ZyXEL MAX318M User Guide - Page 151

The VoIP General Screens, 9.1 VoIP Overview, 9.1.1 What You Need to Know

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CHAPTER 9 The VoIP General Screens 9.1 VoIP Overview The features mentioned in this chapter are for models that has phone port(s) and you can make telephone calls over the Internet using the WiMAX Device. The VOICE > General screens allow you to set up global SIP and Quality of Service (QoS) settings. VoIP (Voice over IP) is the sending of voice signals over the Internet Protocol. This allows you to make phone calls and send faxes over the Internet at a fraction of the cost of using the traditional circuit-switched telephone network. You can also use servers to run telephone service applications like PBX services and voice mail. Internet Telephony Service Provider (ITSP) companies provide VoIP service. A company could alternatively set up an IP-PBX and provide it's own VoIP service. Circuit-switched telephone networks require 64 kilobits per second (kbps) in each direction to handle a telephone call. VoIP can use advanced voice coding techniques with compression to reduce the required bandwidth. 9.1.1 What You Need to Know The following terms and concepts may help as you read through this chapter. Voice Coding A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital signals back into voice signals. The WiMAX Device supports the following codecs. • G.711 is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal amplitudes at regular time intervals (sampling) and converts them into digital bits (quantization). Quantization "reads" the analog signal and then "writes" it to the nearest digital value. For this reason, a digital sample is usually slightly different from its analog original (this difference is known as "quantization noise"). G.711 provides excellent sound quality but requires 64kbps of bandwidth. • G.729 is an Analysis-by-Synthesis (AbS) hybrid waveform codec. It uses a filter based on information about how the human vocal tract produces sounds. The codec analyzes the incoming voice signal and attempts to synthesize it using its list of voice elements. It tests the synthesized signal against the original and, if it is acceptable, transmits details of the voice elements it used to make the synthesis. Because the codec at the receiving end has the same list, it can exactly recreate the synthesized audio signal.G.729 provides good sound quality and reduces the required bandwidth to 8kbps. Quality of Service (QoS) Quality of Service (QoS) refers to both a network's ability to deliver data with minimum delay and the networking methods used to provide bandwidth for real-time multimedia applications. WiMAX Device Configuration User's Guide 151

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WiMAX Device Configuration User’s Guide
151
C
HAPTER
9
The VoIP General Screens
9.1
VoIP Overview
The features mentioned in this chapter are for models that has phone port(s) and you can make
telephone calls over the Internet using the WiMAX Device.
The
VOICE > General
screens allow you to set up global SIP and Quality of Service (QoS) settings.
VoIP (Voice over IP) is the sending of voice signals over the Internet Protocol. This allows you to
make phone calls and send faxes over the Internet at a fraction of the cost of using the traditional
circuit-switched telephone network. You can also use servers to run telephone service applications
like PBX services and voice mail. Internet Telephony Service Provider (ITSP) companies provide
VoIP service. A company could alternatively set up an IP-PBX and provide it’s own VoIP service.
Circuit-switched telephone networks require 64 kilobits per second (kbps) in each direction to
handle a telephone call. VoIP can use advanced voice coding techniques with compression to reduce
the required bandwidth.
9.1.1
What You Need to Know
The following terms and concepts may help as you read through this chapter.
Voice Coding
A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital
signals back into voice signals. The WiMAX Device supports the following codecs.
G.711
is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal
amplitudes at regular time intervals (sampling) and converts them into digital bits (quantization).
Quantization “reads” the analog signal and then “writes” it to the nearest digital value. For this
reason, a digital sample is usually slightly different from its analog original (this difference is
known as “quantization noise”). G.711 provides excellent sound quality but requires 64kbps of
bandwidth.
G.729
is an Analysis-by-Synthesis (AbS) hybrid waveform codec. It uses a filter based on
information about how the human vocal tract produces sounds. The codec analyzes the incoming
voice signal and attempts to synthesize it using its list of voice elements. It tests the synthesized
signal against the original and, if it is acceptable, transmits details of the voice elements it used
to make the synthesis. Because the codec at the receiving end has the same list, it can exactly
recreate the synthesized audio signal.G.729 provides good sound quality and reduces the
required bandwidth to 8kbps.
Quality of Service (QoS)
Quality of Service (QoS) refers to both a network's ability to deliver data with minimum delay and
the networking methods used to provide bandwidth for real-time multimedia applications.