VoIP

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Revision as of 05:19, 28 January 2018 by Milk (talk | contribs) (→‎Systems)
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General

See also Sharing#WebRTC, Media#Telephone, Chat




  • Free Telco Dictionary – This dictionary should be helpful for employees in telecommunications and also for independent hackers interested in this industry


Protocols

RTP

ZRTP

SRTP

SIP

See also Sharing#SIP

Ring

PJSIP

  • PJSIP - a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, embedded systems, to mobile handsets. PJSIP is both compact and feature rich. It supports audio, video, presence, and instant messaging, and has extensive documentation. PJSIP is very portable. On mobile devices, it abstracts system dependent features and in many cases is able to utilize the native multimedia capabilities of the device. PJSIP has been developed by a small team working exclusively for the project since 2005, with participation of hundreds of developers from around the world, and is routinely tested at SIP Interoperability Event (SIPit ) since 2007.

XMPP Jingle

See also Chat#Jingle

WebRTC

See Sharing#WebRTC

Services



  • OpenCNAM provides a simple, elegant, and RESTful API to get Caller ID data. Our service is built for programmers like you, who want simple access to Caller ID information.
  • slydial is a free voice messaging service that connects you directly to someone's mobile voicemail. slydial is a service of MobileSphere.


Systems

Asterisk

  • Asterisk is an open source framework for building communications applications. Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. It is used by small businesses, large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free and open source. Asterisk is sponsored by Digium.

Today, there are more than one million Asterisk-based communications systems in use, in more than 170 countries. Asterisk is used by almost the entire Fortune 1000 list of customers. Most often deployed by system integrators and developers, Asterisk can become the basis for a complete business phone system, or used to enhance or extend an existing system, or to bridge a gap between systems.

FreeSWITCH

GNU SIP Witch

  • GNU SIP Witch is a secure peer-to-peer VoIP server that uses the SIP protocol. Calls can be made peer-to-peer behind NAT firewalls, and without needing a service provider. GNU SIP Witch does not perform codec operations and thereby enables SIP endpoints to directly peer negotiate call setting and process peer to peer media streaming even when when multiple SIP Witch call nodes at multiple locations are involved. This means GNU SIP Witch operates without introducing additional media latency or offering a central point for media intercept or capture. GNU SIP Witch can be used to build secure and intercept-free telephone systems that can operate over the public Internet.

OpenVBX

TropoVBX

GNU Bayonne

  • GNU Bayonne traditionally operates as a generic script driven telephony server that can operate with existing third party telephony api kits, such as those produced by Intel-Dialogic, Aculab, Pika (montecarlo), or Voicetronix. GNU Bayonne also includes pure network drivers for SIP and H323. Specific drivers and api adaptions offer features unique and targetted to those environments. For example, some telephony api's support various forms of conferencing, and these features are available in Bayonne adaptions for those api's.

Common uses include voice messaging, voice broadcast, and prepaid calling in conjunction with a SIP server or H323 gatekeeper such as Ser, SipX, GNU SIP Witch or GNU Gate Keeper; for offering IMS services and for carrier hosted applications; and for integration with legacy digital and analog key telephone systems. These use cases imply that Bayonne will also assume the goals and functionality of the original Babylon PBX integration server as well.

Doubango

  • Doubango Telecom is a young Telco company focused on open source projects. We are specialized in NGN technologies (3GPP, TISPAN, Packet Cabel, WiMax, GSMA, RCS-e, IETF...standards), audio/video coding, cloud computing and WebRTC. Our products include SIP/IMS (VoIP) clients/servers/gateways, TelePresence and Telemedicine systems, VNC stacks and audio/video codecs. Most of our products are already open sourced.
    • https://github.com/DoubangoTelecom/doubango - a mature, open source, 3GPP IMS/LTE framework for both embedded and desktop systems. The framework is written in ANSI-C to ease portability and has been carefully designed to efficiently work on embedded systems with limited memory and low computing power and to be extremely portable.


  • https://github.com/2600hz/kazoo Kazoo, an ambitious project to bring cloud-based VoIP and telecommunications to everyone. Our goal is to provide the world with a free, open telecommunications software platform. Released under the OSI-approved MPL 1.1 open source software license, we're building upon strong FOSS components like GNU/Linux, Erlang, FreeSWITCH, Apache CouchDB, and RabbitMQ. Our project is a great example of the wonderful things that can happen when software is open. Kazoo is an API-based platform that lets you use your existing phones, programming languages and IT skills to build voice, video and SMS services. We focus on building a simple, powerful communications platform and let you focus on marketing, servicing and integrating communications with your clients systems.


  • http://www.pjsip.org/ - PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, embedded systems, to mobile handsets.



Other

  • sscall - A simple UDP based voice chat program. Currently we use libspeexdsp for its resampling capabilities and opus as the audio codec. There will also be ssl support in future versions. We basically need something that works well on many UNIX flavours. Skype is not really the answer to that. We also want something simple so that we can build on top of it. The plan is to create another program called ssvideo for video streaming.

Ekiga


  • Seren - Seren is a simple VoIP program based on the Opus codec that allows you to create a voice conference from the terminal, with up to 10 participants, without having to register accounts, exchange emails, or add people to contact lists. All you need to join an existing conference is the host name or IP address of one of the participants.

Seren creates a dynamic peer-to-peer network of equivalent nodes which exchange text and audio data using a udp connection, and offers the user the ability to change the quality/bitrate on the fly, encrypt the traffic and record the calls.

Linux clients

Skype

Run multiple Skype sessions [3]; 

skype --dbpath=~/.Skype2 &
  • Karaka is a Skype/XMPP gateway that connects the Skype and XMPP clouds.

Mumble

Ekiga

Jitsi

SLPhone

Linphone

Twinkle

I Hear U

  • IHU is a Voice over IP (VoIP) application for Linux (using Qt), that creates an audio stream between two computers easily and with the minimal traffic on the network.

Android clients

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