1 VoIP – Voice over Internet Protocol Patrick Hügenell, Andreas Vetter – TIM01AGR – 2003 VoIP...

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VoIP – Voice over Internet Protocol

Patrick Hügenell, Andreas Vetter – TIM01AGR – 2003

VoIP

Voice over IP

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• Introduction• How VoIP works• Voice over IP Scenarios• The „Pros and Cons“• RTF (Real-time Transport Protocol)• H.323 Standard• Speech Samples• Future Aspects

Contents

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What is VoIP ?

• VoIP = Voice over Internet Protocol

• Transmission of voice and video over data-networks (internet, intranet), using the Internet protocol (IP)

• Possible variants using VoIP :– PC to PC– PC to Phone and the other way around– Phone to Phone using the Internet / Intranet

Introduction

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Microphone transforms acoustic waves into "current fluctuations"

microphone

Signal is transferredby a wire

Loudspeaker transforms the signal back into sounds

Loudspeaker

How VoIP works

The Classical Approach: Analog Transmission

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How VoIP works

The Classical Approach : Digital Transmission

0111000111000001………

In time intervals the amperage is measured and its value is transferred

Microphone transforms acoustic waves into "current fluctuations"

Usual: 256 different values (256=2^8, => 8 bit);8000 measures per second => 64 kBit/s

transmission of 0 and 1 as on and turn-off processes;nearly error free

Converted back into current fluctuations and acoustic waves

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How VoIP works

Tomorrow (?): Voice over IP

The Internet works package-oriented: data stream is distributed on packets, which are sent independently to the target

Transmission of media (audio,video) over the Internet Protocol:

This means for the media (speech), like you would:- record it on a tape- cut the tape into pieces- put the pieces in envelopes- at the destination paste the pieces in the correct order and play the tape

… and this all in real-time (almost)!

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How VoIP works

• General approach in case of submitting speech:– Recording and digitalisazion of speech– Segmentation of data packets– The packets are transported over the internet to the receiver– The receiver‘s hardware turns the data back into speech– Timestamp guarantees correct time and order

How VoIP works

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Synchronisation through Timestamp

56 57 58 59 75 76 77 78 79 95 9697 98 99

sampled

78 98

Sampling Instants

packetized58

Timestamp

sent58 78 98 118 138 158

received58 1581381189878

58 78 98 118 138 158 replayed

How VoIP works

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Site BSite A

Internet

PSTN

Private VoiceNetworkVoice

SwitchVoice

Switch

Intranet

IPRouter PC

Phone

Fax

IPRouterPC

Phone

Fax

Modem Modem

Voice over IP Scenarios

The Classical Approach: Separate Voice and Data Networks

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The Future Approach: Voice/Fax over IP - A Unified Network

Site BSite A

Internet

Intranet

IPRouter

PC withVoice/Fax

IP Phone

PSTNGateway

PSTNGateway

PC withVoice/Fax

IP Phone

IPRouter

PSTN

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The Intermediate Approach: Voice/Fax over IP - A Unified Network

Site BSite A

Internet

Intranet

PC withVoice/Fax

IP Phone

PC withVoice/Fax

IP PhonePSTNGateway

PSTNGateway

VoiceSwitch

Phone

Fax

VoiceSwitch

Phone

Fax

IPRouter

IPRouter

PSTN

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Site BSite A

Internet

IntranetPC with Headset

Roger Wilco Server

The Gamers Approach: Use of “Roger Wilco”

PC with HeadsetRoger Wilco Client

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• Advantages– Cost savings on long distance calls– Less (no) need for private telephone networks– Single RJ-45 connector at the workplace for all services– Enables new multimedia features, e.g. human operator assisted

e-commerce

• Problems / Open Questions– Control of delay, jitter and packet loss over IP-based networks– QoS guarantees– Bandwidth

The „Pros and Cons“

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• Introduction• How VoIP works• Voice over IP Scenarios• The „Pros and Cons“ • RTF (Real-time Transport Protocol)• H.323 Standard• Speech Samples• Future Aspects

Contents

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Real-time Transport Protocol (RTP)

• RTP version 2 is specified by RFC 1889 • RTP covers functions such as

– Payload type identification (which codec and Framing)– arranging the packages by sequence numbers synchronisation by time stamps

(playing time of the individual Samples or Frames; – Synchronisation of several Media Streams – quality control and statistics

• RTP is defined independent of transportation protocol, sets however typically on UDP/IP

• RTP contains no mechanisms to the safety device of the quality of service (QoS)


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RTP and RTCP

• The concept covers two closely linked protocols– RTP (real-time transport Protocol): transport the Media Stream

– RTCP (real-time transport control Protocol): informed about the participants attached at the Media Stream and the quality of service (QoS)

• For each Media Stream and each direction, received from them, a separate RTP session opens, you get– an IP address (multicast address for Conferencing)

– a pair of UDP ports • n for the Media Stream (default 5004) • n+1 for RTCP (default 5005)


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RTCP: Paket-Typs and scalability

• RTCP reports are generated by all transmitters and receivers of the session in regularly intervals (statistic information)

• the interval must be selected in such a way that the total load remains within limits (recommended: approx.. 5% of the range of the session)

• very large groups to make possible (with thousands of participants), the interval computed due to the – amount of active participant – the extent of the individual report


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Header Format

0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|X| CC |M| PT | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | timestamp | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | synchronization source (SSRC) identifier | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | contributing source (CSRC) identifiers | | .... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | optional header extension | | .... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


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Overhead und Header Compression

• RTP header Overhead: Example – compressed language - 8 kbps

– every 20 ms a RTP package with 20 oktetts Payload

– 40 oktetts header per package - 24 kbps

• RTP header compression: – reduces Overhead

– router decompression necessarily - > high processor load

– Slow i-net connection (e.g. VoIP over V.34 modem)

Payload (20) 24 kbit/s

~10 kbit/s

IP (20) UDP (8) RTP (12)

Payload(20)compressed RTP (2 ... 4) cRTP


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VoIP Delay Budget (example: G.711)

• Input queuing not relevant

• Jitter buffer 4 ... 40 ms

• Decoding 1 ms *)

• Access (up) link transmission

• Backbone network transmission t

• Access (down) link transmission

• Coding and framing 20 ms *)

• Packetization 20 ms

• Output queuing 0 ... 8 ms

Total 45 ... 89 + t [ms]ITU-T G.114 Recommended (150 ms)

*) depends on uses codec


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H.323 : Packet-BasedMultimedia Communications Systems

System Control

Video I/OEquipment

System ControlUser Interface

RAS Control

Q.931 Call Setup

H.245 Control

User Data Applications T.120, etc.

LAN

H.323

H.225.0Layer

RTPAudio CodecG.711, G.722

Audio I/OEquipment

Video CodecH.261, H.263

H.323 Standard

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MOS - Mean Opinion Score

Speech Quality

5 - Excellent4 - Good3 - Fair2 - Poor1 - Bad

Poor 2

Fair 3

Good 4

Excellent 5

64 32 16 8 4.8 2.4

Bit Rate [kbps]

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Speech Samples

G. 71164 kbps

LPC2.4 kbps

Single Speaker

Music

Bit Errors0.1%

Bit Errors1%

GSM13 kbps

Speech Samples

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Future Aspects

Future Aspects

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Thanks for your attention

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