VOICE over IP – H - zcu.czledvina/DHT/tugraz/VoIP_Nguyet.pdf · 2008. 3. 6. · ARQ 7. ACF 5. LRQ...
Transcript of VOICE over IP – H - zcu.czledvina/DHT/tugraz/VoIP_Nguyet.pdf · 2008. 3. 6. · ARQ 7. ACF 5. LRQ...
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VOICE over IP – H.323
Advanced Computer NetworkSS2005
Presenter : Vu Thi Anh Nguyet
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OutlinesOutlines
1. Introduction
2. QoS in VoIP
3. H323
4. Signalling in VoIP
5. Conclusions
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1. Introduction to VoIP1. Introduction to VoIP
Voice over IP – the transmission of digitalized voice over packet-switched IP networks
Class 5
PSTN
IP NetworkV
V
City A
City B
Class 5
PSTN
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VoIP AdvantagesVoIP Advantages
• Lower costs per call
• Lower infrastructure costs
• New advanced features
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VoIPVoIP Packet FormatPacket Format
• Link layer size vary per media
• Using UDP protocol without TCP
• Voice carried using the RTP protocol
• Payload size depend on codec type
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2. Quality of Service (QoS )2. Quality of Service (QoS )
• QoS in a packet network is characterized by the main parameters as:
- Bandwidth
- Delay
- Packet loss
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VoIP BandwidthVoIP Bandwidth
• Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)
• PPS = (codec bit rate) / (voice payload size)
• Bandwidth = total packet size * PPS
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VoIP Bandwidth VoIP Bandwidth (cont.)(cont.)
Example:
A G.729 call (8 Kbps codec bit rate) with cRTP andthe default 20 bytes of voice payload requires:
Total packet size (bytes) = (MP header of 6 bytes) + (compressed IP/UDP/RTP header of 2 bytes) + (voice payload of 20 bytes) = 28 bytes
Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits
PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps
(160 bits = 20 bytes (default voice payload) * 8 bits per byte
Bandwidth per call
= voice packet size (224 bits) * 50 pps = 11.2 Kbps
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DelayDelay
• Input queuing
• Jitter buffer
• CODEC
• Access (up) link transmission
• Backbone network transmission
• Access (down) link transmission
• CODEC
• Packetization
• Output queuingVoice Path
Loss+
Delay
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Fixed Delay ComponentsFixed Delay Components (cont.)(cont.)
• Propagation—6 microseconds per kilometer• Processing
- Coding / compression- Decoding / decompression- Packetization
• Serialization
Processing Delay
Propagation DelaySerialization Delay—Buffer to Serial Link
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Variable Delay Components Variable Delay Components (cont.)(cont.)
• Queuing delay
• Jitter buffer
JitterBuffer
Queuing Delay
Queuing Delay
Queuing Delay
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JitterJitter
t
t
Sender
Receives
AA BB CC
AA BB CCD1 D2 = D1
SenderReceiver
Network
D3 = D2D3 = D2
Variation of interpacket arrival time
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Total Delay TimeTotal Delay Time
Total delay for above example : 167 ms
ITU-T: <150ms : not detectable
= 150 –200ms : Acceptatble quality
>200 - 300ms : unacceptable quality
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missing packet
G.729 vocoder algorithm
Packet LossPacket Loss
• The total of number of lost packets can be accepted 5%
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QoS RemarksQoS Remarks
• VoIP frames have to traverse an IP network which is unreliable.
• Frames may be dropped as a result of network congestion or data corruption.
• For real-time traffic like voice, retransmission of lost frames at the transport layer is not practical because of the additional delays.
•Voice terminals have to deal with missing voice samples, also referred to as frame erasures.
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3. 3. H.323 StandardsH.323 Standards
• H.323 is a standard that defines how voice and video devices can communicate. It specifies both signaling characteristics and host-to-host communication protocols
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H.323 StandardsH.323 Standards (cont.)(cont.)
• The H.323 standard consists of the following components and protocols:
Protocol: Feature: • H.225 Call Signalling
• H.245 Media Control
• G.711,G.722, G.723,G.728,G.729 Audio Codes
• H.261, H.263 Video Codes
• T.120 Data Sharing
• RTP/RTCP Media Transport
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H.323 ComponentsH.323 Components
H.324H.324TerminalTerminal
H.323H.323GatekeeperGatekeeper
Packet Network
H.323H.323TerminalTerminal
H.323H.323GatewayGateway
H.323H.323MCUMCU
PSTN ISDN
V.70V.70TerminalTerminal
SpeechSpeechTerminalTerminal
H.320H.320TerminalTerminal
SpeechSpeechTerminalTerminal
e
V
GK
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GatewayGateway
• The H.323 gateway reflects the characteristics of a Switches Circuit Network (SCN) endpoint and H.323 endpoint.
• It converts voice and fax calls, in real time, between the PSTN and an IP network.
• Gateways work as an H323 terminal.
• Gateways are not needed unless the interconnection with the PSTN is required.
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GatekeeperGatekeeper
• An optional H.323 Component
• Defines H.323 Zone
• Provides Centralized Call Control
• Mandatory and Optional Services
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Gatekeeper Mandatory ServicesGatekeeper Mandatory Services(cont.)(cont.)
• Address TranslationTranslates H.323 aliases (e.g. [email protected]) or
E.164 addresses (standard phone numbers) into IP transport addresses (e.g. 10.1.1.1 port 1720)
• Admissions ControlAuthorizes access to the H.323 network
• Bandwidth ControlManages endpoint bandwidth requirements
• Zone ManagementProvides the above functions to all terminals, gateways, and MCUs that register to it
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Gatekeeper Optional ServicesGatekeeper Optional Services(cont.)(cont.)
• Call control signaling
Gatekeeper Routed Call Signaling (GKRCS)
• Call authorizationRestrict certain terminals, gateways, time of day
• Bandwidth management
Reject admission if bandwidth is not available
• Call management
Services include maintaining an active call list that use to indicate busy terminals.
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Media (UDP)RTP StreamRTP StreamRTCP StreamRTCP Stream
Gatekeeper
H.245 (TCP)Open Logical Channel
H.225 (TCP)Q.931
Setup
Alerting / Connect
Open Logical Channel Acknowledge
Capabilities Exchange
RTP StreamRTP Stream
VV
H.323Gateway B
VV
H.225 (UDP)RAS
Admission Request
Admission Confirm
4. 4. H. 323 SignalingH. 323 Signaling
H.323Gateway A
VV
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RAS MessagesRAS Messages
• RAS channel is established between endpoints and Gatekeeper across an IP network.
• RAS channel is opend before any other channels which are established.
• RAS messages are carried by the UDP connection, perform registration, admission, bandwidth changes, etc.
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• GRQ/GCF/GRJ (Discovery)GRQ : A multicast message sent by a GW looking for the GK
GCF: The reply to a GW with it‘s transport address
• RRQ/RCF/RRJ (Registration)RRQ : sent from GW to GK RAS channel address
RCF : sent from GK to GW to confirm a GW registration
RAS MessagesRAS Messages (cont.)(cont.)
GRQ
GCF/GRJ
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• ARQ/ACF/ARJ (Admission)ARQ:– The GK assigned terminal identifier– The type of call (point to point)– The call model that the terminal is willing to use (direct or GK routed)– The destination address (Ex: E.164 address)
ACF:– The call model in use– The transport address and port to use for Q.931 call signalling– The allowed bandwidth for the call
RAS MessagesRAS Messages (cont.)(cont.)
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• DRQ/DCF/DRJ (Disconnect)Get rid of call state
• LRQ/LCF/LRJ (Location)Stateless name - IP address resolutionInter gatekeeper communication
• IRQ/IRR (Information Request)Ping during active callsResource information for gateways
• BRQ/BCF/BRJ (Bandwidth)Ask for more/less bandwidth during call
• URQ/UCF/URJ (Unregistration)Get rid of registration state
RAS Messages (RAS Messages (cont.)cont.)
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RAS Message ExchangeRAS Message Exchange(cont.)(cont.)
Gatekeeper A Gatekeeper B
ARQ
LRQ
IP Network
Phone A
Phone BGateway A Gateway B
H.225 (Q.931) Setup
H.225 (Q.931) Connect
RTP
ACF
LCF
VVVV
ARQ
ACF
H.245
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• SetupIncoming call
• Call Proceeding• Alerting
Phone is ringing• Connect
Media cut through (used for billing)• Release/Release Complete
Tear down call
H.225 Call Control (ISDN Q.931)H.225 Call Control (ISDN Q.931)
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• Capabilities ExchangeExchange the capabilities between two entpoints –entpoint‘s transmit and receive capabilities for audio, video, data.
• Master/ Slave Determination• Open Logical Channel/Ack
The channel is set up before the actual transmission to ensure the entpoints are ready and capable of receiving and decoding information.
H.245 System ControlH.245 System Control
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SS7 Interconnect for Voice SS7 Interconnect for Voice Gateway Call SetupGateway Call Setup
GW A
1. IAM
PSTN/SS7
SC A
PSTN/SS7
2. Setup
3. Call proc 4. ARQ
7. ACF
5. LRQ
6. LCF
9. ARQ
10. ACF11. Setup
13. Call proc12. IAM
15. ACM
8. H225 Setup
14. H225 Call proc
16. Alerting17. H225 Alert
18. Alerting19. ACM
21. Connect
22. Con. ACK
Q. 931
26. ANM
20. ANM
23. H225 Connect24. Connect
25. Con. ACKConnection Established
GK A GK B GW B SC B
SS7
H.323
Phone A
Phone B
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VoIPVoIP ConfigurationConfiguration
N x E1 N x E1
outboundoutbound
POP A
N x E1 N x E1
outboundoutbound
VPSTN N x E1
POP B
VVSLT Router
PSTN N x E1
V
HNI POPVV GW
GKSC
PSTN N x E1
VVV GW
GKSC
E1
E1
POP C
N x E1 N x E1
outboundoutbound
GW
GW
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5. Conclusions5. Conclusions
• One of the major motivations of developing VoIPnetworks is the cost benefit.
• QoS provides reduced delay and fewer dropped packets of voice traffic to ensure the good voice quality to customers.
• H.323 is probably the most important standard supporting packetized voice technology. However it is also the complex standard with many protocols .
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ReferencesReferences
[1] J. Davidson, J. Peters, “Voice over IP Fundamentals“, Cisco Press, 2000.
[2] O. Hersent, D. Gurle, J. P. Petit, “IP Telephony Packet-based multimedia communications systems“, Addison-Wesley, 2000.
[3] L. L. Peterson, B. S. Davie, “Computer Networks - A Systems Approach“, 2nd Edition, Morgan Kaufmann, 2000.
[4] J. Walrand, P. Varaiya, “High-Performance Communication Networks“, 2nd Edition, Morgan Kaufmann, 2000.
[5] http://www.cisco.com/[6] http://www.fcc.gov/voip/[7] http://www.callback4u.com/voice-over-ip/[8] Training documents, “Cisco Advance Services“, 2002.[9] Training documents, “Cisco Voice over IP (CVOICE)“, 2002.[10] Paul J. Fong, Eric Knipp, Charles Riley,“Configuring Cisco Voice over
IP“, Syngress, 2002.