pertemuan 11 - xi_voip_h323_sip [compatibility mode]
Post on 18-Jul-2016
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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
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
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