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Telecom:IPT:Resources

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VoIP resources

  • Heavyreading report index [1]
  • Miercom article index [2]
  • Miercom reports index [3]
  • Large directory of VoIP resources from Packetizer [4]
  • VoIP - Vulnerability over Internet Protocol [5]
  • VoIP resources on TechTarget [6]
  • VDC white paper on SIP Enabling technologies and infrastructure market trends (2005) [7]
  • The business case and business models for IP Telephony [8]
  • VoIP Technologies and Services - in-depth market and technology analysis [9] (local copy)
  • IP Telephony (VoIP) - excellent technology primer (local copy)
  • Essentials of real-time networking - excellent primer on VoIP technology [10] (local copy)
  • Implementing Voice over IP: Chapter 6: VoIP Deployment in Enterprises [11] (local copy)
  • Klaus Darilion's VoIP bookmarks [12]
  • An in-depth overview of Telecommunications [13]
  • "The rape of Ma-Bell" - an homage to what Bell used to be and the deregulation act [14]
  • Downloadable tutorials from iec.org - many on IP Telephony topics [15]
  • Raj Jain VoIP and IP Telephony References [16]
  • IHETS IP Telephony Task Force report:
    • In-depth analysis of VoIP market and technology [17]
    • IP Telephony standards and test planning [18]
    • Site survey methodology for Cisco, Nortel, Avaya and Mitel solutions [19]
  • Henning Schulzrinne "Practical Handbook of Internet Computing" - IP Telephony [20]
  • Peter Welcher's seminars [21]
  • Cisco IOS Configuration Fundamentals Configuration Guide [22]
  • IP Communications in the Enterprise:.Separating Productivity from Connectivity - a Jeff Pulver presentation [23]
  • IP Pulse VoIP & IPTV magazines [24]
  • IP Telephony resources (partly outdated) [25]
  • VoIP Loop - views on Enterprise IP Telephony [26]
  • Audiocodes whitepapers on VoIP/FoIP technology [27]
  • RADCOM VoIP protocols complete reference [28]
  • Ethereal page of Voice over IP protocol families [29]
  • VoIP protocols overview - Telus presentation to CRTC [30]
  • Primer on VoIP protocols [31]
  • ITPRC - H.323/MGCP/SIP/QSIG Resources [32]
  • Integration Resources for Data, Voice and Video [33]
  • DataConnection Whitepapers [34]
  • Telephonie 0-49 - VoIP resources
  • The Voip News channel - daily updates on IP Telephony and related technology topics
  • The Telecom Convergence resources page
  • IP Telephony news and articles courtesy of TMCNet


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VoIP service providers

  • Free World DialUp - free provider of IAX2 and SIP connectivity with Asterisk support
  • IAXTel - free provider of IAX2 connectivity with Asterisk support
  • GlobalVillage - free worldwide calling VoIP, no Asterisk support
  • VoipBuster - provider of SIP connectivity with PSTN termination
  • DIDWW - provider of inbound PSTN call bridging to SIP/IAX2, offers DIDs in major Canadian and US area-codes (as well as worldwide), supports Asterisk, interfacing to GTalk, MSN,Skype and many SIP providers like FWD, VoipBuster or Unlimitel
  • Unlimitel - a Canadian reseller of the Primus VoIP service, supports Asterisk and offers DIDs in major Canadian area-codes (more info)
  • GTalk-to-VoIP - a VoIP bridging service from GTalk to PSTN as well as to SIP-based and proprietary VoIP solutions (Yahoo Messenger, MSN/Live Messenger)
  • Text2it - a callback-based service that makes long distance calling affordable (5.7c/minute). Calls can be initiated from a mobile using SMS messaging, or from a Web browser.
  • Whaleback Systems - provider of Crystal Blue Voice Service, a premises-based managed IPT service for SMB's at an affordable price


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Comparing IP Telephony Solutions

  • Comparison of small IP-PBX [35]
  • Comparison of small and mid-size IP-PBX [36]
  • Comparison of mid-size IP-PBX [37]
  • Comparison of large IP-PBX [38] [39]
  • Comparative report of large IP-PBX [40]
  • Comparison of High-end IP-PBX [41]
  • Comparison of SIP IP-PBX [42]
  • BCR Best-in-test SME IP-PBX [43]
  • Competitive analysis of media gateways [44] [45]
  • Competitive analysis of softswitches [46] [47]
  • NGN VoIP service provider architecture and information [48] courtesy of Packetizer
  • Primer on SIP-based IP Telephony by Henning Schulzrinne [49]
  • Comparison of IP Telephony solutions on the market [50]


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H.323 video-conferencing

  • H.323 Videoconferencing [51]
  • Global Dialing Scheme (GDS) [52] [53]
  • Video-conferencing resources [54]
  • Video Compression Links [55]
  • MPEG-4 and H.261/263 Video Compression [56]
  • H.263, H.263v2, and H.26L [57]
  • RFC 2190 - RTP Payload Format for H.263 Video Streams [58]
  • National Radio Astronomy Observatory - VideoConferencing Manual [59]


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IPTV resources



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IP telephony troubleshooting

  • How To Debug and Troubleshoot VOIP [60]
  • Case Study - Clarus IPC Certification helps IBM Global Services [61]
  • Network monitoring tools for VoIP [62]
  • Cisco IOS Voice Troubleshooting and Monitoring guide [63]
  • Methods for building customer confidence in VoIP Systems with ClarusIPC


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Impairment and load simulation tools

  • NIST Net emulation package - VoIP impairment tools [64]
  • UDP packet reflector/forwarder - impairment tool [65]
  • PNT - XML-driven SIP bulk-call generator for IVR testing [66]


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VoIP troubleshooting with Ethereal

Cain & Abel would scans an ethereal captured file for voip communications and save voice streams them in different files (wav or au).


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Fax over Packet and Fax technology

  • Facsimile(FAX) primer [74]
  • T.37/T.38 Fax Gateway for Cisco Routers [75]
  • FAX Theory [76]
  • Fax Over Packet - a Telogy/TI whitepaper [77]
  • Audiocodes "Fax relay over packet networks" - fax technology primer [78]
  • Voice and Fax over Internet Protocol (V/FoIP) - a Telogy/TI whitepaper [79]
  • Cisco Fax Services over IP Application Guide [80]
  • Sagem-Interstar XMediusFAX 5.5 T.38 FoIP Solution [81]
  • Fax over IP - FoIP [82]
  • Cisco MGCP Based Fax (T.38) and DTMF Relay [83]


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Free Fax Internet services

Here is a FAQ on how to use the free FAX servers on internet and here are a few free Internet Fax Providers

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Telephony resources

  • Analog and Digital Voice Technology Primer (Motorola/Vanguard) [85]
  • 3Com PathBuilder S200 Series Switch Voice Relay guide [86] - see Chapter 2 "Theory of operation"
  • FXS/FXO signaling (channel bank device) [87]
  • T1 Basics (Stratum tech note) [88]
  • Voice/Data Integration Technologies (VoFR, VoATM, VoIP, MGCP) [89]
  • Voice Concepts for the Data Professional [90]
  • Adtran telephony tutorial [91]
  • All about telephony [92]
  • Telephony resources [93]
  • Circuit switching [94] local copy
  • Mitel Networks UK Training Basic Telephony Course [95]
  • QSIG - Handbook for Communications Managers [96]
  • Understanding Voice Signaling Protocols - Cisco primer [97] (local copy)
  • T1/E1/PRI Technology Overview [98]
  • All You Wanted to Know About T1 But Were Afraid to Ask [99]
  • Resolving or signaling unknown-caller phone calls numbers - at least until Bell Canada puts in place the Do Not Call List Register - Whocallsme.com and 800notes.com


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Sending SMS through E-mail

Sometimes one might want to be warned by SMS, for instance whenever a Voice message is deposed in an Asterix Voice Mailbox. This is easily done in Voicemail.conf, by sending an E-Mail to a SMS gateway. Here are a few known E-Mail to SMS free gateways:
  • Complete index of E-Mail to SMS gateways [100]. Examples:
    • Bell Mobility : <10-digit-number>@txt.bellmobility.ca or <10-digit-number>@bellmobility.ca
    • Telus : <10-digit-number>@msg.telus.com
    • Virgin Mobile : <10-digit-number>@vmobl.com or <10-digit-number>@vxtras.com
    • Vodaphone UK : <10-digit-number>@vodafone.net
    • Orange : <10-digit-number>@orange.net
    • Rogers : <10-digit-number>@pcs.rogers.com
    • Fido : <10-digit-number>@fido.ca
It is also possible to invoke the SendSMSWorld Web Service from within an AGI custom application. The ActiveXperts SMS and Pager Toolkit - allows adding SMS and Pager capabilities to an application.

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SS7 resources

Signaling System #7 (SS7) is a family of layered protocols being used for signaling between the components of the Intelligent Networks (IN) that make the backbone of nowadays' Public Switched Telephone Networks (PSTN). SS7 is a Common Channel Signaling protocol, where one separate signaling channel controls multiple bearer channels. SS7 builds a call's end-to-end circuit hop-by-hop, i.e. gets processed by each Service Switching Point (telephone exchange) in its way. The SS7 ISUP signaling is being used by the SSP's to derive billing information that is transmitted to the Service Control Points (SCP). The billing information that a carrier can derive from just the ISUP signaling is far from being sufficient, especially if taking into account one Telco's network resource-usage by traversing calls originating in another Telco's network. For this reason, more advanced billing solutions, based on sniffing the SS7 traffic on the network, have been developed. One example is Agilent's OSS acceSS7 Signaling Meter [101] [102].

Here are some SS7 references:

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Voice and video compression

  • Speex - patent-free speech compression format, based on [CELP] and compressing voice at bitrates ranging from 2 to 44 kbps
  • JSpeex - a Java port of the Speex speech codec
  • Theora - free video compression from the Xiph.Org Foundation
  • Ogg Vorbis - a patent-free, open, professional audio encoding and streaming technology
  • FLAC - Free Lossless Audio Codec, a patent-free audio compression technology similar to MP3, but lossless
  • Open Source g.729 codec


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HD VoIP

The term "high-definition" (HD) VoIP was coined by Iristel on February 1st, 2007, when it introduced the so-called high-definition (HD) VoIP service in Canada. The term indicates a VoIP technology based on wideband codecs, like G.722, and able to deliver CD-equivalent voice quality over packet. There are other, proprietary, wideband codecs, like Skype's ILBC or ISAC. The VoIP, which was predominantly been sold as a means of cost savings, with the advent of wideband technology, also lays the groundwork for revolutionary improvement in the quality of voice communications well beyond what the fixed 8Khz sampling-rate of PSTN would allow. Wideband VoIP only works well when voice communications are end-to-end VoIP without intervening PSTN segments. Here is a comparative list of codecs' quality performance, with SPEEX, an open-source, patent-free audio compression format, being one of the best wideband codecs. By comparison, Skype's GIPS codecs are proprietary and patent-protected, having been developed by Global IP Solutions (GIPS) Inc. According to unconfirmed information, the undocumented GIPS iLBC and iSAC algorithms are pretty much unrelated. iLBC (Internet Low Bitrate Codec) is a narrowband fixed rate codec operating at 13.3 kbps or 15.2 kbps, based on an IETF algorithm (RFC 3951 and 3952) and using RTP as transport. iSAC is an unrelated wideband variable rate codec, which can adapt its operating rate between 10 kbps and 32 kbps. Texas Instruments is licensing, for its VoIP chip-sets, the Low Delay AAC super wideband codec from the Fraunhofer Institute for Integrated Circuits IIS (headquartered in Erlangen, Germany). The super-wideband AAC codec technology features voice-sampling rates of up to 48 KHz, offering near-CD quality audio at data-rates of 48 - 64 kbps. Texas Instruments also integrated on its VoIP chip-sets the Microsoft the RT Audio codec, for improved compatibility with Microsoft unified communications platform. Microsoft's RT Audio Codec software is an advanced wideband speech codec designed for real-time two-way VoIP applications and featuring adaptive bandwitdh control and forward error correction.

References
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