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Mastering FreeSWITCH

You're reading from  Mastering FreeSWITCH

Product type Book
Published in Jul 2016
Publisher Packt
ISBN-13 9781784398880
Pages 300 pages
Edition 1st Edition
Authors (8):
Russell Treleaven Russell Treleaven
Profile icon Russell Treleaven
Seven Du Seven Du
Profile icon Seven Du
Darren Schreiber Darren Schreiber
Profile icon Darren Schreiber
Ken Rice Ken Rice
Profile icon Ken Rice
Mike Jerris Mike Jerris
Profile icon Mike Jerris
Kalyani Kulkarni Kalyani Kulkarni
Profile icon Kalyani Kulkarni
Florent Krieg Florent Krieg
Profile icon Florent Krieg
Charles Bujold Charles Bujold
Profile icon Charles Bujold
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Table of Contents (21) Chapters close

Mastering FreeSWITCH
About the Authors
About the Reviewers
1. Typical Voice Uses for FreeSWITCH 2. Deploying FreeSWITCH 3. ITSP and Voice Codecs Optimization 4. VoIP Security 5. Audio File and Streaming Formats, Music on Hold, Recording Calls 6. PSTN and TDM 7. WebRTC and Mod_Verto 8. Audio and Video Conferencing 9. Faxing and T38 10. Advanced IVR with Lua 11. Write Your FreeSWITCH Module in C 12. Tracing and Debugging VoIP 13. Homer, Monitoring and Troubleshooting Your Communication Platform Index

Traditional telephony codecs constrain audio

There are so many ways to compress and digitize audio to be sent through the wire. A lot of codecs are available for use with FreeSWITCH, from ultra-wide band high definition (the quality of an audio CD) to the ultra-low bandwidth utilization, and all the variables involved can be confusing.

So, let's start with a bold simplifying assumption (we'll see complexity later): You only need to be aware of two codecs — G711 (which is available in two flavors: Ulaw and Alaw, also known as PCMU and PCMA) and G729.

G711 is the original, uncompressed format used since the beginning of time by telecom companies worldwide. It was designed to carry speech so that it only gets a very narrow audio band (300-3400 Hz), and to cut out the rest (humans can hear from 20 to 20,000 Hz; that's why music on hold sounds so bad on the phone). It samples that narrow speech band 8,000 times per second (8 khz sampling) in a logarithmic way (mimicking human hearing for different...

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