Code : STM#220 Samsung Electronics Co., Ltd. IP Telephony System Error Handling & Management IP Telephony System Error Handling & Management Distribution.

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Code : 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 Contents Delay/Latency Delay/Latency Jitter Jitter Echo Echo Packet Loss Packet Loss Voice Compression Voice Compression Overview Overview

© Samsung Electronics Co., Ltd. 3 Overview This book details various issues facing Voice over IP (VoIP) and explains how they can affect packet networks The issues of IP Telephony Delay/latency Jitter Echo Packet Loss Voice Compression

© Samsung Electronics Co., Ltd. 4 Delay/Latency Delay/Latency

© Samsung Electronics Co., Ltd. 5 Delay/Latency Definition The 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 Delay Handling delay Processing delay actual packetization, compression, packet switching Queuing delay Packets are held in a queue because more packet are sent out than the interface can handle at a given interval Propagation delay Speed of light in fiber or copper-based networks Being almost Imperceptible to the human ear Serialization delay

© Samsung Electronics Co., Ltd. 6 Delay/Latency First Bit Transmitted Last Bit Received Network AA SenderReceiver t PBX PBX Network Transit Delay Processing Delay Processing Delay End-to-End Delay

© Samsung Electronics Co., Ltd. 7 Delay/Latency Effect of Delay on Voice Quality 0 to 150 ms: Acceptable for most user applications 50 to 300 ms: Acceptable provided that Administrations are aware of the transmission time impact on the transmission quality of user applications above 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 time Encode + Packet + Queuing + Transmission + Decode + Jitter Buffer

© Samsung Electronics Co., Ltd. 8 Jitter Jitter

© Samsung Electronics Co., Ltd. 9 Jitter The 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

© Samsung Electronics Co., Ltd. 10 Jitter t t Sender Transmits Sink Receives A A B B C C A A B B C C D1D1 D 2 = D 1 Sender Receiver Network D 3 = D 2

© Samsung Electronics Co., Ltd. 11 Jitter Jitter Buffer Conceals 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

© Samsung Electronics Co., Ltd. 12 Jitter Jitter Buffer RTP Timestamp From Router A Interframe gap of 20ms A A Sender Receiver IP Network V V V V B B C C RouterA RouterB ms RTP Timestamp From Router A Variable Interframe Gap (Jitter) A A B B C C ms80ms RTP Timestamp From Router A Delitter Buffer removes Variation A A B B C C ms

© Samsung Electronics Co., Ltd. 13 Echo Echo

© Samsung Electronics Co., Ltd. 14 Echo What? In a phone conversation, you hear your own voice repeated The audible leak-through of your own voice into your own receive (return) path. Cause Echo is normally occurred by impedance mismatch Two basic characteristics of echo The louder the echo (echo amplitude), the more annoying it is The 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 audible Echos arriving after very short delays(25 ms) are masked by the physical and electrical sidetone signal Voice Network senderReceiver TxSender’s voice Rx Rx Receiver’s voice Tx Echo of sender’s voice

© Samsung Electronics Co., Ltd. 15 Echo Locating an Echo Leak-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 WAN GW FXO:FXS 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

© Samsung Electronics Co., Ltd. 16 Echo Effects of Network Elements on Echo Loudness The loudness contributes to echo Hybrid Transformers Echo sources are points of signal leakage between analog transmit and receive paths Hybrid transformers are often prime culprits for this signal leakage Analog signals can be reflected in the hybrid transformer in the tail circuit Ensure that output and input impedances are matched between the hybrid and the terminating device Telephones Extending the digital transmission segments closer to the actual telephone will decrease the potential for echo Routers Network delay increase user annoyance for an echo of equal strength Adding router does not cause echo; it exacerbates existing echo problem

© Samsung Electronics Co., Ltd. 17 Packet Loss Packet Loss

© Samsung Electronics Co., Ltd. 18 Packet Loss Packet 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 ping Ping plotter chariot 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

© Samsung Electronics Co., Ltd. 19 Voice Compression Voice Compression

© Samsung Electronics Co., Ltd. 20 Voice Compression Digitizing Voice: PCM Waveform Encoding Nyquist Theorem: sample at twice the highest frequency Voice frequency range: Hz Sampling 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 PCM 64 Kbps = DS-0

© Samsung Electronics Co., Ltd. 21 Voice Compression Objective: reduce bandwidth consumption Compression algorithms are optimized for voice Drawbacks/tradeoffs Quantization distortion Tandem switching degradation Delay (echo) Voice coding standards Compression method Bit rate (Kbps) Sample Size (ms) MOS Score G.711 PCM G.726 ADPCM G,728 Low Delay Codec Excited Linear Predictive (LD-CELP) G.729 Conjugate Structure Algebraic Codec Excited Linear Predictive (CS-ACELP) G.729a CS-ACELP G MP-MLQ G ACELP

© Samsung Electronics Co., Ltd. 22 Voice Compression Voice Compression Technologies Bandwidth (Kbps) Quality Unacceptable Business Quality Business Quality Toll Quality Toll Quality * 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)

Samsung Electronics Co., Ltd.