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Презентация была опубликована 10 лет назад пользователемАльбина Телепнева
1 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Introducing Cisco CallManager Express Explaining Differences Between Traditional Telephony and VoIP
2 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Basic Components of a Telephony Network
3 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Central Office Switches
4 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v What Is a PBX?
5 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v What Is a Key System?
6 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Basic Call Setup
7 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Digitizing Analog Signals 1. Sample the analog signal regularly. 2. Quantize the sample. 3. Encode the value into a binary expression. 4.(Optional) Compress the samples to reduce bandwidth (multiplexing).
8 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Nyquist Theorem
9 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Quantization
10 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Quantization Techniques Linear –Uniform quantization Logarithmic quantization –Compands the signal –Provides a more uniform signal-to-noise ratio Two methods –a-law (most countries) –mu-law (Canada, United States, and Japan)
11 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Waveform algorithms –PCM –ADPCM Source algorithms –LD-CELP –CS-ACELP Voice-Compression Techniques
12 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v PCM –Waveform coding scheme ADPCM –Waveform coding scheme –Adaptive: automatic companding –Differential: changes encoded between samples only ITU standards: –G.711 rate: 64 kbps = (2 x 4 kHz) x 8 bits/sample –G.726 rate: 32 kbps = (2 x 4 kHz) x 4 bits/sample –G.726 rate: 24 kbps = (2 x 4 kHz) x 3 bits/sample –G.726 rate: 16 kbps = (2 x 4 kHz) x 2 bits/sample Example: Waveform Compression
13 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v CELP –Hybrid coding scheme High-quality voice at low bit rates; processor intensive G.728: LD-CELP – 16 kbps G.729: CS-ACELP – 8 kbps –G.729A variant – 8 kbps, less processor- intensive, allows more voice channels encoded per digital signal processor –Annex-B variant – VAD and CNG Example: Source Compression
14 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v G.729 and G.729A Comparison Both are ITU standards. Both are 8-kbps CS-ACELP. G.729 is more complex and processor intensive. G.729 is slightly higher quality than G.729A. Compression delay is the same (10 to 20 ms). Annex-B variant can be applied to either.
15 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Real-Time Transport Protocol Provides end-to-end network functions and delivery services for delay-sensitive, real-time data, such as voice and video Works with queuing to prioritize voice traffic over other traffic Services include: –Payload type identification –Sequence numbering –Time-stamping –Delivery monitoring
16 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Monitors the quality of the data distribution and provides control information Provides feedback on current network conditions Allows hosts that are involved in an RTP session to exchange information about monitoring and controlling the session Provides a separate flow from RTP for UDP transport use Real-Time Transport Control Protocol
17 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v RTP header compression saves bandwidth by compressing packet headers across WAN links. RTP Header Compression
18 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Congested WAN links Slow links (less than 2 Mbps) Bandwidth on a WAN interface that needs to be conserved When to Use RTP Header Compression
19 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Summary Traditional telephony networks are composed of edge devices such as telephones, local loops, switches, and trunks. CO switches terminate local loops and provide battery, current detection, dial tone, ring generation, and the digit registers. PBXs are privately owned switches that provide basic telephone connectivity within a corporate environment and that connect to supplementary services such as voice mail. The three parts of the analog-to-digital conversion process are sampling, quantization, and encoding. The two parts of the digital-to-analog conversion process are decoding and filtering. Digital signal technology is based on the Nyquist theorem. Quantization involves dividing the range of amplitude values of an analog signal sample.
20 © 2005 Cisco Systems, Inc. All rights reserved. IPTX v Summary (Cont.) The two techniques used for voice compression are waveform compression and source compression. G.729 and G.729A compression algorithms are similar variations of CS-ACELP. The three common voice compression standards are PCM, ADPCM, and CELP. RTP carries packetized audio traffic over an IP network. RTCP provides feedback on the quality of the call, including statistics on packet loss, delay, and jitter. RTP header compression compresses the IP/UDP/RTP header in an RTP data packet from 40 bytes to approximately 2 to 4 bytes most of the time. RTP header compression is useful if you are running VoIP over narrowband or slow links or if you need to conserve bandwidth on a WAN interface.
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