Motorola VT1000 Manuel d'utilisateur

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Page 1 - VOIP, Linux, and Asterisk

VOIP, Linux, and AsteriskMaking Beautiful Voice TogetherDaryll StraussPresidentDigital OrdnanceSCALE 3xFeb 13th, 2005

Page 2 - POTS World – Ma Bell

VOIP Encoding●Voice is digitized and compressed for transmission.●Each voice channel requires some bandwidth.●Converting between encodings is called t

Page 3 - POTS World - Today

Network Protocols●Network Adress Translation – Allow multiple machines to share on network address●Quality of Service – A protocol for prioritizing ne

Page 4 - Channels and one D Channel

Starting to VOIPISPVOIPProviderISP●Headset is highly recommended forbetter voice quality●VOIP Providers – Free World Dialup, Sipphone, Earthlink, orSk

Page 5 - Networked World

Making a SIP CallISPVOIPProviderISP●Register your SIP device. Let a proxy server know you're thereso that it can ring you.●Dial a SIP URL (or a n

Page 6

ISPVOIPProviderInternetPSTNInterfaceProviderPSTNInternet●Some providers will route PSTN callsto your SIP phone number for free●No choice of phone numb

Page 7 - VOIP Gear

ISPEthernetVOIPProviderInternetInternet●There are many residential VOIPproviders. (Vonage, Broadvoice,packet8, VoicePulse, Sipphone, etc)●You connect

Page 8

ISPEthernetVOIPProviderInternetInternet●If possible calls are sent entirelyvia the internet.●If not, then they are routed via theInternet to the close

Page 9 - VOIP Protocols

ISPEthernetVOIPProviderInternetInternet●Add a device that supports an FXOport and it can be connected to thelocal exchange carrier.●Sipura 3000 is an

Page 10 - VOIP Encoding

Asterisk●Asterisk can speak SIP, IAX, and H323over an ethernet port●Asterisk supports cards that talk to analog lines via FXO or FXS●Asterisk allows m

Page 11 - Network Protocols

●Configure Asterisk to register withFWD using IAX●Configure Asterisk to play a soundwhen it receives a call●Use a soft phone with FWD to callAsterisk-

Page 12 - Starting to VOIP

POTS World – Ma BellCentralOfficeTelephoneCompanyWireHome WiringCentralOfficeTelephoneCompanyWireHome WiringNetworkInterfaceDevicePoint ofDemarcationN

Page 13 - Making a SIP Call

Config Files[general]bandwidth=lowdisallow=lpc10 ; Icky sound quality... Mr. Roboto.allow=ulawallow=gsmallow=alawallow=ilbcallow=adp

Page 14 - PSTN to VOIP

[general]format=wav49|gsm|wavservermail=asteriskattach=yesmaxsilence=10silencethreshld=128maxlogins=3fromstring=Digital Ordnance Voicemailpagerfromstr

Page 15 - Replace a Phone

●Soft phones●ATA's with analog phones●SIP phones●Analog phones into cards●VOIP Providers over ethernet●PSTN connection via cards●PSTN via gateway

Page 16 - Replace a Phone (cont)

[extensions]exten => 201,1,Macro(stdexten,201)exten => 202,1,Macro(stdexten,202)exten => 444,1,Meetme(1234)[fwd-forced]exten => _7.,1,Macr

Page 17 - Connecting Your PSTN and VOIP

Interfacing With Asterisk[general]disallow=all ; Disallow all codecsallow=gsmallow=ilbcallow=adpcmallow=ulawallow=alawdtmfmode=rfc2833srvlookup=yesreg

Page 18 - Asterisk

Additional Features●Asterisk can monitor and record calls●Asterisk can provide features, like putting calls on hold, even if the phone doesn't su

Page 19 - First Tests With Asterisk

Going Beyond Your Father's PBX●Asterisk can read/write values from/to a database●Asterisk can send data to/read data from from an application●Ast

Page 20 - Config Files

Example Applications●Credit card/Prepaid calling●Dating service●Live chat●Follow me●Call center (Asterisk agents)●Games (Lost Vault, Taboo)●Training●V

Page 21

Gotchas●SIP behind NAT is hard, because SIP encodes RTP port numbers in packets. Use IAX or a Virtual Private Network to tunnel behind a NAT. Simple T

Page 22 - Interfacing With Asterisk

Gotchas (cont)●Asterisk doesn't support SIP URLs well.●Learning curve is steep – read the docs,take small steps and test changes.●Overloading the

Page 23

POTS World - TodayCentralOfficeTelephoneCompanyWireHome WiringCentralOfficeNetworkInterfaceDeviceILECCLECIXC*LECIncumbent Local Exchange CarrierCompet

Page 24

Gotchas (cont)●Network traffic can cause you to loose quality. QoS can prioritize voice traffic over data. Consider private/VLAN voice ethernet.●Fax a

Page 25 - Additional Features

Asterisk Add Ons●ASTMan is manager that lets you manipulate Asterisk while it is running via a network connection.●AMP is GUI for configuring Asterisk

Page 26

Other Open Source VOIP Systems●SIP Express Router – A SIP processor that does not handle the media stream. Scales to very large numbers of users. SER

Page 27 - Example Applications

A Brave New WorldQ: Why do we use phone numbers?A: SIP URLs are easier to remember. SRV records allow you to do that.Q: How do I know if a phone numbe

Page 28

Conclusions●My goal was to introduce you to telephony and VOIP. Teach you the basic terminology.●Give you examples you can do yourself for very little

Page 29 - Gotchas (cont)

Q&ADon't forget the VOIP panel at 3:00 today.

Page 30

ResourcesWebsites:http://www.voxilla.comhttp://www.asterisk.orghttp://www.voip-info.orghttp://www.asteriskdocs.orgMailing Lists:asterisk-users mailing

Page 31 - Asterisk Add Ons

Connections to the Telephone Company●Analog phone lines●ISDN – Digital phone lines. Two B Channels for voice and one D Channel for control●Primary Rat

Page 32

Networked WorldCentralOfficeEthernetEthernetCoaxialCableADSLRouterCableModemCableHead EndInternetSeverNIDHome WiringISP

Page 33 - A Brave New World

Crossover Into Voice Over Internet Protocol●VOIP crosses over between the Internet and the PSTN at several possible locations●Intraoffice – VOIP phone

Page 34 - Conclusions

VOIP Gear●Foreign eXchange Station – analog telephone●Foreign eXchange Office – Device that to phones●Analog Telephone Adapter – An interface with eth

Page 35

VOIP Gear●Portable Branch eXchange – A local telephone switch●Interactive Voice Response – A voice menu●Key System – A type of PBX that tightly tracks

Page 36 - Resources

VOIP Protocols●Session Initiation Protocol – Manages a phone connection●Realtime Transport Protocol – Carries the voice data●Inter Asterisk eXchange –

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