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WebRTC SIP Gateway

What is WebRTC

WebRTC is a new open framework for the web that enables Real Time Communications in the browser. Moreover it enables the voice calling, video chat, and P2P file sharing without plugins for browser-to-browser applications. WebRTC offers web application developers the ability to write realtime multimedia applications like video chat on the web, without requiring plugins, downloads or installs. It's purpose is to help build a strong RTC platform that works across multiple web browsers and multiple platforms. The major component of the WebRTC are: getUserMedia, PeerConnection, DataChannels. The first allows a web browser to access the camera and microphone. The second sets up audio/video calls. The last one allows browsers to share data via peer-to-peer.

What is SIP

SIP is the abbreviation for Session Initiation Protocol. SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions such as Internet telephony calls. SIP can also invite participants to already existing sessions, such as multicast conferences. Media can be added to and removed from an existing session.

SIP supports five facets of establishing and terminating multimedia communications:

User location: determination of the end system to be used for communication;
User availability: determination of the willingness of the called party to engage in communications;
User capabilities: determination of the media and media parameters to be used;
Session setup: "ringing", establishment of session parameters at both called and calling party;
Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.

What is a Gateway

A gateway interconnects networks with different network protocol technologies by preforming the required protocol conversion. It allows communication between the webphone using the RTMP and the SIP phone. A gateway is a network point that acts as an entrance to another network. In enterprises, the gateway is the computer that routes the traffic from a workstation to the outside network that is serving the Web pages. In homes, the gateway is the Internet Service Providers that connects the user to the internet. In enterprises, the gateway node often acts as a proxy server and a firewall. The gateway is also associated with both a router, which use headers and forwarding tables to determine where packets are sent, and a switch, which provides the actual path for the packet in and out of the gateway.

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WebRTC SIP Gateway

One of the greatest benefits of using WebRTC SIP Gateway is that every new web browser supports it. You do not have to install a plugin to use it, unlike with flash, java or silverlight. Ozeki Webphone Gateway supports WebRTC and thanks to this if you create a webphone by using Ozeki Webphone Gateway it will be able to communicate with Ozeki Phone System XE through WebRTC. Another benefit of using the WebRTC technology is that it is supported by the web browsers of the smartphones unlike the technologies which use flash.

Here is how it works:
The call gets to Ozeki Webphone Gateway through WebRTC, then the Ozeki Phone System XE plays the role of the gateway by converting the media data so that any kind of SIP phones connected to the Ozeki Webphone Gateway can answer the call. The webphone communicates with the Ozeki Webphone Gateway through the JavaScript API of the Ozeki Webphone Gateway.

Make peer-to-peer calls based on WebRTC with OZEKI Webphone Gateway learn more >>