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Overview of Circuit and Packet Switch. - PowerPoint PPT Presentation
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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)
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
Overview of Circuit and Packet Switch
IP Telephony
Voice over Internet Protocol (VoIP)
SIP
RTP
H.323 RTSP
EthernetATM
AAI.5AAI.3/4
Sonet
PPP
UDPTCP
RTCPRSVP
Media EncapsH.261, MPEG
IPv4, IPv6
V.34
PPP
Signalling IP telephony
KOMPONEN Standard H.323
Inter-Operabilitas-VoIP
Terminal
Audio CodecG.711G.723G.729
Audio CodecG.711G.723G.729
Video CodecG.261H.263
User DataT.120
System Control
H.245 Control
Q.931
RAS Controling
RTP
LANINTERFACE
Audio Equiptment
Video Equiptment
Data Equiptment
System ControlUser Interface
Komponen H.323
Hubungan komponen H.323 dan lingkungannya
SIP ProtocolSIP 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)
User Agent Servers (UAS)
2. SIP Servers
Proxy server Redirect server
Location server Registrar server
Related Protocol of SIP
SIP Messages►SIP messages are defined for two formats:requests, sent from a client to a server :
1. INVITE 4. BYE
2. ACK 5. CANCEL
3. OPTION 6. REGISTER
responses, sent from a server to a client.
1xx: Provisional 4xx: Client Error
2xx: Success 5xx: Server Error
3xx: Redirection 6xx: Global Failure
Komunikasi antara SIP Agent dan SIP Server SIP SERVER
PROXY
REDIRECT
REGISTRAR
LOCATION
(1)
UAC
UAS
UAC
UAS
SIP USER AGENT SIP USER AGENT
Signaling Messages BetweenUser Agent And Server
Keterangan : (1) Fungsi server tidak harus berada pada satu komputer
UAC : User Agent Client
UAS : User Agent Server
@2005 STT Telkom
Procedure of call setup endpoint SIP
`
SIP User [email protected]
`
SIP User [email protected] Server
10.14.200.60
SIP : INVITE
SIP : INVITE
SIP : TRYING
SIP : TRYING
SIP : RINGING
SIP : RINGING
SIP : OK
SIP : OK
SIP : ACK
RTP media stream
SIP : BYE
SIP : BYE
SIP : OK
RTP media stream
SIP : ACK
SIP : OK
Architecture of H.324 protocol
Video I/O Equipment
System Control
User Data AplicationT.120, etc
Audio I/O Equipment
Video CodecH.263/H.261
Audio CodecG.723.1
Data ProtocolV.14, LAPM,etc
Control protocol SRP,LAPMH.245 Procedure
Receive Path Delay Multiplex/
Demultiplex H.223
ModemV.34/V.8
Modem ControlV.250
PSTNNetwork
Scope of Protocol H.324
@ 2005 STT Telkom
Delay Standardization
Mean Opinion Score (MOS)
MOS Opinion
5 Very good
4 Good
3 Enough
2 Bad
1 Very bad
Method is used to define voice quality in IP network based on
ITU-T P.800 Recommendation
Relation between MOS and R Factor
2,6
3,6
4,0
4,3
Tingkat Kepuasan
Sangat Baik
Baik
Cukup Baik
Buruk / tidakdiperkenankan
Kurang Baik
Buruk / berkualitasrendah
0
50
60
70
80
90
100
1,0
4,494
R faktor MOS
Nilai MaksimumITU - T G.107
3,1
Topology Design
Network Analyzer
`
SIP User [email protected]
SIP Proxy Server
Gateway
HUB
HUB
`
SIP User [email protected]
GatewayVideophone
Videophone
Delay AnalysisOne 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
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
Jitter AnalysisPacket Loss Analysis
Observation
Packet Loss (%)
Endpoint SIP
Videophone
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
Packet Loss is usual thing in IP network. In VoIP network, packets are sent using RTP (Real Time Protocol) and UDP (User Datagram Protocol).
Throughput Analysis 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)
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
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 ComputationR Factor Computation
R = 94.2 – Id– IefIef = 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 7 86.1900125
MOS
4.1201
4.2348
