Real-time%20multimedia%20and%20communication%20in%20packet%20networks

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Real-time multimedia and communication in packet
networks
Asterisk The open source IP PBX
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Some House Rules

• Practical component of the course
• Workings and power of asterisk, an IP Private
  Branch eXchange (PBX)
• Small tutorials will be given on a daily basis
  before each lecture
• Large practical write your own application that
  adds value to an Asterisk PBX
• This will be demonstrated to the class at the end
  of the course.
• Practical to be done on a Linux machine you can
  ssh into cc

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Some Admin
You should have by now - found your extension
on pbx.ict.ru.ac.za - registered on iLanga your
two phones (sj and hardphone) - explored the
messaging between the two phones and the SIP
proxy server in iLanga at least in these
situations 1. Registration 2. Call
establishment with callee answering and without
answer (voicemail) - checked the media stream in
iLanga and discovered possible difference with
respect to the case of calling directly an end
point. What happens if you call a telephone
registered with iLanga directly via its IP
number? Please make sure that problems that you
had yesterday are cleared in the first part of
the lecture. Enjoy Asterisk!
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Some House Rules

• Practical component of the course
• Workings and power of asterisk, an IP Private
  Branch eXchange (PBX)
• Small tutorials will be given on a daily basis
  before each lecture
• Large practical write your own application that
  adds value to an Asterisk PBX
• This will be demonstrated to the class at the end
  of the course.
• Practical to be done on a Linux machine you can
  ssh into cc

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What is Asterisk ()?

• A private or enterprise grade exchange generally
  referred to as a private branch exchange (PBX)?
• Designed to interface telephony hardware or
  software with any telephony application
  seamlessly and consistently
• i.e. Asterisk can be moulded to fit any telephony
  application
• Asterisk can be used in any of these applications
• Heterogeneous Voice over IP gateway (MGCP, SIP,
  IAX, H.323)?
• Private Branch eXchange (PBX)?
• Custom Interactive Voice Response (IVR) server
• Conferencing server
• Softswitch?

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What is Asterisk ()?

• A private or enterprise grade exchange generally
  referred to as a private branch exchange (PBX)?
• Designed to interface telephony hardware or
  software with any telephony application
  seamlessly and consistently
• i.e. Asterisk can be moulded to fit any telephony
  application
• Asterisk can be used in any of these applications
• Heterogeneous Voice over IP gateway (MGCP, SIP,
  IAX, H.323)?
• Private Branch eXchange (PBX)?
• Custom Interactive Voice Response (IVR) server
• Conferencing server
• Softswitch?

7
What is Asterisk ()?

• A private or enterprise grade exchange generally
  referred to as a private branch exchange (PBX)?
• Designed to interface telephony hardware or
  software with any telephony application
  seamlessly and consistently
• i.e. Asterisk can be moulded to fit any telephony
  application
• Asterisk can be used in any of these applications
• Heterogeneous Voice over IP gateway (MGCP, SIP,
  IAX, H.323)?
• Private Branch eXchange (PBX)?
• Custom Interactive Voice Response (IVR) server
• Conferencing server
• Softswitch?

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Asterisk Supported Communication Technologies

• Asterisk is designed to allow new interfaces and
  technologies to be added easily
• Asterisks goal is to support every kind of
  telephony technology possible
• Asterisk interfaces divided into 3
• Zaptel hardware
• Non-Zaptel hardware
• Packet voice

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Zaptel Hardware

• Check out http//www.zapatatelephony.org/
• Provide integration with traditional and legacy
  analogue and digital telephone interfaces
• Zaptel interfaces available from Digium
  (www.digiumcards.com)?
• Zaptel interfaces available for a number of
  telephony interfaces
• ISDN Basic Rate Interface (BRI)?
• ISDN Primary Rate Interface (PRI)?
• Analog FXS interface connect to a station i.e.
  analogue phone
• Analog FXO interface connect to an office i.e.
  PBX

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Zaptel Hardware
Digium 4 x FXS card 342 USD
Digium 2 x FXS, 2 x FXO card 360 USD
Digium BRI card 469 USD
Digium BRI card 1345 USD
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Packet Voice Protocols

• Standard protocols for communication over packet
  networks
• Only interfaces that do not require specialised
  hardware
• E.g.
• SIP
• IAX
• H.323
• IAX
• MGCP

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Asterisks Architecture

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Modules and Applications

• Asterisks core contains several engines that
  play a critical role in the softwares operation
• At startup, Dynamic module loader loads various
  modules for
• Channel drivers
• File formats
• Codecs
• Applications
• Custom applications launcher i.e. the iLanga
  Prepaid Application
• Asterisks switching core accepts calls from any
  of the various interfaces and routes them
  according to the dialplan
• Codec translator permits channels which are
  compressed with different codecs to talk to each
  other
• Scheduler and IO Manager which can be used by
  applications and drivers

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Asterisks Architecture

• Modular API for Asterisk responsible for
  Asterisks success
• Channel API, File Format API, Codec API,
  Application API

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Some Asterisk configurations (basic)?

• Asterisk box contains
• 1 analog interface for telephone (FXS interface)?
• 1 analog interface to PSTN (FXO interface)?
• Ethernet interface for VoIP

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Some Asterisk configurations

• Asterisk box contains
• One E1 or (PRI) interface connected to a digital
  to analog converter or channel bank
• 15 phones connected channel bank
• 15 lines to PSTN (i.e. Telkom)?

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Some Asterisk configurations

• In this example we illustrate the possibility of
  distributing a number of Asterisk boxes
• Each Asterisk box can be interconnected using
• TDM technology e.g. BRI or PRI
• Data technology/VoIP e.g. Inter Asterisk Exchange
  (IAX)?

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Asterisk Filesystem Organisation

• /etc/asterisk
• Contains Asterisk configuration files NB
  directory
• /usr/sbin
• Contains Asterisk binaries
• /usr/lib/asterisk/modules
• Contains runtime modules for channel drivers,
  codecs, file formats, applications
• /usr/include/asterisk
• Contains Asterisk C header files for the building
  the software
• /var/lib/asterisk/agi-bin
• Location of Asterisk Gateway Interface (AGI) for
  use in dialplan

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Asterisk Filesystem Organisation

• /var/lib/asterisk/astdb
• Asterisk internal database
• Roughly equivalent to Windows registry
• /var/lib/asterisk/mohmp3
• Storage directory mp3s used for music on hold
• /var/lib/asterisk/sounds
• Storage directory for Asterisk audio files e.g.
  voice prompts to be used in IVR menus
• /var/spool/asterisk/outgoing
• Spooling directory for making outgoing calls
• Can be used for callback function
• /var/spool/asterisk/voicemail
• Storage directory for Asterisk voicemail boxes,
  announcements, etc

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Asterisk Channels

• Channel naming convention in Asterisk is standard
• Outgoing channel names (used in Dial application)
  named in format
• lttechnologygt/ltdialstringgt
• lttechnologygt represents type of interface you
  want to address or use
• E.g. Zap, SIP, IAX2, etc
• ltdialstringgt is a driver-specific string
  representing destination desired

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Asterisk Channels (Zap)?

• lttechnologygt/ltdialstringgt
• Zap / g ltidentifiergt
• ltidentifiergt number of the channel you are
  trying to address
• If ltidentifiergt prefixed by g then number is
  interpreted as a group instead of as a channel
• e.g.
• Zap/g1/0027466223458 (Any available line in group
  1)?
• Zap/1/0027466223458 (TDM channel 1)?

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Asterisk Channels (SIP)?

• Outgoing channels typical of the form
• SIP / exten_at_ ltdomaingt ltportnogt
• E.g.
• SIP/mos
• SIP3000_at_sip.ict.ru.ac.za
• SIP/3000_at_sip.ict.ru.ac.za5060

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Asterisk Channels (IAX)?

• IAX2 / ltusergt ltsecretgt _at_ ltdomaingt
  ltportnogt /ltextengt_at_ltcontextgt/ltoptionsgt
• Where ltusergt and ltsecretgt are optional username
  and secret to connect to the host identified by
  ltpeergt and
• ltportnogt optional port number,
• ltextengt specific extension at an optional
  context ltcontextgt, and optionally with ltoptionsgt
  connection options
• E.g
• IAX2/authnamesecretpass_at_voiptalk.org/442081234567
  _at_default
• Call to voiptalk.org using authname as username
  and secretpass as password, and requesting
  extension 442081234567 in default context

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Running Asterisk and Environment

• Asterisk can be run in console mode or as a
  daemon process
• E.g. asterisk vvvgc (console mode with verbose3
  debugging
• Asterisk (daemon) started by typing asterisk
• Please always run asterisk as daemon and connect
  to daemon process using
• asterisk r
• asterisk -vvvvvr
• When connecting to daemon process you will be
  connected to the command line interface of
  Astrerisk (CLI)?
• vitalstatistixCLIgt

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Asterisk CLI

• When connected to the Asterisk CLI there are a
  number of commands you can use
• Go and test them out, see what they do,
  familiarise yourself with the environment
• E.g.
• help
• show applications
• show application x
• show codecs'
• show translation'
• extensions reload
• sip reload
• CLI include command completion via the tab key

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sip.conf

• Please set your phones up to connect to your
  development box
• Box
• IP 146.231.121.165
• Create a sip.conf file in your home directory,
  you can use the reference http//www.voip-info.org
  /wiki/view/Asteriskconfigsip.conf

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Tomorrows Tutorial

1. Create an account for your phone
2. Play around with some of the settings in the
   sip.conf file