code : stm#220 samsung electronics co., ltd. ip telephony system error handling & management ip...

Code : STM#220Code : STM#220

Samsung Electronics Co., Ltd.

IP Telephony System Error Handling & Management

IP Telephony System Error Handling & Management

Distribution

EnglishED01

© Samsung Electronics Co., Ltd. 2

ContentsContents

Delay/LatencyDelay/Latency

JitterJitter

EchoEcho

Packet LossPacket Loss

Voice CompressionVoice Compression

OverviewOverview


OverviewOverview

This book details various issues facing Voice over IP (VoIP) and explains how they can affect packet networks

The issues of IP TelephonyDelay/latency

Jitter

Echo

Packet Loss

Voice Compression



DefinitionThe amount of time it takes for speech to exit the speaker’s mouth and reach the listener’s ear

VoIP technologies impose a fundamental transmission delay due to packetization and the buffering of received packets before playout at the receiving endpoint

The Type of DelayHandling delay

Processing delayactual packetization, compression, packet switching

Queuing delayPackets are held in a queue because more packet are sent out than the interface

can handle at a given interval

Propagation delaySpeed of light in fiber or copper-based networksBeing almost Imperceptible to the human ear

Serialization delay



First Bit Transmitted

Last Bit Received

Network

AA AA

Sender Receiver

t

PBXPBX PBXPBX

Network Transit Delay

ProcessingDelay

ProcessingDelay

End-to-End Delay



Effect of Delay on Voice Quality0 to 150 ms: Acceptable for most user applications50 to 300 ms: Acceptable provided that Administrations are aware of the transmission time impact on the transmission quality of user applicationsabove 300 ms: Unacceptable for general network planning purposes; however, it is recognized that in some exceptional cases this limit will be exceeded.End to end transmission timeEncode + Packet + Queuing + Transmission + Decode + Jitter Buffer

Perceived Link Quality

Excellent Good Poor Unacceptable

0msec ~ 150 151 ~ 300 301 ~ 450 451 ~


JitterJitter


JitterJitter

JitterThe variation in the delay of received packets

Due to network congestion, improper queuing, or configuration errors, this steady stream can become lumpy, or the delay between each packet can vary instead of remaining constant.

Jitter can cause missing syllables or some parts of word


JitterJitter

t

t

Sender Transmits

Sink Receives

AA BB CC

AA BB CC

D1 D2 = D1

Sender Receiver

Network

D3 = D2D3 = D2


JitterJitter

Jitter BufferConceals interarrival packet delay variation

The Jitter Buffer adds to the end-to-end Delay

The more jitter, the larger jitter buffer needs to be compensate for the unpredictable nature of the packet network


JitterJitter

Jitter Buffer

RTP Timestamp From Router AInterframe gap of 20ms

AA

Sender Receiver

IPNetworkVV VV

BB CC

RouterA RouterB

10 30 50

20ms 20ms

RTP Timestamp From Router AVariable Interframe Gap (Jitter)

AA BB CC10 30 50

20ms 80ms

RTP Timestamp From Router ADelitter Buffer removes Variation

AA BB CC10 30 50

20ms 20ms


EchoEcho


EchoEcho

What?In a phone conversation, you hear your own voice repeatedThe audible leak-through of your own voice into your own receive (return) path.

CauseEcho is normally occurred by impedance mismatch

Two basic characteristics of echoThe louder the echo (echo amplitude), the more annoying it isThe longer the round-trip delay (the “later” the echo), the more annoying it is

Echos must be delayed by at least 25 ms to be audibleEchos arriving after very short delays(25 ms) are masked by the physical and elec

trical sidetone signal

VoiceNetwork

sender Receiver

Tx Sender’s voice Rx

Rx Receiver’s voice Tx

Echo of sender’s voice


EchoEcho

Locating an EchoLeak-through happens only in the analog portion of the network

Analog signals can leak from one path to another, or acoustically trough the air from a loudspeaker to the a microphone

Voice traffic in the digital portions of the network dose not leak from one path into another

The analog signals that represent bits can tolerate a lot of distortion

PBX PBX

WAN

GW GW

FXO:FXS FXO:FXS

E&M E&M

Analog(echo signal returns

too quickly to be audible)

Digital(long delay, >30

ms each direction)

Analog (Tail circuit)(good candidates for

echo sources)

sender Receiver


EchoEcho

Effects of Network Elements on EchoLoudness

The loudness contributes to echo

Hybrid TransformersEcho sources are points of signal leakage between analog transmit and

receive pathsHybrid transformers are often prime culprits for this signal leakageAnalog signals can be reflected in the hybrid transformer in the tail circuitEnsure that output and input impedances are matched between the

hybrid and the terminating device

TelephonesExtending the digital transmission segments closer to the actual

telephone will decrease the potential for echo

RoutersNetwork delay increase user annoyance for an echo of equal strengthAdding router does not cause echo; it exacerbates existing echo problem



Packet LossPacket loss in data networks is both common and expected

When putting critical traffic on data networks, it is important to control the amount of packet loss in that network

pingPing plotterchariot

When putting voice on data network, it is important to build a network that can successfully transport voice in a reliable and timely manner

The suggested rate of packet loss that is allowed for VoIP communication is 3% or less

< 1% < 3%

> 3%

Business Communication Quality

The Rate of Packet loss


Voice Compression Voice Compression


Voice Compression Voice Compression

Digitizing Voice: PCM Waveform EncodingNyquist Theorem: sample at twice the highest frequency

Voice frequency range: 300-3400 HzSampling frequency = 8000/sec (every 125us)Bit rate: (2 x 4 Khz) x 8 bits per sample = 64,000 bits per second (DS-0)

By far the most commonly used method

PCM64 Kbps

= DS-0



Voice CompressionObjective: reduce bandwidth consumption

Compression algorithms are optimized for voice

Drawbacks/tradeoffsQuantization distortionTandem switching degradationDelay (echo)

Voice coding standardsCompression method Bit rate

(Kbps)Sample Size

(ms)MOS Score

G.711 PCM 64 0.125 4.1

G.726 ADPCM 32 0.125 3.85

G,728 Low Delay Codec Excited Linear Predictive (LD-CELP) 15 0.625 3.61

G.729 Conjugate Structure Algebraic Codec Excited Linear Predictive (CS-ACELP) 8 10 3.92

G.729a CS-ACELP 8 10 3.7

G.723.1 MP-MLQ 6.3 30 3.9

G.723.1 ACELP 5.3 30 3.65



Voice Compression Technologies

Bandwidth(Kbps)

Quality

UnacceptableUnacceptable BusinessQuality

BusinessQuality

TollQuality

TollQuality

8

16

32

24

64

0

*PCM (G.711)

*PCM (G.711)

*ADPCM 32 (G.726)

*ADPCM 32 (G.726)

*ADPCM 24 (G.726)

*ADPCM 24 (G.726)

*ADPCM 16 (G.726)

*ADPCM 16 (G.726) *

LDCELP 16 (G.728)*

LDCELP 16 (G.728)

*CS-ACELP 8 (G.729)

*CS-ACELP 8 (G.729)*

LPC 4.8*

LPC 4.8

(Cellular)(Cellular)

Samsung Electronics Co., Ltd.

code : stm#220 samsung electronics co., ltd. ip telephony system error handling & management ip...

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