Category: WebRTC standards

Fault Tolerance and Error Correction in WebRTC

Fluctuating NetworksDynamic Bandwidth estimation JitterBuffer SDP renegotiationDemand for High Quality Video Tradeoff between Latency vs Quality Layering for adaptive streamingBetter compression algorithms vs CPU computeFull INTRA-frame Request (FIR)Picture Loss Indication (PLI)Redundant Encoding (RED) in Media Packets CongestionFeedback Loop Overcome congestion with lower bitrate Reduce frame quality and resolution Congestion control Algorithms : Google Congestion Control … Continue reading Fault Tolerance and Error Correction in WebRTC

SCTP (Stream Control Transmission Protocol)

SCTP is a reliable, message-oriented transport layer protocol. It was initially developed for telephony signaling (specifically for SS7 (Signaling System No. 7)) over IP, which requires robust and reliable messaging, but is widely used in WebRTC for data channels today as a standard. Being suitable for high availability , fault tolerant networks it is a … Continue reading SCTP (Stream Control Transmission Protocol)

Performance of WebRTC sites and electron apps

As security is a broad topic touching on many sections of WebRTC this section is not meant to address all topics but instead to focus on specific “hot spots”, areas that require special attention due to the unique properties of the WebRTC service. —There are several security related topics that are of particular interest with respect to WebRTC. They can be grouped into the following areas: Identity Management Browser Security Authentication Media encryption Syntax checks using regex

WebRTC APIs

WebRTC (Web Real-Time Communication) provides a set of JSEP APIs that enable real-time peer-to-peer (P2P) communication in web browsers. These handle media capture, encoding, network traversal (ICE), and secure data transmission. This article describes those in detail and also highlights usecases around varying states of these APIs. MediaDevices getUserMedia()getDisplayMedia()enumerateDevices()getSupportedConstraints()MediaStream and MediaStreamTrackPeer-to-peer connectionsRTCPeerConnection InterfaceCreateOffer() CreateAnswer()Offer/Answer Options - VoiceActivityDetectionRTCSessionDescriptionProfile … Continue reading WebRTC APIs

JavaScript Session Establishment Protocol (JSEP) in WebRTC handshake

SEP is used during signalling via w3c's recommended RTCPeerConnectionAPI interface to set up a multimedia session. The multimedia session description specifies the critical components of setting up a session between local and remote such as transport ports, protocol, profiles. It also handles the interaction with the ICE state machine.

WebRTC compatible android client

This post describes the requirement of creating a SIP phone application on android over the same codecs as WebRTC ( PCMA , PCMU , VP8) . In my project concerning the demonstration of WebRTC inter operability ( presence , audio / video call , message )  with a native android client , I had to … Continue reading WebRTC compatible android client

Difference between WebRTC and plugin based communication

A lot of service providers ie telecom operators had deduced their own ways to provide Web based communication even before WebRTC was born . With time , as WebRTC has become stronger , more secure , resilient to failure they have come around to migrate their existing system from previous closed box native APIs to … Continue reading Difference between WebRTC and plugin based communication

WebRTC

WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and P2P file sharing without the need for either internal or external plugins. Enables browser to browser media streaming over secure RTP profile Standardization, on an API level at the W3C and at the protocol level at the IETF. Enables web browsers with Real-Time Communications (RTC) capabilities written in c++ and javascript BSD style license free, open project available in all major browsers

WebRTC Stack Architecture and Layers

WebRTC offers web application developers the ability to write rich, realtime multimedia applications (think 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, across multiple platforms.

 WEBRTC CALL BETWEEN BROWSER AND SIP PHONE

HTML5 and WebRTC enabled Web Client : We are using open source HTML5 SIP client entirely written in javascript to make it light and to have easy integration with the SIP server. No extension, plugin or gateway is needed to initiate the call from the web Client. The media stack rely on WebRTC. The client … Continue reading  WEBRTC CALL BETWEEN BROWSER AND SIP PHONE