VoIP

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General

See also Sharing#WebRTC, Media#Telephone, Chat

  • https://en.wikipedia.org/wiki/Telepresence - refers to a set of technologies which allow a person to feel as if they were present, to give the appearance of being present, or to have an effect, via telerobotics, at a place other than their true location. Telepresence requires that the users' senses be provided with such stimuli as to give the feeling of being in that other location. Additionally, users may be given the ability to affect the remote location. In this case, the user's position, movements, actions, voice, etc. may be sensed, transmitted and duplicated in the remote location to bring about this effect. Therefore information may be traveling in both directions between the user and the remote location.

A popular application is found in telepresence videoconferencing, the highest possible level of videotelephony. Telepresence via video deploys greater technical sophistication and improved fidelity of both sight and sound than in traditional videoconferencing. Technical advancements in mobile collaboration have also extended the capabilities of videoconferencing beyond the boardroom for use with hand-held mobile devices, enabling collaboration independent of location.



  • http://en.wikipedia.org/wiki/Voice_over_Internet_Protocol - a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).

The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, the digital information is packetized, and transmission occurs as IP packets over a packet-switched network.


  • VOIP Wiki - a reference guide to all things VOIP, covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.




  • RichardNeill.org: A quick, personal guide to VOIP under Linux - this guide was written in 2007. It's mostly still correct, but many of the applications have substantially improved. Also, FreeWorldDialup is no more; try iptel.org instead. The best approach is now Jitsi, notably Video Bridge and meet.jit.si.


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


  • https://en.wikipedia.org/wiki/Federated_VoIP - a form of packetized voice telephony that uses voice over IP between autonomous domains in the public Internet without the deployment of central virtual exchange points or switching centers for traffic routing. Federated VoIP uses decentralized addressing systems, such as ENUM, for location and identity information of participants and implements secure, trusted communications (TLS) for identify verification.
  • https://en.wikipedia.org/wiki/Internet_telephony_service_provider - offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet. ITSPs provide services to end-users directly or as whole-sale suppliers to other ITSPs. ITSPs use a variety of signaling and multimedia protocols, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), Megaco, and the H.323 protocol. H.323 is one of the earliest VoIP protocols, but its use is declining and it is rarely used for consumer products. Retail customers of an ITSP may use traditional analog telephone sets attached to an analog telephony adapter (ATA) to connect to the service provider's network via a local area network, they may use an IP phone, or they may connect a private branch exchange (PBX) system to the service via media gateways. ITSPs are also known as voice service providers (VSP).
  • https://en.wikipedia.org/wiki/Telephone_number_mapping - a system of unifying the international telephone number system of the public switched telephone network with the Internet addressing and identification name spaces. Internationally, telephone numbers are systematically organized by the E.164 standard, while the Internet uses the Domain Name System (DNS) for linking domain names to IP addresses and other resource information. Telephone number mapping systems provide facilities to determine applicable Internet communications servers responsible for servicing a given telephone number using DNS queries. The most prominent facility for telephone number mapping is the E.164 Number to URI Mapping (ENUM) standard. It uses special DNS record types to translate a telephone number into a Uniform Resource Identifier (URI) or IP address that can be used in Internet communications.


  • https://en.wikipedia.org/wiki/VoIP_phone - or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet, instead of the traditional public switched telephone network (PSTN). Digital IP-based telephone service uses control protocols such as the Session Initiation Protocol (SIP), Skinny Client Control Protocol (SCCP) or various other proprietary protocols.
  • https://en.wikipedia.org/wiki/Softphone - a software program for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other computer and allows the user to place and receive calls without requiring an actual telephone set. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a handset, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or with a USB phone.



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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