, ACF, ARJ Admission messagesARQ, ACF, ARJ Admission messages LRQ, LCF, LRJ Location Request messagesLRQ, LCF, LRJ Location Request messages IRQ, IRR, IACK, INAK Status messagesIRQ, IRR, IACK, INAK Status messages BRQ, BCF, BRJ, RAI, RAC Bandwdith BRQ, BCF, BRJ, RAI, RAC Bandwdith

messagesmessages


►ARQ – admission requestARQ – admission request Gateway A requests admission to Gateway A requests admission to

make a call.make a call.

►ACF – admission confirmACF – admission confirm Gatekeeper A responds with IP Gatekeeper A responds with IP

address of destination gateway.address of destination gateway.


►Request in ProgressRequest in Progress RIPRIP

►Bandwidth changeBandwidth change BRQ, BCF, BRJBRQ, BCF, BRJ

►Resource AvailabilityResource Availability RAI (Indicator)RAI (Indicator) RAC (Confirm)RAC (Confirm)


►Gatekeeper DiscoveryGatekeeper Discovery GRQ, CCF, GRJGRQ, CCF, GRJ

►Terminal/Gateway RegistrationTerminal/Gateway Registration RRQ, RCF, RRJRRQ, RCF, RRJ

►Terminal/Gateway RegistrationTerminal/Gateway Registration URQ, UCF, URJURQ, UCF, URJ

►Disengage Disengage DRQ, DCF, DRJDRQ, DCF, DRJ


►Status QueriesStatus Queries IRQ – info requestIRQ – info request IRR – info request responseIRR – info request response IACK – info request ACKIACK – info request ACK INACK - info request NACKINACK - info request NACK

H.323 – Q.931 MessagesH.323 – Q.931 Messages

► AlertingAlerting Called user has been alerted, (phone is ringing)Called user has been alerted, (phone is ringing)

► Call ProceedingCall Proceeding Call has been established, no more call Call has been established, no more call

establishment information will be acceptedestablishment information will be accepted► ConnectConnect

Acceptance of call by called partyAcceptance of call by called party► SetupSetup

Indicates H.323 party wants to setup a Indicates H.323 party wants to setup a connection to called partyconnection to called party

H.323 – Q.931 MessagesH.323 – Q.931 Messages

►Release CompleteRelease Complete H.225 (Q.931) call has been released, H.225 (Q.931) call has been released,

signaling channel is now open signaling channel is now open ►StatusStatus

Sent when unknown call signaling Sent when unknown call signaling message or a status inquiry message is message or a status inquiry message is receivedreceived

►Status InquiryStatus Inquiry Requests a call’s statusRequests a call’s status

H.323 – H.245H.323 – H.245

► Establishes logical channels for transmission Establishes logical channels for transmission of H.323 data of H.323 data

► Negotiates:Negotiates: channel usagechannel usage master/slave configurationmaster/slave configuration flow control flow control Codec usedCodec used

► H.245 portsH.245 ports 1024-5000 TCP in Cisco implementation1024-5000 TCP in Cisco implementation

H.323 – H.245 MessagesH.323 – H.245 Messages

► Master/Slave DeterminationMaster/Slave Determination Determines which terminal will be master which Determines which terminal will be master which

will be slave in the callwill be slave in the call► Terminal Capability SetTerminal Capability Set

Contains information on a terminal’s ability to Contains information on a terminal’s ability to send and receive multimedia streamssend and receive multimedia streams

► Open Logical ChannelOpen Logical Channel Opens logical channel for transport of multimedia Opens logical channel for transport of multimedia

data data ► Close Logic ChannelClose Logic Channel

Closes the logical channel between two endpointsCloses the logical channel between two endpoints


►Request ModeRequest Mode Receive terminal requests type of Receive terminal requests type of

transportation from a transmit terminaltransportation from a transmit terminal Types of Modes:Types of Modes:

►VideoVideo►AudioAudio►DataData►Encryption Encryption


►Send Terminal Capacity SetSend Terminal Capacity Set Instructs far-end terminal to send transmit Instructs far-end terminal to send transmit

and receive capabilitiesand receive capabilities

►End Session CommandEnd Session Command Indicates the end of the H.245 sessionIndicates the end of the H.245 session

SIP Call FlowSIP Call FlowOutbound Proxy Inbound Proxy

BobAlice

INVITE

INVITE

INVITE

100 Trying 180 Ringing

100 Trying

180 Ringing180 Ringing 200 OK

200 OK

200 OK

RTP Voice

BYE BYE

BYE

Alice Calls Bob

Steve answers Bob’s phone

Is Bob there?

Sorry, no, can I help you

No. I need Bob.

Thanks. Bye.

ACK

Hello.

InterceptionInterceptionOutbound Proxy Inbound Proxy

BobAliceRTP

Kevin

REFER

Kevin forges a REFER from Bob

Hello?Yak

202 Accepted

Yak

REFER

202 Accepted

202 Accepted

Yak

SIP

INVITE

200 OK

BYE

BYE

BYEINVITE

200 OK

H.323 Call Setup via H.323 Call Setup via GatekeepersGatekeepers

6. Q.931 Call Setup

5. ACF8. ACF7. ARQ1. ARQ

8. Q.931Call Proceed

RTP

Directory Gatekeeper

VoIP PBXVoIP PBX

IP Phone IP Phone

Directory and Tier 1 Directory and Tier 1 Gatekeeper Call Setup Gatekeeper Call Setup

H.245

Tier 1 Gatekeeper Tier 1 Gatekeeper3. LRQ

4. LCF

2. RIP

2. LRQ

IP WAN

PSTN

Zone A Zone B

FXOE&ME1/T1

CallManager /H.323 MCU

CallManager /H.323 MCU

H.323 Gateway

H.323 Gateway

H.323 Gatekeeper

(Zone A)

H.323 Gatekeeper

(Zone B)

IP phonePBX PBX IP phone

RAS

RAS

RAS

H.225, H.245, RTCP, RTP

Direct dialing

RAS RAS

H.225, H.245, RTCP, RTPH.225, H.245, RTCP, RTP

Analog phoneAnalog phone

FXOE&ME1/T1

SGCP for Cisco IP phones

SGCPfor Cisco IP phones

H.323 call stages and signalling flows H.323 call stages and signalling flows

H.323 call stagesH.323 call stages

1) discovery and registration 1) discovery and registration (RAS)(RAS)

2) call setup (H.225)2) call setup (H.225)

3) call signalling flows 3) call signalling flows

4) media stream and media control 4) media stream and media control flowsflows

5) call termination (RAS)5) call termination (RAS)

Call-flowCall-flow

PSTN Phone MGC SIP Proxy

INVITE100 TRYING

IAM

18XACM

ANM200 OKACK

CONVERSATION

REL

RLCBYE

200 OK

INVITE18X

200 OK

ACK

BYE200 OK

SIP Phone

Codec ITU G.711Codec ITU G.711

► G.711 is the international standard for G.711 is the international standard for encoding telephone audio on a 64 kbps encoding telephone audio on a 64 kbps channel. It is a pulse code modulation (PCM) channel. It is a pulse code modulation (PCM) scheme operating at a 8 kHz sample rate, scheme operating at a 8 kHz sample rate, with 8 bits per sample, fully meeting ITU-T with 8 bits per sample, fully meeting ITU-T recommendations. The module is designed recommendations. The module is designed and tested on the TI TMS320C54x platform and tested on the TI TMS320C54x platform but can be ported to other DSP and RISC but can be ported to other DSP and RISC platforms, as well as MS Windows. platforms, as well as MS Windows.

Information from http://www.spiritcorp.com/Information from http://www.spiritcorp.com/

Codec ITU G.711(cont)Codec ITU G.711(cont)► FeaturesFeatures► Fully compliant with ITU-T G.711 Fully compliant with ITU-T G.711 ► 64 kbit/s expander input rate 64 kbit/s expander input rate ► 104 or 112 kbit/s expander output rate 104 or 112 kbit/s expander output rate ► A-law or mu-law expander input A-law or mu-law expander input ► Uniform PCM expander output Uniform PCM expander output ► 104 or 112 kbit/s compressor input rate 104 or 112 kbit/s compressor input rate ► 64 kbit/s compressor output rate 64 kbit/s compressor output rate ► Uniform PCM compressor input Uniform PCM compressor input ► A-law or mu-law compressor output A-law or mu-law compressor output ► Selectable frame/buffer memory size according to the system Selectable frame/buffer memory size according to the system

needs needs ► Very simple application interface Very simple application interface ► Compliant with TI's eXpressDSP standard. Code is reentrant, Compliant with TI's eXpressDSP standard. Code is reentrant,

supports multithreading and dynamic memory allocation. At supports multithreading and dynamic memory allocation. At the same time allows direct (non-eXpressDSP) interface to the same time allows direct (non-eXpressDSP) interface to enable static memory allocation enable static memory allocation

► Can be easily ported to any DSP or RISC platform Can be easily ported to any DSP or RISC platform

Codec ITU G.722.1 Codec ITU G.722.1

►G.722.1 is a low-bit-rate wideband G.722.1 is a low-bit-rate wideband coder, which codes speech at 24 kbps coder, which codes speech at 24 kbps or 32 kbps. The quality at 32 kbps is or 32 kbps. The quality at 32 kbps is the same as that of G.722 SB-ADPCM at the same as that of G.722 SB-ADPCM at 64 kbps. It uses a transform-coding 64 kbps. It uses a transform-coding scheme called Modulated Lapped scheme called Modulated Lapped Transform (MLT), with a 20 ms frame Transform (MLT), with a 20 ms frame size. The algorithmic delay is 40 ms (20 size. The algorithmic delay is 40 ms (20 ms frame size + 20 ms look-ahead). ms frame size + 20 ms look-ahead).

Codec ITU G.722.1(cont)Codec ITU G.722.1(cont)

► Supports all bit rates viz. 16/32 kbps at 16 Supports all bit rates viz. 16/32 kbps at 16 khz sampling rate khz sampling rate

► C callable API for initialization, encoding and C callable API for initialization, encoding and decoding of speech data decoding of speech data

► Supports Multi-channel capability Supports Multi-channel capability ► Optimized implementation Optimized implementation ► Bit Compliant with ITU-T test vectors Bit Compliant with ITU-T test vectors

Information found at http://www.ittiam.com/pages/products/g722-1.htmInformation found at http://www.ittiam.com/pages/products/g722-1.htm

Codec ITU G.723.1Codec ITU G.723.1

► G.723.1 is a speech compression algorithm G.723.1 is a speech compression algorithm standardized by ITU. G.723.1 has dual coding rates at standardized by ITU. G.723.1 has dual coding rates at 5.3 and 6.3 kbps. The vocoders process signals with 5.3 and 6.3 kbps. The vocoders process signals with 30 ms frames and have a 7.5 ms look-ahead and low 30 ms frames and have a 7.5 ms look-ahead and low distortion while passing DTMF tones through. The distortion while passing DTMF tones through. The input/output of this algorithm is 16 bit linear PCM input/output of this algorithm is 16 bit linear PCM samples. samples.

► Middle bit rate G.723.1 vocoder delivers one of the Middle bit rate G.723.1 vocoder delivers one of the highest compression ratios of any of the current ITU highest compression ratios of any of the current ITU standards without compromising speech quality. This standards without compromising speech quality. This vocoder can perform full duplex compression and vocoder can perform full duplex compression and decompression functions for multimedia, visual decompression functions for multimedia, visual telephony, wireless telephony, and videoconferencing telephony, wireless telephony, and videoconferencing products. products.

Codec ITU G.723.1 (cont)Codec ITU G.723.1 (cont)

► FeaturesFeatures► Fully bit exact with ITU-T G.723.1 Fully bit exact with ITU-T G.723.1 ► 5.3 and 6.3 Kbps encoded bit stream rates 5.3 and 6.3 Kbps encoded bit stream rates ► Discontinuous transmission support (DTX) using Voice Discontinuous transmission support (DTX) using Voice

Activity Detection (VAD) and Comfort Noise Generation Activity Detection (VAD) and Comfort Noise Generation (CNG) (CNG)

► Includes optional High Pass Filter and optional Post Includes optional High Pass Filter and optional Post Filter Filter

► Direct interface with PCM 8KHz sampled data. Both Direct interface with PCM 8KHz sampled data. Both sample-by-sample and block based processing sample-by-sample and block based processing supported supported

► Very simple application interface Very simple application interface ► Can be easily ported to any platform. Can be easily ported to any platform.


► ITU-T G.726 has speech compression ITU-T G.726 has speech compression and decompression at rates of 16, 24, and decompression at rates of 16, 24, 32 and 40 Kbps based on Adaptive 32 and 40 Kbps based on Adaptive Differential Pulse Code Modulation Differential Pulse Code Modulation (ADPCM). It can be effectively used (ADPCM). It can be effectively used for speech compression in such for speech compression in such applications as speech storing, digital applications as speech storing, digital circuit multiplication and telephony circuit multiplication and telephony applications. applications.

Codec ITU G.726 (cont)Codec ITU G.726 (cont)

► FeaturesFeatures► Fully bit exact with ITU-T G.726 Fully bit exact with ITU-T G.726 ► Sample-by-sample or block based analog input Sample-by-sample or block based analog input ► 16, 24, 32 or 40 Kbps bit stream rate 16, 24, 32 or 40 Kbps bit stream rate ► A-law, mu-law and 14-bit uniform 8 kHz PCM A-law, mu-law and 14-bit uniform 8 kHz PCM

input/output input/output ► Direct interface with PCM 8KHz sampled data. Direct interface with PCM 8KHz sampled data.

Both sample-by-sample and block based Both sample-by-sample and block based processing supported processing supported

► Very simple application interface Very simple application interface ► Can be easily ported to any DSP or RISC Can be easily ported to any DSP or RISC

platform platform


► ITU-T recommendation G.728 Annex G is the ITU-T recommendation G.728 Annex G is the fixed-point version of the coding of speech fixed-point version of the coding of speech at 16kbps using Low Delay Code Excited at 16kbps using Low Delay Code Excited Linear Prediction (LD-CELP). It uses Linear Prediction (LD-CELP). It uses backward adaptation of predictors and gain backward adaptation of predictors and gain to achieve an algorithmic delay of 0.625 ms. to achieve an algorithmic delay of 0.625 ms. Under error-free transmission conditions the Under error-free transmission conditions the perceived quality of a 16 kbit/s LD-CELP perceived quality of a 16 kbit/s LD-CELP codec is equivalent to that of a codec codec is equivalent to that of a codec conforming to 32 kbit/s ADPCM. The codec is conforming to 32 kbit/s ADPCM. The codec is suitable for applications such as VoIP.suitable for applications such as VoIP.

Codec ITU G.728(cont)Codec ITU G.728(cont)

►Features:Features:►API functions for initialization, API functions for initialization,

encoding and decoding of speech data encoding and decoding of speech data ►Supports Multi-channel operation Supports Multi-channel operation

Codec ITU G.729Codec ITU G.729► ITU-T recommendation G.729 codec belongs to the ITU-T recommendation G.729 codec belongs to the

Code-Excited Linear-Prediction coding (CELP) model Code-Excited Linear-Prediction coding (CELP) model speech coders and uses Conjugate-Structure speech coders and uses Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS_ACELP) Algebraic-Code-Excited Linear-Prediction (CS_ACELP) for coding speech signals at 8 kbits/sec. The coder for coding speech signals at 8 kbits/sec. The coder operates on speech frames of 10 ms corresponding operates on speech frames of 10 ms corresponding to 80 samples at a sampling rate of 8000 samples to 80 samples at a sampling rate of 8000 samples per second and the total algorithmic delay is 15 per second and the total algorithmic delay is 15 milliseconds. The encoder functionality includes milliseconds. The encoder functionality includes Voice Activity Detection and Comfort Noise Voice Activity Detection and Comfort Noise Generation (VAD/CNG) and the decoder is capable of Generation (VAD/CNG) and the decoder is capable of accepting silence frames. G.729 provides near toll accepting silence frames. G.729 provides near toll quality performance under clean channel conditions quality performance under clean channel conditions and is the default codec as prescribed by the Frame and is the default codec as prescribed by the Frame Relay Forum and is also suitable for voice over Relay Forum and is also suitable for voice over network (VoIP) applications.network (VoIP) applications.

Codec ITU G.729 (cont)Codec ITU G.729 (cont)

► Features:Features:► C-callable API functions for initialization, C-callable API functions for initialization,

encoding and decoding of speech data encoding and decoding of speech data ► Voice Activity Detection and Comfort Noise Voice Activity Detection and Comfort Noise

Generation Generation ► Supports Multi-channel operation and Supports Multi-channel operation and

Reentrancy Reentrancy ► Code passes all test vectors specified by ITU-T Code passes all test vectors specified by ITU-T ► Optimized implementation Optimized implementation

Notes on TableNotes on Table

► All codecs are voice-band and run at an All codecs are voice-band and run at an 8kHz sampling rate, except for G.722 which 8kHz sampling rate, except for G.722 which has a 7kHz bandwidth and 16kHz sample has a 7kHz bandwidth and 16kHz sample rate rate

► * These rates are nominal due to utilization * These rates are nominal due to utilization of silence compression schemes of silence compression schemes

► G.711 and G.722 are provided free of G.711 and G.722 are provided free of charge with G.728 if required for H.320 charge with G.728 if required for H.320

► Algorithmic delay of "non-predictive" codecs Algorithmic delay of "non-predictive" codecs is effectively zero is effectively zero

VOIP Codecs - bandwidth vs. VOIP Codecs - bandwidth vs. QualityQuality

►The tradeoffs:The tradeoffs: How much do you need (quality)?How much do you need (quality)? How much can you afford?How much can you afford? How much coding delay can you tolerate?How much coding delay can you tolerate? Do you have special needs?Do you have special needs?

Circuit SwitchingCircuit Switching

(a)(a) Circuit switching. Circuit switching.(b)(b) Packet switching. Packet switching.

SwitchingSwitching

A comparison of circuit switched and packet-A comparison of circuit switched and packet-switched networks.switched networks.

Comparison of Comparison of Packet vs. Circuit SwitchingPacket vs. Circuit Switching

CircuitCircuit PacketPacket

Call SetupCall Setup Database / Database /

SS 7 OverlaySS 7 Overlay

H.323 & SIPH.323 & SIP

CommunicationCommunications Channels Channel

DedicatedDedicated SharedShared

AddressingAddressing NANPNANP IPv4 & IPv6IPv4 & IPv6

Voice Over IP - the reasons Voice Over IP - the reasons that we have all heard!that we have all heard!

►PerceptionPerception It is cheaper to run just one network.It is cheaper to run just one network. It is easier to integrate advanced technology It is easier to integrate advanced technology

when your phone is on the network (CTI).when your phone is on the network (CTI). If you don’t do it someone else will.If you don’t do it someone else will.

►RealityReality Convergence will occur some day so it is Convergence will occur some day so it is

important that we build the required important that we build the required relationships now.relationships now.

Why VoIP? (2)Why VoIP? (2)►Why use IP for voice?Why use IP for voice?

Circuit switching works very wellCircuit switching works very wellBut now we have at least two networksBut now we have at least two networks

►The voice network – equipment that costs A LOTThe voice network – equipment that costs A LOT►The data network – the Internet – equipment that The data network – the Internet – equipment that

might not cost as muchmight not cost as much

If I could carry both types of service over If I could carry both types of service over the same network, wouldn’t that be a good the same network, wouldn’t that be a good thing?thing?

►The phone company already had this idea: B-ISDN The phone company already had this idea: B-ISDN – Broadband Integrated Services Digital Network– Broadband Integrated Services Digital Network

►How about operation and management savings?How about operation and management savings?

Why VoIP? (3)Why VoIP? (3)► Lower Equipment CostLower Equipment Cost

Phone systems are generally proprietaryPhone systems are generally proprietary► You buy a PBX and the handsets (phone) that go with itYou buy a PBX and the handsets (phone) that go with it► If you buy an e-mail server program, does everyone have If you buy an e-mail server program, does everyone have

to use the same e-mail client??? (Outlook can work with to use the same e-mail client??? (Outlook can work with Pine, Elm, Netscape, etc.)Pine, Elm, Netscape, etc.)

► PCs running Linux can often be used unlike PSTNPCs running Linux can often be used unlike PSTN Hard to develop third-party software for phone Hard to develop third-party software for phone

systemssystems► The WWW is the complete opposite of this paradigmThe WWW is the complete opposite of this paradigm

Traditional telephony is like mainframesTraditional telephony is like mainframes IP networksIP networks

► More open standardsMore open standards► More competition (for the most part)More competition (for the most part)► Moore’s Law tends to motivate data networkingMoore’s Law tends to motivate data networking

Why VoIP? (4)Why VoIP? (4)► Voice/data integration and advanced servicesVoice/data integration and advanced services

User interfaces with HTML (point-and-click) are much User interfaces with HTML (point-and-click) are much easier to deal with than keypad sequences (e.g., #33#7)easier to deal with than keypad sequences (e.g., #33#7)

Unified Messaging coming alongUnified Messaging coming along Easier new service introductionEasier new service introduction

► Lower bandwidth requirementsLower bandwidth requirements New voice communications gear can take advantage of New voice communications gear can take advantage of

low-rate vocoderslow-rate vocoders Legacy telephone network largely stuck with 64 kbps per Legacy telephone network largely stuck with 64 kbps per

call call ► Some bandwidth reduction in some areasSome bandwidth reduction in some areas

Bandwidth management is better (think of silence Bandwidth management is better (think of silence periods)periods)

Why VoIP? (5)Why VoIP? (5)►Mobility Mobility

IP address not tied to a particular areaIP address not tied to a particular area

►Widespread availability of IPWidespread availability of IPIt’s EVERYWHEREIt’s EVERYWHERE

►But so are telephones But so are telephones Other packet-based mechanisms: FR, Other packet-based mechanisms: FR,

ATMATM►Not as presentNot as present

Still to rememberStill to remember►IP IP

has no guaranteeshas no guaranteesengineered for data trafficengineered for data traffic

►data integrity is very important; TCP does a great data integrity is very important; TCP does a great job at thisjob at this

►latency – and VARIABLE latency – was not a major latency – and VARIABLE latency – was not a major concern in the design and engineering of TCP/IPconcern in the design and engineering of TCP/IP

►asynchronous – can start, stop, tolerate variability asynchronous – can start, stop, tolerate variability in transmission speedsin transmission speeds

►Quality is generally lower than PSTN but Quality is generally lower than PSTN but the gap is closingthe gap is closing

► PSTN better in handling 911 and power PSTN better in handling 911 and power outagesoutages

VoIP ChallengesVoIP Challenges►Speech QualitySpeech QualityNot quite there yet! But it is a relative Not quite there yet! But it is a relative

measuremeasure

►Network Reliability and ScalabilityNetwork Reliability and ScalabilityDe-coupling between application and networkDe-coupling between application and network

►Managing access and prioritizing trafficManaging access and prioritizing trafficWhose voice is more important than data?Whose voice is more important than data?

►Business CaseBusiness CaseVendor bewareVendor beware

►SecuritySecurity

voip vs pstn

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