Circuit Switch # Dedicated transmission path# Continoues transmission of data# Message are not stored# The path is established for entire conversation# Call set-up delay, negligible transmission delay# Fixed Bandwidth transmission# no overhead bits after call set-up# prefer for long data message (minimum time connect)

Overview of Circuit and Packet Switch

Packet Switch # No Dedicated path# Transmission of packet, packet maybe stored until delivered# Route established for each packet (for datagram packet switching)# packets transmission delay# Network maybe response for individual packets# Dynamic use of bandwidth # Overhead bits in each packet# Prefer for short data message (variance time connect)# more efficiency in Bandwidth

Signalling IP Telephony

Voice over Internet Protocol (VoIP)

SIP

RTP

H.323 RTSP

UDPTCP

RTCPRSVP

Media EncapsH.261, MPEG

EthernetATM

AAI.5AAI.3/4

Sonet

PPP

IPv4, IPv6

V.34

PPP

RTSP : real time Streaming protocolRSVP : Resource Reservation ProtocolRTCP : Realtime TCP

KOMPONEN Standard H.323

� Inter-Operabilitas-VoIP

� Terminal

Komponen H.323

Hubungan komponen H.323 dan lingkungannya

SIP ProtocolSIP ProtocolSIP ProtocolSIP Protocol

� SIP is An application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between usersbetween users

COMPONENTS OF SIP ProtocolCOMPONENTS OF SIP ProtocolCOMPONENTS OF SIP ProtocolCOMPONENTS OF SIP Protocol

1. SIP User AgentsUser Agent Clients (UAC) : sends SIP requestUser Agent Servers (UAS) : receives request and returns A SIP

response2. SIP Servers� Proxy server : intermediate entity that acts as both a server and a client , plays the � Proxy server : intermediate entity that acts as both a server and a client , plays the

role of routing, enforcing policy� Redirect server : user agent server that generates 3xx response� Registrar server : server that accepts REGISTER request and places the

information request into the location service for domain it handles� Location server

Related Protocol of SIPRelated Protocol of SIPRelated Protocol of SIPRelated Protocol of SIP

SIP MessagesSIP MessagesSIP MessagesSIP Messages

►SIP messages are defined for two formats:� requests, sent from a client to a server :

1. REGISTER : used by UA to indicate current IP address and URLs to receive calls

2. INVITE : used to establish media session between UA3. ACK : confirm reliable message exchange4. CANCEL : terminate a pending request5. BYE : terminates a session between two users in

conferences6. OPTION : request information about the capabilities of caller

w/o setting up a call

SIP MessagesSIP MessagesSIP MessagesSIP Messages

►SIP messages are defined for two formats:� responses, sent from a server to a client.

1xx: Provisional : request received and being processed

2xx: Success : the action was successfully received, understood, 2xx: Success : the action was successfully received, understood, and accepted 3xx: Redirection : further action need to be taken (typically by sender) to complete the request4xx: Client error : the request content bad syntax 5xx: Server Error : the server failed to fulfill a valid request

6xx: Global Failure : the request cannot be fulfilled at any server

Komunikasi antara SIP Agent dan SIP Server

Procedure of call setup endpoint SIP

Architecture of H.324 protocol

Delay Standardization

Mean Opinion Score (MOS)

MOS Opinion

5 Very good

Method is used to define voice quality in IP networ k based on ITU-T P.800 Recommendation

Relation between MOS and R FactorTingkat Kepuasan

100

R faktor MOS

Nilai Maksimum

4 Good

3 Enough

2 Bad

1 Very bad 2,6

3,6

4,0

4,3

Sangat Baik

Baik

Cukup Baik

Buruk / tidakdiperkenankan

Kurang Baik

Buruk / berkualitasrendah

0

50

60

70

80

90

1,0

4,494Nilai Maksimum

ITU - T G.107

3,1

Topology Design

Delay Analysis� One Way Delay = coder processing delay(compression and

algorithmic delay) + packetization delay+ serialization delay + network delay

Terminal One Way Delay (ms)

SIP 42.0828125

Videophone 110.6678625

� It is a variation of packets incoming due to the difference of the packets’ path

ObservationJitter (ms)

Endpoint SIP Videophone

1 0.358125 0.01

Jitter AnalysisPacket Loss Analysis

Observation

Packet Loss (%)

Endpoint SIP

Videophone

Packet Loss is usual thing in IP network. In VoIP network, packets are sent using RTP (Real Time Protocol) and UDP (User Datagram Protocol).

2 0.183125 0.0531

3 0.044375 0.1637

4 0.1725 0.9693

5 0.40625 0.11125

6 0.03125 0.015

7 0.04125 0.00875

8 0.16125 0.01375

9 0.0475 0.005625

10 0.03625 0.075625

Rata-rata 0.1481875 0.14261

1 0 0.71

2 0 0.41

3 0 0.38

4 0 0

5 0 0.41

6 0 0.45

7 0 0.48

8 0 0.32

9 0 0.27

10 0 0.33

Rata-rata 0 0.376

Throughput Analysis� Throughput means the effective data

transfer rate, which measured in bps.

� Throughput = Packet receive Time between first and last packet

ObservationThroughput (Mbps)

Endpoint SIP Videophone

1 0.057 0.060

2 0.053 0.075

3 0.056 0.072

4 0.057 0.074

Mbps (Mega bit per second)4 0.057 0.074

5 0.042 0.064

6 0.053 0.070

7 0.056 0.072

8 0.056 0.075

9 0.054 0.077

10 0.058 0.072

Rata-rata (Mbps)

0.0542 0.0711

R Factor And MOS Computation� R Factor Computation

R = 94.2 – Id– Ief� Ief = 7 + 30 ln ( 1 + 15e)� Id = 0.024 d + 0.11(d – 177.3) H(d – 177.3)� MOS = 1 + 0.035 R + 7x10-6 R(R-60)(100-R) � MOS = 1 + 0.035 R + 7x10-6 R(R-60)(100-R)

Terminal Nilai Id Nilai Ief Nilai R Factor

Videophone 2.6560287 8.893 82.6509713

SIP 1.0099875 7 86.1900125

MOS4.1201

4.2348