OpenSourcePBX / Softswitch


Android

【参照ページ】

OpenSourceTelephonyList

【オープン・ソース系PBXプラットフォーム概況】

PBX Plarform(Asterisk、&verb(CallWeaver)、OpenPBX、sipX、SIPxchange) &aname(f88e2951,super,full){†};

名称ベース種別コミュニティー派生企業概 況
AsteriskPBX系IP-PBXAsterisk.orgFonality社、Intuitive Voice Technology社、Four Loop Technology社、&verb(AstLinux?)Digium社のMark Spencer氏が主開発者。専業9名90名のボランティア、5年の歳月を掛けて開発された、高度なIP-PBXセット。GPLライセンスと独自ライセンスの2種類を持っている。現在、Asteriskコミュニティーでは、PBX間通信(enable VoIP connections between servers as well as client-server communication)を円滑にするIAX(Inter-Asterisk eXchange)の開発が進んでいる。
&verb(CallWeaver?)PBX系IP-PBX&verb(CallWeaver?)---&verb(CallWeaver?)(旧名称 OpenPBX.Org)は、コミュニティー主導、ベンダー中立のオープン・ソースPBXプロジェクト。Asteriskから派生している。現在、アナログおよびデジタルPSTNをサポートし、VoIPやファックス(含ソフトファックス)、音声応答システム、電話会議、コールセンター・マネージメントなどの機能を実装している。<特徴> Cross-platform (Linux、 FreeBSD、 NetBSD、 OpenBSD、 MacOS X/Darwin、 Open/Solaris) 、PSTN connectivities (FXS/FXO、 ISDN、 PRI、 E1、 T1) 、Multi-protocol voice over IP (H.323、 IAX2、 MGCP and SIP and SCCP) 、Fax 、T.38 Fax over IP (pass-through、 termination and gateway) 、IVR 、Conferencing 、Queues 、Reliability
OpenPBXPBX系PBXOpenPBX.orgVoicetronixOpenPBXはLinuxベースのPBXソフト。パールで書かれており、軽量。基本的には、コンピュータ・テレフォニー・サーバーおよび関連機器を取り扱うVoicetronix社が、自社製品向けに開発したOS-PBXと言える。同社は、OpenPBX.orgと言うコミュニティーをサポートしているほか、Bayonne、Asteriskなどのオープン・コミュニティーにも参加している。
sipX by SIPfoundrySIP系SIP-PBXSIPfoundryPingtel社SIPfoundryはオープンソース系非営利団体。SIP ForumやIETFと協力してSIPの標準化、相互運用などを進める。sipX(=sipXpbx)プロジェクトはオープンソース系SIP-PBXの開発プロジェクト。フルファンクションのPBX開発をめざす。Pingtel社はsipXプロジェクトのメンバーと「sipXecs」と言うsipXのアップグレード・プロジェクトを進めている。
SIPxchange by PingtelSIP系SIP-PBX---Pingtel社Pingtel社がsipXをベースに開発した企業グレードのSIP-PBX製品。Linuxベースで各種IP電話端末やゲートウェーに対応。

【OS-PBX系統図】

http://192.168.0.7/wiki/photo/OS-PBX1.jpg

【sipXとSIPxchangeの比較】

sipXSIPxchange
Audio QualitysipX includes its own media framework that is fully available in open source but of lesser quality as compared to GIPS.SIPxchange relies on the GIPS media framework、 a commercial third-party solution that improves audio quality due to better packet loss concealment and jitter buffer management. It is licenses by GIPS on a per copy basis.
User Interface TemplatesipX Config Server comes with the SIPfoundry colored user interface template.SIPxchange uses a different template.
DocumentationOnly community provided documentation is available for sipX.SIPxchange is shipped with a comprehensive installation and administrators manual. In addition the package includes application notes on how to use and configure end points and other peripheral devices.
SupportOnly community based support is available for sipX.Pingtel and its resellers offer different levels of commercial support for SIPxchange. Pingtel guarantees long term support for its releases to support production installations where customers are unable to upgrade every couple of months. In contrast users of the community release will have to follow the upgrade path of the community release.
TrainingNo formal training is available.Pingtel and its resellers offer formal online training for administrators of SIPxchange.
Q&A and end point certificationTesting based on community peer review.SIPxchange is thoroughly regression and stress tested before release. Specific end points such as phones and gateways are certification tested and guaranteed to work.
Online HelpsipX links to this Wiki for online help.SIPxchange provides access to an online administrators manual for help.

Source:&link(SIPfoundry,http://sipx-wiki.calivia.com/index.php/SIPxchange_vs._sipX_Comparison)


Telephony platform(FreeSWITCH、GNU Bayonne、YATE)

名称ベース種別コミュニティー派生企業概 況
FreeSWITCHTelephony platformsoftphone〜softswitchfreeswitch.org---FreeSWITCHはソフトフォンからソフトスイッチまで多彩なアプリケーションを組めるテレフォニー・プラットフォーム。シンプルなスイッチング・エンジンやメディア・ゲートウェー、メディア・サーバとして音声応答システムなどのホスティングに使える。Linux、BSD系、Solaris、MacOS X、Windowsなどマルチプラットフォーム。SIP、IAX2、&verb(GoogleTalk?)などにも対応
GNU BayonneIVR platformIVRGNU TelephonyVoicetronix、Dialogic、Aculab、Synwayのハードウェアが対応GNU BayonneはGNU Telephonyプロジェクトの一環。音声応答システム(IVR)プラットフォームで、Linux、BSD類、MacOS X、Windowsなどをサポート。
YATETelephony platformVoIP server〜IVRNULL teamYate(Yet Another Telephony Engine)は、VoIPおよびPSTNをサポートするTelephony PlatformでSIP、IAX、H.323に対応。

<注意>

  • OpenPBXは、以下3つ動きがある。
    • &verb(VoiceTronix?)社のOpenPBX(パールベースのOS-PBX)
    • OpenPBX VXMは、ウィンドウズ・ベースのボイスメール・ソフトウェア。伝統的なPBXといっしょに利用する。
    • &verb(CallWeaver?)は昔、OpenPBX.Orgと称していた。&verb(CallWeaver?)はAsteriskをベースにしたOS-PBX開発プロジェクト。

【Open Source PBX/Softswitch】

Digium社 &link(Asterisk,http://en.wikipedia.org/wiki/Asterisk_%28PBX%29) &aname(l0b24576,super,full){†};

Fonality社

  • 製品名 &link(trixbox,http://www.trixbox.com/)
    • アスタリスクをコアに付加機能を付け加えたクローン・エンジン
    • trixbox、 spelled with a lowercase 't'、 is a line of Asterisk-based IP-PBX products designed to meet the needs of companies from 2 to 500 employees. trixbox comes in two flavors: the open-source community edition and a hybrid--hosted、 commercially-proven solution.(出典:同社ホームページ)
      tribox Proの概要
      http://192.168.0.7/wiki/photo/trixbox1.jpg
  • 製品名 &link(PBXtra,http://www.fonality.com/pbxtra_features.html)
    • The birth of PBXtra? was quite by accident. Fonality was founded in early 2003 and launched a residential VoIP product later that year. As our company grew、 and our phones started ringing、 we found we needed to "graduate" to a multi-line phone system. So、 we went shopping. We were shocked at the price of just a basic PBX system. For this reason、 we developed our own in-house IP-PBX system. As time went by、 we added an increasing number of features to our in-house prototype. One day we looked at what we had built and realized we had something special. Thus PBXtra was born.
    • PBXtra has matured since then by undergoing a major overhaul to introduce scalability、 security and ease-of-use into every corner of the product. Today's PBXtra is a robust and economical product、 largely because it is built upon the OS backbone of Linux and the shoulders of the open-source IP-PBX platform named Asterisk.(出典:同社ホームページ)

Intuitive Voice Technology社 &link((IVT),http://www.intuitivevoice.com/) &aname(u773dec8,super,full){†};

  • アスタリスク・ベース・PBX
  • Evolution PBX(製品名)
    • Introducing the next generation of voice communications. The Evolution PBX offers state of the art IP Telephony features in a highly intuitive low maintenance PBX. With the Evolution PBX small businesses、 entrepreneurs、 and sales professionals can have a communications system that functions similar to the million-dollar PBX systems FORTUNE 500 companies use. With features that don't require a computer degree to use!
    • Designed to enable a small business without an IT staff to benefit from the recent wave of telephony enhancements、 Evolution packs features commonly available in high end systems in a simplified easy to use package. The days of calling your telecommunications supplier to enable another extension on your PBX are gone. With Evolution you simply plug your phone into a standard network jack and let the system do the rest. All the configuration is done through an easy to use web interface. IVT has partnered with most of the major VoIP Providers and integrated their service offerings directly into our product. In a small company、 good communication with customers and suppliers is critical. Let us make it your reality.(出典:同社ホームページ)

Four Loop Technologies社 &link(Switchvox,http://www.switchvox.com/) &aname(eca34216,super,full){†};

  • 2007年9月27日、Digium社がSwitchvoxを買収。目的は中小企業マーケットの営業強化。
  • Switchvox、 with an estimated 65、000 end points in operation、 is the world's largest and most successful supplier of open source-based IP PBX products for businesses. The combination of Digium and Switchvox will provide open source-based products and solutions that are unrivaled in the industry.(出典:同社ホームページ)

&link(AstLinux,http://www.astlinux.org/node/1) &aname(d2337398,super,full){†};

  • &verb(AstLinux?) is a custom Linux distribution centered around Asterisk, the Open Source PBX.&verb(AstLinux?) is available for multiple processor architectures. &verb(AstLinux?) has specific images for the SC1100 series of SBC's (single board computers), including those from Soekris Engineering and PC Engines. There are also images targeted for the exciting Gumstix SBC and the VIA line of mini-itx boards. The "generic" image runs on pretty much any standard PC.(出典:同社ホームページ)

SIP Express Router &link((SER),http://www.iptel.org/ser/) &aname(cf579b87,super,full){†};

  • IPTEL.ORGのオープンソースSIPサーバー
  • OpenSER &link(ホームページ,http://www.openser.org/)
  • SIP Express Router (SER) is a high-performance、 configurable、 free SIP server licensed under the open-source GNU license . It can act as SIP (RFC 3261) registrar、 proxy or redirect server. SER can be configured to serve specialized purposes such as load balancing or SIP front-end to application servers、 SEMS for example.
  • SER features complete support of RFC 3261 functionality、 a variety of database backends (mysql、 oracle、 postgres、 radius、 text-db)、 management features (remote management via XML-RPC、 load-balancing)、 NATi traversal、 telephony features (LCR、 speeddial)、 multidomain hosting、 ENUM、 presence、 and even more.
  • SER is additionally enhanced by a variety of additional SIP tools、 which provide functionality for management、 media processing、 CDRi processing、 etc.(出典:IPTEL.ORG)

&link(FreeSWITCH,http://wiki.freeswitch.org/wiki/Main_Page) &aname(a3d18ce6,super,full){†};

  • &verb(FreeSWITCH) is an open source communications platform. FreeSWITCH is a library which ships with a small executable that loads the library、 launches the core、 and performs the various tasks that are defined by the modules. In its base form FreeSWITCH is a soft-switch or PBX telephony application、 not completely unlike Asterisk but capable of handling thousands of simultaneous calls. FreeSWITCH makes it possible to build a softphone、 PBX system、 soft-switch、 or interface with other open source PBX systems such as &verb(CallWeaver?) (formerly known as OpenPBX.org)、 Bayonne、 YATE or Asterisk. It can also be used to build a voip switching platform uniting various technologies such as SIP、 H.323、 IAX2、 LDAP、 Zeroconf、 XMPP / Jingle etc.(出典:FreeSWITCH.org)

&link(CallWeaver,http://www.callweaver.org/blog) &aname(m18c356f,super,full){†};

  • &verb(CallWeaver?) is a community-driven、 vendor-independent、 cross-platform、 open source、 PBX software project (formerly known as OpenPBX.org). It was originally derived from Asterisk. Now it supports analog and digital PSTN telephony、 multi-protocol voice over IP telephony、 fax、 software-fax、 T.38 fax over IP and many telephony applications such as IVR、 conferencing and callcenter queue management.

OpenPBX

  • an open source software-PBX written in Perl, by Australian vendor &link(Voicetronix,http://www.voicetronix.com/open-source.htm)
  • A full function、 web-enabled small office PBX package. Turn your PC into a PBX using a Voicetronix card and the OpenPBX software package. Integrate PBX and other back-office functionality such as email、 web and file serving into a single PC to build a fully functional PBX/IT solution for SOHO environments.
  • The web based GUI means the PBX can be easily configured、 compared to other PBXes that have clunky、 hard to use touch-tone based user interfaces. For example your web browser can be used to access your voice mail、 rather than pressing buttons on your phone.
  • &link(olineDemo,http://www.voicetronix.com/openpbx)
  • Technical
    • The OpenPBX is an Open Source Linux PBX written entirely in Perl. It interfaces to the Voicetronix telephony cards via the CT Server middle-ware described below. The code is very compact、 only 1000 lines of Perl code are required for the basic PBX functionality. This made development extremely rapid、 compared to using traditional approaches like C/C++. It also means the PBX is very easy to extend and customize、 for example add unlimited voice mail boxes、 or a custom IVR menu ("Press 1 for Sales、 2 for Support....") with a small amount of Perl code.
  • Download
    • OpenPBX is part of the CT Server package described below. Just download the CT Server package to get started with OpenPBX!

&link(Pingtel,http://www.pingtel.com/) &aname(aa392b66,super,full){†};

  • Pingtel、 now a Bluesocket Company、 is reshaping the communications market by delivering the only 100% SIP-based enterprise class communications platforms that support the features and functionality required by business markets. Pingtel combines the best attributes of open source development?low cost、 adaptability and flexibility?with the reliable solutions and support enterprises require for voice applications. For more information visit http://www.pingtel.comhttp://www.sipfoundry.org、 and http://www.bluesocket.com.

【Open Telephony Frame/Platform】

&link(Bayonne,http://wiki.gnutelephony.org/index.php/GNU_Bayonne) &aname(o91aa247,super,full){†};

  • Bayonne is the telephony server of the GNU Telephony project. It offers a free open source、 scalable、 media independent software environment for development and deployment of computer telephony solutions for use with current and next generation telephone networks.
  • サポートページ:&link(voip-info.org,http://www.voip-info.org/wiki/view/Bayonne)
  • Reviews (Mattf writes)
    • the strengths of Bayonne: 1) Runs on Dialogic、Pika and other widely available hardware 2) extremely reliable、mine never crashes
    • here are the weaknesses: 1)nowhere near as active of a support community as Asterisk has configuration of the hardware/drivers is a nightmare compared to Asterisk/Digium 2)it is quite limited in it's included apps、 IVR and voicemail not as many options for scripting as Asterisk 3)it was not designed to have full PBX functionality、 4)some PBX functionality is added as afterthought the code/organization/flow is not as well thought out or documented as Asterisk is.
    • (Note: This review refers to Bayonne 1. Some of the criticisms do not apply to Bayonne 2)

&link(YATE,http://yate.null.ro/pmwiki/) &aname(k3951ca7,super,full){†};

  • YATE is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN、 its power lies in its ability to be easily extended. Voice、 video、 data and instant messaging can all be unified under Yate's flexible routing engine、 maximizing communications efficiency and minimizing infrastructure costs for businesses.
  • Yate can be used as a:
    VoIP serverVoIP clientVoIP to PSTN gateway
    PC2Phone and Phone2PC gatewayH.323 gatekeeperH.323 multiple endpoint server
    H.323<->SIP ProxySIP session border controllerSIP router
    SIP registration serverJingle serverISDN passive and active recorder
    IAX server and/or clientIP Telephony server and/or clientCall center server
    IVR enginePrepaid and/or postpaid cards system
  • The software is written in C++ and it supports scripting in various programming languages (such as those supported by the currently implemented embedded PHP、 Python and Perl interpreters) and even any Unix shell. The PHP、 Python and Perl libraries have been developed and made available in order to ease development of external functionalities for Yate.
  • Yate is production-ready software and is easily extensible.
  • Yate is licensed under the GPL with an exception for linking with OpenH323 and PWlib (licensed under MPL).

&link(SIPfoundry,http://en.wikipedia.org/wiki/SIPfoundry) &aname(mb8b062d,super,full){†};

  • SIPfoundry is a not-for-profit open source community, whose mission is to promote and advance Session Initiation Protocol (SIP) - related open source projects. Through SIPfoundry, the users, developers, and distributors of SIP-based products can collectively support each other and accelerate the growth and adoption of SIP.
  • Founded in March 2004, SIPfoundry established close ties with the SIP Forum as well as the IETF. SIPfoundry actively promotes the standardization of SIP and interoperability of SIP products and solution across the industry through the SIP Forum Test Framework (SFTF).
  • SIPfoundry is also the place where the development of sipX, an open source SIP PBX for Linux, takes place. This project aims at commoditizing PBXs by offering a fully featured, standards-compliant, and easy-to-use SIP IP PBX for free as an open source solution. SIPfoundry would like SIP to become part of the Internet the same way HTTP, SMTP, and XML became ubiquitous and drove rapid adoption of new services across the Internet.(出典:英文ウィキペディア)

【業界/規格団体/関連雑誌】

Linux Phone Standards Forum &link(LiPS,http://www.lipsforum.org/) &aname(z55fbe91,super,full){†};

  • ウィキの&link(解説,http://en.wikipedia.org/wiki/Linux_Phone_Standards_Forum)
  • The Linux Phone Standards Forum (LiPS) is a consortium founded by a group of telephony operators, device manufacturers, silicon and software vendors who have a strategic focus on Linux® telephony.
  • 小池メモ:Open Source PBXとは直接関係ないが、Linux携帯端末は重要な課題。今後、この方向にiPBXは伸びてゆくのでは?

関連情報サイト

【資料メモ】


Trixbox (Fonalty社)

  • trixbox, formerly Asterisk@Home, which started as a home-grown Asterisk project, has rapidly become one of most active open source projects in the world. In 2006, trixbox attracted the attention of Fonality, a Los Angeles-based IP-PBX company. The synergies between the companies quickly became clear and Fonality acquired trixbox in October 2006. Fonality now offers PBXtra?, a turnkey enterprise-class phone systems for small and medium-sized businesses; trixbox Pro, a hybrid-hosted software-based solution for resellers; and trixbox CE, an open-source flexible solution for the "build it yourself" enthusiast.
  • HUD is the award-winning employee presence and communication management application that comes free with all versions of trixbox. HUD empowers its users with company-wide visibility and information on the "presence" of every colleague, making it easy to interact with one another via a single, simple interface.(出典:trixboxホームページ)

[作成日:2007/09/14]