Overview of Circuit and Packet Switch

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)
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

Voice over Internet Protocol (VoIP)

Signalling IP Telephony
Media Encaps
H.261, MPEG
H.323

SIP

RTSP

RSVP

RTCP
RTP

TCP

UDP

IPv4, IPv6

PPP

Sonet

AAI.3/4

AAI.5

ATM

RTSP : real time Streaming protocol

RSVP : Resource Reservation Protocol
RTCP : Realtime TCP

PPP

Ethernet

V.34

KOMPONEN Standard H.323

Inter-Operabilitas-VoIP

Komponen H.323
Terminal

Hubungan komponen H.323 dan lingkungannya

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

COMPONENTS OF SIP Protocol

1. SIP User Agents
User Agent Clients (UAC) : sends SIP request
User Agent Servers (UAS) : receives request and returns A SIP
response
2. SIP Servers
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 SIP

SIP Messages
SIP messages are defined for two formats:
requests, sent from a client to a server :
1. REGISTER
2. INVITE
3. ACK
4. CANCEL
5. BYE
6. OPTION

: used by UA to indicate current IP address and

URLs to receive calls
: used to establish media session between UA
: confirm reliable message exchange
: terminate a pending request
: terminates a session between two users in
conferences
: request information about the capabilities of caller
w/o setting up a call

SIP 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,
and accepted
3xx: Redirection : further action need to be taken (typically by sender)
to complete the request
4xx: 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)

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

MOS
5
4
3

Opinion

Relation between MOS and R Factor

Very good
Good
Enough

Nilai Maksimum
ITU - T G.107

R faktor

Tingkat Kepuasan

MOS

100
94

Sangat Baik

4,4
4,3

90
Baik

4,0

80
Cukup Baik

3,6

70

Bad

Kurang Baik
60

Very bad

50
0

Buruk / berkualitas
rendah
Buruk / tidak
diperkenankan

3,1
2,6
1,0

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

Jitter Analysis
It is a variation of packets
incoming due to the difference
of the packets path

Packet Loss Analysis

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

Jitter (ms)

Packet Loss (%)

Observation
Endpoint SIP

Videophone

0.358125

0.01

0.183125

0.0531

0.71

0.044375

0.1637

0.41

0.1725

0.9693

0.38

0.40625

0.11125

0.41

0.03125

0.015
6

0.45

0.04125

0.00875
7

0.48

0.16125

0.01375

0.32

0.0475

0.005625

0.27

10

0.03625

0.075625

10

0.33

Rata-rata

0.1481875

0.14261

Rata-rata

0.376

Observation

Endpoint
SIP

Videophone

Throughput Analysis
Throughput (Mbps)
Observation
Endpoint SIP

Videophone

0.057

0.060

0.053

0.075

0.056

0.072

0.057

0.074

0.042

0.064

0.053

0.070

0.056

0.072

0.056

0.075

0.054

0.077

10

0.058

0.072

Rata-rata
(Mbps)

0.0542

0.0711

Throughput means the effective data

transfer rate, which measured in bps.
Throughput = Packet receive
Time between first and last packet

Mbps (Mega bit per second)

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)
Terminal

Nilai Id

Nilai Ief

Nilai R Factor

Videophone

2.6560287

8.893

82.6509713

SIP

1.0099875

86.1900125

MOS
4.1201
4.2348