Asterisk Overview

Asterisk Overview
- What is Asterisk ?
  - A PBX
  - An IVR Platform
  - FreeBSD friendly - version 1.4 compiles without patches
- Usage Scenarios
  - Home VoIP PBX
    - Saves costs on long distance calls
    - Flexible Answering Machine
  - Office VoIP PBX
    - Free voice communication between offices
    - Support Call Center and Queues
    - Saves on telephone costs
      - ENUM
      - Other VoIP enabled offices
    - Flexible customer support
    - Video and Audio Conferencing
  - VoIP Termination Services
    - Appropriate for INET Providers
    - Offer telephone services to existing customers
    - Serve business customers with VoIP traffic
  - VoIP Gateway
    - Ported on different platforms
    - Usable on embedded devices
  - Usage Scenarios Diagram
- Capabilities
  - Channel Protocols
    - H323
      - Old, but still usable and popular VoIP protocol
      - TCP Signaling with RTP Media
    - SIP
      - Session Initiation Protocol
      - De-fakto standart for VoIP
      - Not NAT Friendly
        STUN
        ICE
        X-Tunnels
      - HTTP Signaling
        UDP
        TCP
      - Contact Presence support
      - RTP Media Stream
      - Relatively Complex, implementations are getting better
    - IAX2
      - Inter-Asterisk eXchange Protocol Ver. 2
      - Asterisk-native communication protocol
      - Signaling and Media Stream use single pair of UDP ports
      - NAT-Friendly
      - Trunk support, more efficient for simultaneous calls
    - XMPP
      - Not a voip standard, but generally useable for signalling
      - Google's Jingle extensions used for VoIP (GTAlk, chan_jabber)
      - Messaging
    - MGCP
      - Media Gateway Control Protocol
      - Client Server model for distributed VoIP Networks with access to PSTN
      - UDP Based
  - Codecs
    - Transcodable
      - Only narrowband codecs
      - Stream is decoded and coded between peers
      - Audio G711(U/ALAW),GSM,iLBC,Speex, G729(with license per channel basis)
      - LPC-10, Linear, etc
    - Passthrough
      - Stream is only piped between peers
      - Audio G·723
      - Video H.263, H.263+, H.264
  - Dialplans
    - Pattern-based matching
    - Macros
    - AEL for extension "programming"
    - ENUM - shared number maps
    - DUNDI - distributed dialplan sharing for Asterisk
  - Call Detail Records (CDR)
    - CSV call log
    - Support for ODBC, postresql backends
    - Support for FreeTDS (tabular data stream)
  - Hardware
    - Interface Cards
    - Channel Banks
    - ISDN Devices (highly dependable on chipset + driver)
    - etc
    - more info: http://voip-info.org/wiki-asterisk+hardware
- Architecture
  - Overall Architecture Diagram
    - - Resource: digium.com
  - Module Types
    - app
      - pluggable commands
      - the native way of integrating external programs in asterisk (no-AGI)
    - pbx
      - inner parts of asterisk are defined as modules
      - realtime, configuration, etc
    - codec
      - coding/decoding capabilities
      - operate mainly on the payload of media stream
    - format
      - handles media format
      - pcm,gsm,wav, etc
    - channel
      - protocols for communication
      - responsible for call state representation
    - cdr
      - resources and managers for storing call records
      - odbc, postgre, mysql
  - Asterisk Realtime
    - Asterisk Configuration mapped onto abstract database schema
    - Database support ODBC, MySQL support
    - Easifies billing and tracking
    - Allows flexible integration with existing business apps
  - Local Integration
    - AGI
      - Asterisk Gateway Interface, implemented as app_
      - allows execution of AGI-capable local processes
      - accessible dialplan
    - EAGI
      - Inherits AGI functionality
      - extended with access to the audio channel
  - Remote access
    - Manager API
      - TCP based control interface
      - Call state monitoring
      - Call origination, transfer, answer, hold, parking, etc
      - Existing bindings for perl, python, java
    - FastAGI
      - remote access to the AGI interface
      - Implementations for perl, python, java
  - Sample Integration Scenario
- Applications
  - Dial
  - Answer
  - Hangup
  - Transfer
  - MusicOnHold
  - Background
  - WaitExten
  - Say
  - Agi
  - many many more
  - resource: http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands
- Configuration Basics
  - Asterisk Configuration
    - modules.conf
    - rtp.conf
    - musiconhold.conf
  - Peer configuration
    - sip.conf
    - iax.conf
  - Extenions Configuration
    - extensions.conf
    - AEL2
      - Programming language for flexible dialplanning
      - Dynamic extension handling
      - Loops, Conditions, Subroutines
      - More flexible than extensions.conf
  - Configuration Dependencies
- Available Service Provisiders
  - Proxying/Brokering
    - FWD
      - SIP Proxying, first of the free voip networks
      - IAX access
      - US Toll-Free Calls
    - SIPBroker
      - Peered with hundreds SIP Providers
      - Access by prefix
      - Free brokering
      - Free ENUM Support
        enum.org
        e164.org
        others
    - e164.org
      - Free ENUM mapping of regular phone numers to alternative communication addresses
      - Works over DNS, highly scalable, free and open
      - Email, ICQ, SIP, IAX, XMPP
    - Many others - more info - voip-info.org
  - Termination
    - MutualPhone
      - Cheap A-Z SIP Termination
      - Apparantly lower delay overseas
    - VoipJet
      - IAX-Only A-Z Termination
      - Low rates, US low delay, Europe not that well
    - There are hundreds more
  - DID
    - IPKall
      - Free US Washington State DIDs
      - Forwarding to any SIP address
    - Voipuser
      - Free UK DIDs, forwardable to any SIP/IAX
      - SIP Outbound calls
      - IAX Termination
      - UK-Active users may generally profit from community jar of free minutes
- End User Clients
  - Sofphones
    - SIP
      - Linphone
        Unix/linux
        GTK + libosip
        No STUN Support
      - KPhone
        Unix/linux
        KDE + libosip
        No STUN Support
        Apparently Video Support
      - Eyebeam (X-Lite)
        Excellent Windows Support, bad Linux/Mac
        Freeware scaled-down version available
        Full support of STUN/ICE/Tunnels
        Video (freeware), Video Conferencing (paid versoin)
        H263, H263+, H264
        Most of the audio codecs available for free
        Not freely rebrandable, no skins
      - SJPhone
        Excellent Windows, limited Mac and Linux support
        STUN Support
        Adware version available
        Excellent Freeware (no adware) WinCE Version
        Rebrandable, Skinnable User Interface
      - More
    - IAX2
      - IAXComm
        Basic UI Interface
        Windows, Linux, Mac support
        Uses wxWindows + libiaxclient
        Free, Open Source, GPL
      - Kiax
        Simple User Interface
        Windows, Linux, FreeBSD support
        Uses libqt3 + libiaxclient
        Free, Open Source, GPL
      - QTIax
        Linux
        Minimalistic UI
        qtlib + stripped down version of iaxclient
        GPL
      - Firefly
        Qute UI
        Only Windows
        Based on libiax2
        Freeware
      - More
  - ATA Devices
    - Grandstream ATA 486
    - Many Others (also in combination with WLAN Routers)
  - SIP Phones
- References
  - asterisk.org - asterisk download, resources
  - voip-info.org (The ultimate chaotic VoIP resource)
  - asteriskdocs.org - detailed asterisk documentation
  - others