widyatama.lecture.applied networking.iv week 09 voip

8/9/2019 Widyatama.lecture.applied Networking.iv Week 09 VOIP

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Applied Networking-IV (2231114)Lecture Week-9

VOIP Description & Operation

77--JunJun--1010 WidyatamaWidyatama UniversityUniversity--InformaticsInformatics 11

Lecture by:Lecture by: Djadja.SardjanaDjadja.Sardjana, S.T., M.M., S.T., M.M.www.slideshare.net/djadjawww.slideshare.net/djadja

[email protected]@widyatama.ac.id


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77--JunJun--1010 IMT MMIMT MM--BiztelBiztel 22

DescriptionDescription

VOIP2m00


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What is Voice over IP (VoIP)?What is Voice over IP (VoIP)?

) ) )) ) )10101010000101010101000010

1001010101010100101010101010010101010101001010101010

01010100010010101010001001

IP PacketIP Packet

10101010000101010101000010

10010101010101001010101010

10010101010101001010101010

01010100010010101010001001

Internet/Private IP Network

IP PacketIP Packet

10101010000101010101000010

10010101010101001010101010

1001010101010100101010101001010100010010101010001001

10101010000101010101000010

10010101010101001010101010

10010101010101001010101010

01010100010010101010001001

( ( (( ( (

77--JunJun--1010 33IMT MMIMT MM--BiztelBiztel


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What is IP Telephony ?What is IP Telephony ?

VoIP : Use of Private Networks

Internet Telephony : Use of Public NetworkIP Telephony : VoIP + Internet Telephony

ITU DefinitionsITU DefinitionsITU DefinitionsITU Definitions

IMT MMIMT MM--BiztelBiztel

End points are IP

AlternativesAlternativesAlternativesAlternatives

This deployment is VoIP(end points are PSTN)

77--JunJun--1010 44


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Audio Packet TransferAudio Packet Transfer

Introduction to VoIPIntroduction to VoIP 55

. .,. ., ,, ,,

i.ei.e,, 128128 kb/s orkb/s or 320320 bytes perbytes per 2020 ms)ms)

RealReal--time compression/time compression/encodingencoding (e.g., G.(e.g., G.729729A atA at 88 kb/s)kb/s)

TransportTransport to remote IP address and port number overto remote IP address and port number over

UDP (Why not TCP?)UDP (Why not TCP?) Processing on receiver side is the reverseProcessing on receiver side is the reverse


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+127

1010111101101101

Encode each quantized

sample into 8 bit code word

Sampling, Quantization,Sampling, Quantization,EncodingEncoding

Introduction to VoIPIntroduction to VoIP 66

+

-127

Sample at twice the

highest voice frequency

2 x 4000=8000 Hz

(interval of 125 sec)

Round off samples to one

of 256 levels (introduces

noise)

: x ts = s

Other techniques (differential

coding, linear prediction)

2.4 kb/s to 64 kb/s


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VoIP DeploymentsVoIP Deployments

Phone to Phoneis mainly provided byService Provider or

Private Network

(E.g. Singtels 019)

1998

PC(Web) to Phoneis mainly provided

by Service Provider

1999

IP Phone to IP Phoneis mainly provided by

Service Provider or

self Managed

(E.g. Free World Dial)

2002

post2003

. .

Wireless IP Phone

to Wireless IP Phonewill be provided by whom?

New Comer:

Skype, Vonage etc.77--JunJun--1010 77IMT MMIMT MM--BiztelBiztel


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IPIP--toto--IPIP CallsCalls

InternetInternet



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IPIP--toto--PSTNPSTN CallsCalls

InternetInternet

PSTNPSTN



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The Internet Multimedia Protocol StackThe Internet Multimedia Protocol Stack



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SIP and H.SIP and H.323323 comparisonscomparisonsSIP H.323

"New World" - a relative ofInternet protocols - simple, open

"Old World" - complex,deterministic and vertical

SIP H.323

A simple toolkit upon whichsmart clients and applications

H.323 specifies everythingincluding the codec for the

SIP H.323

Minimal delay - simplifiedsignalling scheme makes itfaster

Possibilities of delay (up to 7 or8 seconds!)

Slim and Pra matic The suite is too cumbersome to

IETF ITU

Carrier-class solution addressingthe wide area

Borne of the LAN - focusing onenterprise conferencing priorities

Taken from www.sipcenter.com

. -

elements (URLs, MIME andDNS)

packets in RTP

Leaves issues of reliability tounderlying network

Assumes fallibility of network -an unnecessary overhead

SIP messages are formatted astext.

H.323 messages are ASN.1binary encoded, addingcomplexity

deploy easily

Seamless interaction with othermedia - services are only limited

by the developers imagination

Services are nailed-down andconstricted

Many vendors developingproducts

The majority of existing IPtelephony products rely on theH.323 suite



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SIP Protocol FlowSIP Protocol FlowSIP UA

INVITE

(with SDP)100 Trying

407 Auth ReqACK

INVITE

With auth100 Trying

180 Rin in200 OK

(with SDP)ACK

BYE

RTP

200 OK


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H.323 protocol flowH.323 protocol flowH.323 Endpoint H.323 Endpoint

Call Proceeding

Open logical channel

Open logical channel inc

TCS

Alerting Connecting

Close logical channel ACK

RTP

Close logical

channel


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SIP to H.323 Protocol FlowSIP to H.323 Protocol FlowSIP UA IP-IP Gateway H.323

INVITE

(with sdp)100 Trying

Call Proceeding

Alerting

Connecting

Open logical

channel

Open logical channel

inc TCS

180 Ringing

200 OK (with

BYE Close logical channel

ACK

SDP)ACK

RTP

Close logical

channel

ACK


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CiscoCisco SIPSIP--H.323 and H.323H.323 and H.323--SIP support was introduced in IOSSIP support was introduced in IOS

release 12.3(11)Trelease 12.3(11)T

Supports Voice calls. Video is supported for some callsSupports Voice calls. Video is supported for some calls

Not a big performance impact, as no media transcoding isNot a big performance impact, as no media transcoding isinvolved. At worst case, just copy the media stream from inputinvolved. At worst case, just copy the media stream from inputbuffer to output bufferbuffer to output buffer

Costs money (2 cisco routers, 1or 2 expensive softwareCosts money (2 cisco routers, 1or 2 expensive softwarelicense)license)


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SIPSIP--H.323 Gateway NetworkH.323 Gateway NetworkArchitecture (Cisco IPArchitecture (Cisco IP--IP Gateway)IP Gateway)


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SIPSIP--H.H.323323 Gateway NetworkGateway NetworkArchitecture (Asterisk Server)Architecture (Asterisk Server)


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-- nternat onanternat ona

Gateway/Gatekeeper GDSGateway/Gatekeeper GDSpeering configurationpeering configuration


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77--JunJun--1010 IMT MMIMT MM--BiztelBiztel 1919

OperationOperation


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ImplicationsImplications

No more time-basedtele hon char es

No more time-basedtele hon char es

No concept of local call, long-distance call or IDD call

No concept of local call, long-distance call or IDD call



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Change in Business ParadigmChange in Business Paradigm

Data Service

Value-Added

Service

Value-Added Service

Voice

Traditional Telecom Model New Telecom Model

Infrastructure Infrastructure

Voice Service Data Service

The most powerful paradigm shift is the fact that applications are not woven into theplatforms. Now to be a phone company, you don't have to weave tightly the voice service

into the infrastructure. They (Vonage) turn voice into a application and shoot it across one

of these platforms. And, suddenly, you're in your business.

Michael Powell, Chairman of FCC



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Technical IssuesTechnical Issues

To achieve interoperability between different IP Telephony services,

What should be the technical common platform?What are the technical specifications to adopt?

Technical ArchitectureTechnical ArchitectureTechnical ArchitectureTechnical Architecture

How secure is IP Telephony?What about wiretapping requirements?

Security and PrivacySecurity and PrivacySecurity and PrivacySecurity and Privacy

What is the acceptable QoS for IP Telephony?How to ensure QoS?



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Business IssuesBusiness Issues

Who can interconnect to PSTN?

What is the pricing model?

InterconnectionInterconnectionInterconnectionInterconnection

Market StudiesMarket StudiesMarket StudiesMarket Studies

Whats is the numbering plans for IP Telephony?How are numbers assigned for IP Telephony services?What if there is no service provider?

NumberingsNumberingsNumberingsNumberings

How would it change the telco landscape?(cost to setup IP Telephony service ~= cost to setup Email service)



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Regulatory IssuesRegulatory Issues

How do we classify all the different IP Telephony services?

Are they subjected to similar regulation?

ClassificationClassificationClassificationClassification

LicensingLicensingLicensingLicensing

Who needs to apply for license?What if there is no service provider?

Is it applicable to IP Telephony?How about emergency numbers?

Universal AccessUniversal AccessUniversal AccessUniversal Access



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Conclusion & Final WordsConclusion & Final Words

FutureConvergence

5m00

77--JunJun--1010 2525Widyatama UniversityWidyatama University--InformaticsInformatics

widyatama.lecture.applied networking.iv week 09 voip

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