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. sliu@cisco.com) 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.