Telecom API Platform Providers

COMPANY

DESCRIPTION

Alcatel Lucent

Alcatel-Lucent WebRTC helps service providers deliver communications as a web experience. Building on the WebRTC standard, this solution enables you to extend your brand to the
web and create a new conversation experience that unifies the telecom and web domains. With Alcatel-Lucent WebRTC, you get the tools you need to create new applications for
consumers and empower enterprise customers with IT-centric communications. The Alcaltel-Lucent WebRTC solution makes it easy to embed real-time communications into
applications, websites and browsers. It opens up new opportunities to capitalize on existing network investments and enhance the customer experience. By using your network to
extend cohesive value across telecom and the web, you can increase your total addressable market, create new markets and inspire web developers innovation.

AudioCodecs
Doubango Telecom

AudioCodecs provides an SBC as a bridge between WebRTC and existing telephony networks.
Doubango Telecom offers webrtc2sip which is their gateway between RTCWeb and SIP to turn your browser into a phone with audio, video and SMS capabilities. The gateway allows
your web browser to make and receive calls from/to any SIP-legacy network or PSTN. The gateway contains four modules: SIP Proxy | RTCWeb Breaker | Media Coder | Click-to-Call.

Ericsson

The Ericsson Web Communication Gateway (WCG) translates between telecom and Internet protocols. It exposes communication APIs in an Internet-friendly way, making it easy to
add communication services to Internet-based applications. The WCG uses well known web technologies like HTTP and WebRTC to significantly reduce time-to-market for new
combinational services that can be used on practically any device.
Flashphoner Web Call Server 3.0 is a communication platform which enables audio video calls and instant messaging between web browsers and SIP, VoIP PBX, GSM gateways, and
other SIP devices. WCS is based on HTML5 Websockets, WebRTC, and Flash RTMFP technology the most advanced technology for web calls.
FreeSWITCH is free and open source communications software for the creation of voice and messaging products. It is licensed under the Mozilla Public License (MPL), a free software
license. Its core library, libfreeswitch, is capable of being embedded into other projects, as well as being used as a stand-alone application.
GENBAND's SPiDR WebRTC Gateway provides an intelligent bridge between traditional Voice over Internet Protocol (VoIP) networks and the open ecosystem of the Internet. SPiDR
empowers network operators to deliver competitive applications over the Internet and unlocks new revenue potential from their existing wireless and fixed communication assets.
SPiDR sits at the edge of the network and provides open, web-centric APIs that allow application developers to leverage the rich communications services of the telecommunications
network including voice, video, presence, shared address book, call history, instant messaging, and collaboration. SPiDR seamlessly and intelligently interworks both the signaling and
media planes between the web and telecom worlds.
Intel offers a WebRTC conference server, gateway, and client SDKs.
Mavenir's WebRTC Gateway enables service delivery of IMS core network services to WebRTC clients. Mavenirs WebRTC Gateway acts as a bridge between Web and mobile operator
networks allowing mobile operators to extend their communication services to the Internet. The Gateway provides interworking between IMS services and WebRTC clients. It can also
enable real time communication interworking between various Web based communities and the mobile operator network.
Janus is a WebRTC Gateway developed by Meetecho conceived to be a general purpose one. As such, it doesn't provide any functionality per se other than implementing the means to
set up a WebRTC media communication with a browser, exchanging JSON messages with it, and relaying RTP/RTCP and messages between browsers and the server-side application
logic they're attached to. Any specific feature / application is provided by server side plugins, that browsers can then contact via the gateway to take advantage of the functionality
they provide. Example of such plugins can be implementations of applications like echo tests, conference bridges, media recorders, SIP gateways and the like.

Flashphoner
FreeSwitch
Genband

Intel
Mavineer

Meetecho

Mobicents
Nex Gen Bits, LLC

Oracle
Sansay

Solaiemes
Telestax

WIT Software

Carl Drueckhammer

Mobicents is an Open Source VoIP Platform written in Java to help create, deploy, manage services and applications integrating voice, video and data across a range of IP and legacy
communications networks. It drives convergence with the following open source projects:
NGVX or Nex Gen Video Exchange is a video calling server bringing together the world of SIP and WebRTC. NGVX is deployed like an MCU allowing seamless interoperability between
SIP/IMS mobile clients, SIP video phones, and WebRTC. Enabling you to easily and securely host a meeting, record the video and audio content, while simultaneously webcasting the
content to any number of native devices such as iPhones, Android, Blackberry, and desktop clients.
The Oracle Communications WebRTC Session Controller enables communications service providers (CSPs) and enterprises to offer WebRTC services from virtually any device, across
virtually any network with carrier-grade reliability and security.
Sansay, Inc. was founded in June 2002 with the objective of developing the highest quality and most useful VOIP infrastructure systems for carriers and service providers worldwide.
The Sansay WebSBC platform is positioned to provide the WebRTC gateway functionality needed to bridge these WebRTC endpoints into the existing VoIP service network and
applications while providing a growth path for its customers to evolve to the new application enablement capabilities promised by the emerging WebRTC infrastructure.
Based in the Solaiemes WebRTC-Telco GW SDK, it becomes the "new SDP". Build a lot of monetizable services combining telco and other types of identies as the ones used in Social
Networks. Finally telcos can get revenue from the internet space ;-)
TeleStax provides open source communications software and services that facilitate the shift from legacy SS7 based IN networks to IP based LTE and IMS networks hosted on private
(on-premise), hybrid or public clouds. TeleStax powers cloud communications for the worlds leading service providers in countries across 5 continents.
The WIT WebRTC gateway enables real-time communication between any browsers in the market without requiring any additional plugin installation. The Gateway leverages the best
use of WebRTC technology and provides the necessary conversion into Telecom protocols. It also enables real-time communications from the web browsers into smartphones, tablets,
PC softphones, SIP phones and GSM/PSTN phones.

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2/19/2015