Kommunikatsiooniteenuste arendus IRT0080

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Kommunikatsiooniteenuste arendusIRT0080

• Loeng 10, telefonisõlmedAvo
  Otstelekommunikatsiooni õppetool,
• TTÜ raadio- ja sidetehnika inst.avo.ots_at_ttu.ee

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Voice Telephony Voice Mail Unified
Messaging Conferencing Find-Me-Follow-Me Call
Center
Presence
Video Messaging/Mail Conferencing Surveillance
Integration WEB-Portal, Desktop, Devices and
Server-Business Process
Collaboration Instant Messaging eMeetings Web
Conferencing
Business Apps CRM, Supply Chain, Call Center
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What is Asterisk

• Asterisk is a PBX replacement system, designed to
  reproduce the features of standard office phone
  systems. Asterisk is also a Voice over IP
  toolkit which allows interaction between these
  PBX features and IP-based networks (local and
  remote.) Asterisk is hardware independent, and
  is designed to run on OSS operating systems.

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Goals of Asterisk

• Provide Open-Source implementations of basic PBX
  functionality
• Be vendor neutral (despite last point here)
• Be as all-encompassing as possible for features
• Be flexible and provide hooks for advanced
  features
• Move proprietary hardware features into open
  source software
• Sell TDM hardware cards for Digium

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Channel types VoIP

• SIP - Session Initiation Protocol
• H.323
• MGCP - Media Gateway Control Protocol
• SCCP - Skinny Client Control Protocol (Cisco)
• All of these use UDP for setup/transport except
  for SCCP, which uses a mix of UDP/TCP

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Channel types - non-VoIP

• TDM POTS cards (Digium, Zapata, Voicetronix,
  etc.)
• TDM Digital (AdTran VoFR, Digium E1/T1, etc.)
• All TDM cards require install of Zaptel driver
  suite
• CAPI (ISDN card support for Linux ISDN driver)
• USB dongle for FXS
• Modem drivers for certain modems (yuck)
• Speaker/headphones (dont try this at home, kids)

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Some Applications

• Dial - connects an inbound call with some other
  channel. One specifies the technology (SIP, Zap,
  H323, etc.) the number to be dialed, the
  Ring-No-Answer delay, and options (if desired)
• exten gt 1234,1,Dial(SIP/1234,25)
• exten gt 1234,2,Voicemail2(u1234)

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Some Applications (contd)

• Playback(filename)
• Plays a sound file in .gsm format
• Background(filename)
• Plays a sound file while listening for DTMF
  (touch tone) input
• test
• exten gt 123,1,Background(press-a-number)
• exten gt 123,2,Goto(1)
• exten gt _X,1,SayDigits(EXTEN)

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Some Applications (contd)

• MeetMe(conf)
• Adds the caller to a conference room (optionally
  muted or unmuted)
• Monitor
• Records channel (in and out) to .wav or .gsm
  files
• PrivacyManager
• Forces anonymous calls to enter valid ANI

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Some Applications (contd)

• DISA
• Lets callers from one channel get dialtone on
  another channel
• SetMusicOnHold
• You can specify .mp3 files as music on hold
  selections (random or sequential)
• MP3Player
• Fairly useless, but fun. You can specify files
  or streams of .mp3 to be played to callers.

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Some Applications (contd)

• There are over 80 different applications in the
  system - no time to talk about them all
• Applications are easily created and added if
  youre a decent C coder
• Channels are generic, so you dont have to know
  about any of the ugly VoIP or TDM stuff

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Voicemail

• Voicemail can be sent as email as well as stored
  on disk (1 minute 100kb)
• Short pages can be sent to email addresses when
  VM received
• Customizable timezones and time readouts per user
  - supports multiple languages
• .wav, .gsm file storage or email
• Dial by name directory hinges on VM data

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Practical Uses (home)

• Ditch your long distance company! SIP long
  distance (domestic and int) providers starting to
  crop up at low rates. Use Asterisk to gateway to
  them.
• Prevent phone spam! Callers with no CID get
  ditched.
• Filter your phone lines. Allow or forward
  callers who are on priority lists based on ANI.

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Practical Uses (office)

• Ditch your LD company (see prior slide)
• Interconnect office PBXs at zero network cost
• Get Unified Messaging
• Give ubiquitous access to the PBX for
  home/travelling employees
• Disaster recovery scenarios
• Move phones into your IT department and away from
  your expensive PBX consulting firm
• Eliminate adds/moves/changes as physical chores

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Advanced Topics

• Call queues - you can build a call center with
  Asterisk, with various call weightings and agent
  logins/hot seating
• Multi-ring, cascading ring with different
  technologies (inbound calls forward to your desk
  line and your cell phone - first answer gets it)
• Multi-language support with same dialplan
• Festival integration for voice synthesis

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Really Advanced Topics

• Manager interface TCP socket based interface for
  controlling and monitoring the system meant for
  automated manager tools (see gastman)
• AGI scripts built-in scriptable hooks to allow
  passing of data back and forth between Asterisk
  and external programs.
• Asterisk.pm - Perl module that works with AGI to
  handle gruntwork of call handling

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Really Advanced Topics(contd)

• Sybase and MySQL modules
• CDR (call detail record) output can be customized
  or put into database instead of flat file
• Use IAX2 trunk mode to get up to 200 more calls
  in the same bandwidth as other VoIP systems
• Route your calls to least-cost providers

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Crazy Extra Stuff That Didnt Fit

• Can run PPP or HDLC over channels - Asterisk can
  be a RAS server or a router (masochism)
• Can use speaker/microphone as a phone line
• Can do video calls or conferencing
• ENUM e.164 DNS-based call routing
• E.G. 2.1.2.1.2.5.4.3.0.5.1.e164.arpa.
• TDM over ethernet for front-end processing

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SIP Specifications Supported
RFC 2617 - HTTP Authentication Basic and Digest
Access Authentication
RFC 3261 - Session Initiation Protocol
RFC 2705 Media Gateway Control Protocol (used
for Digit Map implementation)
RFC 2833 - RTP Payload for DTMF Digits
RFC 3265 - SIP-Specific Event Notification
draft-ietf-sip-refer-07-Refer-To Header
RFC 1321 - MD5 Message-Digest Algorithm
RFC 3264 - An Offer/Answer Model with SDP
RFC 783 - TFTP Protocol (used for transferal of
configuration files to the gateway)
draft-ietf-sipping-mwi-01 - Message Waiting
Indication
draft-burger-sipping-netann-05
draft-ietf-sipping-cc-transfer-01
RFC 2327 - SDP Session Description Protocol
draft-ietf-sipping-dialog-package-01
draft-ietf-sipping-service-examples-04
RFC 1889 - RTP Transport Protocol for Real-Time
Applications
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Lingid

• http//nerdvittles.com/
• http//www.trixbox.org/
• http//www.counterpath.com/
• http//www.loligo.com/asterisk/
• http//www.onlamp.com
• http//www.voip911.gov/
• http//www.e164.org/

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Presentatsioonid

• http//ws.edu.isoc.org/data/2006/12675549354482287
  a4f488/telephony.ppt
• http//www.educause.edu/upload/presentations/E06/S
  ESS072/Production20Quality20Open20Source20VoIP
  .ppt