Lessons from VoIP {Tools}

Concepts

• History of Voice over Internet Protocol (VoIP) according to Wikipedia
  • initiated in 1974 by IEEE publication of "A Protocol for Packet Network Interconnection."
• Public switched telephone network (PSTN) according to Wikipedia
  • "is the network of the world's public circuit-switched telephone networks, in much the same way that the Internet is the network of the world's public IP-based packet-switched networks."
• Softphone according to Wikipedia
  • "a software program for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware."
• Direct Inward Dialing (DID) according to Wikipedia
  • "feature offered by telephone companies for use with their customers' private branch exchange (PBX) systems."
• Interactive voice response (IVR) according to Wikipedia
  • "interactive technology that allows a computer to detect voice and keypad inputs."
• Plug-n-Dial
  • standards-based, feature-rich dialtone for your SIP hardware or softphone
• Mobile VoIP according to Wikipedia
  • "an extension of mobility to a Voice over IP network."
• Telecom Abbreviations according to Abbreviation.com

Tools

• software clients
  • Wikipedia list of SIP software
  • Wikipedia Comparison of VoIP software
  • sipdroid native SIP/VoIP client for Android
  • SIPsoftware with recommended software for Plug-n-Dial
• hardware
  • SIPhardware The VOIP hardware encyclopedia
• servers
  • Asterisk The Open Source Telephony Project
    • #asterisk IRC channel on freenode
    • Asterisk Gateway Interface (AGI) according to Wikipedia
      • "software interface and communications protocol for application level control of selected features of the Asterisk PBX."
  • OpenVXI portable open source library that interprets the VoiceXML dialog markup language
  • VoiceGlue wiki and its architecture diagram
  • CMUSphinx speech-using tools and applications
    • LIUM French dictionary with 65K words made automatically with lia_phon, from LIA labs
  • FreeSWITCH media server to host IVR applications using simple scripts or XML to control the callflow
    • #freeswitch and freeswitch-fr IRC channel on freenode
    • FreeSWITCH: The Story Behind The Software 2007
    • twitt requesting a bit of help to @mercutioviz
  • Cloudvox open phone API platform, in minutes (call your code: Asterisk/AGI, HTTP/JSON, Adhearsion/Ruby, Asterisk-Java)
  • Yate VoIP for Professionals
    • #yate IRC channel on freenode
• potential Voice Based Input solutions for Seedea
• whocallsme.com Phone Call Comments
  • user supplied database of phone numbers of telemarketers, non-profit organizations, charities, political surveyors, SCAM artists, and other companies that don't leave messages, disconnect once you answer, ignore the Do-Not-Call List regulations, and simply interrupt your day.
  • potential resource for filtering (a la Gmail bayesian spam filtering)
• GSM
  • OpenBSC GSM (Abis /BSC/MSC/HLR) network in a box software, implementing the minimal necessary parts to build a small, self-contained GSM network.
• Siproxd proxy/masquerading daemon for the SIP protocol.

Providers

• France
  • ippi
    • N° de client : 280869
    • Mon identifiant : utopiah
    • Mon adresse SIP : utopiah@ippi.fr
    • Mon numéro géo principal : 0177623990
  • OVH
• International DIDs
  • DID Service Providers, including free ones, by voip-info.org
  • bind DIDs from multiple countries to one PDX
  • route on language based on used DID
• call-in solutions
  • Gizmo5 - Callin Numbers
    • $35/year and not available in France as of October 2009
  • SkypeIn - Skype personal internet number
    • $60/year
  • Yahoo! Voice - PhoneIn
    • $29.90/year
• Google Voice (invite only as of October 2009)
  • formely known as GrandCentral: The New Way to Use Your Phones

My current configuration

• free geo french DID from ippi.fr
  • configured for Asterisk
  • note that ippi client is actually SIP Communicator configured and skinned
• SIP account on benetou.fr
  • Asterisk sip:ippi_phone@seedea.org
  • [HOW-TO] Installation & configuration d'Asterisk, Forum OVH 2008
    • encouraging to bind to the server IP rather than default 0.0.0.0
    • Installation asterisk sur serveur ovh, pk|concept 2008
  • FreeSWITCH (for it's bundled TTS/IVR capabilities)
    • Quick and Dirty Install on FreeSWITCH Wiki
    • IRC channels from the Community and Support section in the wiki
    • default test account
      • SIP identity = sip:1000@94.23.59.19
        • use your own user number between 1000 and 1019
        • replace by rps7452.ovh.net if you are on an IPv6 trunk
      • SIP proxy = sip:94.23.59.19:5060
        • replace by rps7452.ovh.net if you are on an IPv6 trunk
      • SIP password = 1234
      • extension 5000 for IVR test
      • extension 3000 for conference room
    • turned of as never used
      • mainly because of UDP tunneling problems behind wifi networks (thus too unstable)
  • check freephonie by Free for call-out
    • Provider Configuration: FreePhonie for FreeSWITCH
• Linphone open-source sip video-phone for linux and windows
  • note the CLI linphonec (package linphone-nox)
  • replacing SIP Communicator (heavy) Java client
• irssi (aka my integrated communication center ;)
  • /alias callread window goto phone; exec - tail /var/log/asterisk/cdr-csv/Master.csv; window last

Tasks

1. create my own account
   1. utopiah/fabien.benetou
2. configure my french local DID
   1. ippi.fr
3. configure freephonie for outgoing calls
   1. free account
4. use IVR
   1. Seedea services (cf API)
   2. redirect to the dedicated conference room (rather than the default 888 one)
   3. Speech Recognition/Text-to-Speech
      1. Mod PocketSphinx
5. automatically create accounts (API)
   1. for new Seedea accounts
6. add new DIDs
   1. Canada
   2. USA
   3. UK

Note

My notes on Tools gather what I know or want to know. Consequently they are not and will never be complete references. For this, official manuals and online communities provide much better answers.