chap5 VoIP

7/29/2019 chap5 VoIP

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Voice and Multimedia over IP

Voice over IP scenarios

PC to PC, PC to phone, phone to phone, Corporate, core PSTN

Digital voice

Elements of the sampling theory

PCM and TDM, Classical PSTN architecture

Quality, Codec, Technical issues

RTP - Transport of voice

and other multimedia, soft realtime applications

The H323 architecture

SIP signalling


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Phone-to-Phone VoIP

Use existing telephone with an Analogue Telephone Adapter (ATA) or use anIP phone

Both connect to Broadband modem

Called party may be another VoIP user

Or, via a gateway, a traditional PSTN customer


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Many traditional PSTN calls are carried as VoIP in part

For efficiency reasons they may travel with other IP traffic Different from normal VoIP

Invisible to end customer

Private IP network, not internet

Controlled quality

VoIP in the PSTN


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Corporate VoIP

Increasingly growing popularity

Inter-site voice carried as IP over leased line or VPN (huge money-saving especially for international communications)

Additionally, a single desk wiring infrastructure (LAN) may carry bothdata and voice (as VoIP)

Voice may stay as IP or be converted to PSTN


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Digital voice


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Digital voice

Sine wave Sampling Quantizing

Analog Signal Nyquist Fs=2Fc Quantization Noise


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Sampling

8KHz Sampling Rate One sample each 125 s (8KHz), 8 bits per sample x 8KHz 64Kbps

Nyquists theorem

To be able to restore without loss a signal with a cut frequencyfc, weneed to sample it at a frequency of at least 2fc.

Sampling at 8KHz we loose frequency components above 4KHz. Formusic audio this is not acceptable (MP3:44.1 KHz)


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Quantization

We choose the quantized value closest to the Analog signal

The truncation difference is called Quantization Noise

Linear Digitization leads to a high SNR (Signal-to-Noise Ratio) becausesmall amplitudes suffer the same treatment as big ones


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Logarithmic Quantization Better SNR

We give more levels (more precision) to small values => better SNR

A-law (Europ) and -law (USA)


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PCM - Pulse-Code Modulation

DS0 (Digital Signal level 0

single digital voice channel in the PCM system

Synchronously one sample each 125s 64 Kbps total throughput

A hierarchy of TDM multiplexes (PDH & SDH) used in PSTN core

Analog only at the local loop

ADC and DAC at the toll office (a word about Echo here)


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Synchronous Digital Hierarchy

E1 (or DS1 - Digital Signal level 1)

Multiplexes synchronously 32 DS0 channels, but channels 0 and 16reserved for signaling (remains 30 voice channels)

Sample rate must remain 8000 samples (Bytes) /s for all channels =>the frame time must always remain 125s (at all the levels DS2 DS3 etc)

E1 bit rate: 8bits x 32 / 125s = 2.048 Mbps


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Synchronous Digital Hierarchy E carriers

Similarly, E2 multiplexes 4 E1 channels, E3=16E1, etc..

The T-carrier system, similar but different, is used in USA

Ex: T1 = 24 DS0


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Codec The Encoder-Decoder

G.711 The classical codec

Simply what we saw so far.. 8000 Byte samples, 64 Kbps encoded in the

DS0 format

Can we achieve the same quality with a smaller bandwidth?

Silence suppression

Compression

Many encoding algorithms and corresponding Codecs havebeen invented and standardized, all seeking the reduce

bandwidth with a minor (or no) loss of quality.

Drawback: adding some computational delay

This is minor compared to bandwidth gain


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Some standardized Codecs

V i Q li


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Voice Quality

Subjective: Mean Opinion Score (MOS)

5 - Perfect. Like face-to-face conversation or radio reception. 4 - Fair. Imperfections can be perceived, but sound still clear.

3 - Annoying.

2 - Very annoying. Nearly impossible to communicate.

1 - Impossible to communicate

Objective

Derived from bandwidth, delay and packet loss rate

Toll (payable) service must be in the 4 to 5 range

D l


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Delay

Delay should be


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Jitter

Voice is a CBR service

A packet that arrives too late must be dropped

Continuous playback

What is Jitter?

Different packets experience different delays in the network

Varying queue sizes

Different paths introduce jitter, but load balancing algorithms work on amicro-flow level, not on packet level (coarse grain)

TCP transport not suitable

Congestion window management increases jitter

retransmitted packets will be dropped anyway. VoIP uses RTP/UDP

Solution

Jitter buffer at the receiver

Deliberately adding some delay to ensure continuous playback

Jitt B ff


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Jitter Buffer

The bigger the playback delay, the lesser the jitter

It is important to have low jitter from the network, otherwisewe need a big jitter delay.

P k t L t


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Packet Loss rate

IP network (Best effort service)

Routers might drop packets (RED: Random Early Detection)

Packets arriving too late are deliberately droped by receiver Loss rate tolerable up to 5%

10% if we use Interleaving


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RTP - The Realtime Transport Protocol


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RTP - The Realtime Transport Protocol

Transport mechanism designed for Soft Realtime applications The most important feature is Timestamping

Manage jitter

Synchronize multiple streams, i.e. audio and image in a movie

RTP


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RTP

P: padding (multiple of 4 bytes)X: exist extension headers (unused)M: app-specific marker (ex: start of video frame)Payload Type: specifies type and encoding algorithm (ex: G.711 voice)Sequence numbering to detect loss of packetsTimestamp: jitter management and synchronization of streams

SSRC: identifies a stream (many streams may be multplexed on a RTP stream, such asvideo and audio)CSRC combined with CC: permits many sources to be mixed (CC is the count andSSRC is a list). The mixer would be the SSRC


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H.323 architecture

H323 A hit t d P t l St k


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H323 Architecture and Protocol Stack

H 323 ll


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H.323 call sequence

1. PC discovers Gatekeeper (broadcast discovery request packet)

2. Gatekeeper responds

3. Client registers in the Gatekeeper zone (RAS protocol H225)4. Client requests bandwidth (this is to manage call admission and QoS!)

5. Client establishes TCP session with Gatekeeper for signalling

6. Client sends SETUP message (Q.931 signalling)

7. Gatekeeper contacts Gateway and sends CALL_PROCEEDING to client

8. Gateway Calls destination number

9. Callee rings, Q.931 ALERT message (GWclient)

10. Callee responds, Q.931 CONNECT message (GWclient)

11. H245 messages to negotiate capabilities (codec, video, teleconf)

12. Data flow proceeds using RTP13. Call terminates (callee hangs), GW alerts client (Q.931)

14. Client releases Bandwidth to gatekeeper (RAS message)


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SIP

The Session Initiation Protocol

SIP S i I iti ti P t l


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SIP - Session Initiation Protocol

LOOKUP and REPLY methods not specified in SIP (free implementation)

REGISTER method allows clients to register to proxy and inform of location

Another option to use SIP is P2P. Caller and callee exchange directly

INVITE/OK/ACK through a TCP connection and then start exchanging data


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SIP Methods

The encoding of messages is based on HTTP

C i f H 323 d SIP


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Comparison of H.323 and SIP

MP3 (for music audio)


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The masking effectThreshold of audibility asfunction of frequency

MP3 (for music audio)

Sampling usually is at 44.1 KHz


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Video Analog Systems

The scanning pattern used for NTSC video andtelevision.


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The JPEG Standard

The operation of JPEG in lossy sequential mode.


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JPEG Block Preparation

RGB input data After block preparation

Y=.3R+.59G+.11B I=.6R-.28G-.32B Q=.21R-.52G+.31B


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JPEG -- DTC

One block of the Ymatrix The DTC coefficients

G


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JPEG

Computation of the quantized DTC coefficients.

Th JPEG S d d


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The JPEG Standard

The order in which the quantized values are transmitted

Run-length encoding: exploiting repetitions

Th MPEG St d d


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The MPEG Standard

Synchronization of the audio and video streams in MPEG-1.

MPEG F T


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MPEG Frame Types

I (Intracoded) frames: Self-contained JPEG-encoded still

pictures.

P (Predictive) frames: Block-by-block difference with the lastframe.

B (Bidirectional) frames: Differences between the last and nextframe.

D (DC-coded) frames: Block averages used for fast forward.

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chap5 VoIP