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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
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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
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KOMPONEN Standard H.323
� Inter-Operabilitas-VoIP
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� Terminal
Komponen H.323
Hubungan komponen H.323 dan lingkungannya
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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
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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
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Related Protocol of SIPRelated Protocol of SIPRelated Protocol of SIPRelated Protocol of SIP
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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
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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
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Komunikasi antara SIP Agent dan SIP Server
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Procedure of call setup endpoint SIP
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Architecture of H.324 protocol
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Delay Standardization
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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
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Topology Design
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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
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� 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
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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
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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