Motorola VT1000 Uživatelský manuál

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

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

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

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

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

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

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

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

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

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

●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-

POTS World – Ma BellCentralOfficeTelephoneCompanyWireHome WiringCentralOfficeTelephoneCompanyWireHome WiringNetworkInterfaceDevicePoint ofDemarcationN

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

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

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

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

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

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

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

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

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

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

POTS World - TodayCentralOfficeTelephoneCompanyWireHome WiringCentralOfficeNetworkInterfaceDeviceILECCLECIXC*LECIncumbent Local Exchange CarrierCompet

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

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

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

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

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

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

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

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

Networked WorldCentralOfficeEthernetEthernetCoaxialCableADSLRouterCableModemCableHead EndInternetSeverNIDHome WiringISP

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

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

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

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