WebRTC/RTP Server

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* W3C(CR) 2009-06-21:  WebRTC 1.0: Real-time Communication Between Browsers
 
* W3C(CR) 2009-06-21:  WebRTC 1.0: Real-time Communication Between Browsers
 
* [http://tools.ietf.org/html/rfc3550 IETF RFC 3550(2003)]: RTP: A Transport Protocol for Real-Time Applications
 
* [http://tools.ietf.org/html/rfc3550 IETF RFC 3550(2003)]: RTP: A Transport Protocol for Real-Time Applications
* IETF RFC 3551(2003): RTP Profile for Audio and Video Conferences with Minimal Control
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* [http://tools.ietf.org/html/rfc3551 IETF RFC 3551(2003)]: RTP Profile for Audio and Video Conferences with Minimal Control
* IETF RFC 3711(2004): The Secure Real-time Transport Protocol (SRTP)
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* [http://tools.ietf.org/html/rfc3711 IETF RFC 3711(2004)]: The Secure Real-time Transport Protocol (SRTP)
* IETF RFC 4566(2006): SDP: Session Description Protocol
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* [http://tools.ietf.org/html/rfc4566 IETF RFC 4566(2006)]: SDP: Session Description Protocol
* IETF RFC 5389(2008): Session Traversal Utilities for NAT (STUN)
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* [http://tools.ietf.org/html/rfc5389 IETF RFC 5389(2008)]: Session Traversal Utilities for NAT (STUN)
* IETF RFC 6184(2011): RTP Payload Format for H.264 Video  
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* [http://tools.ietf.org/html/rfc6184 IETF RFC 6184(2011)]: RTP Payload Format for H.264 Video  
* IETF RFC 6347(2012): Datagram Transport Layer Security Version 1.2
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* [http://tools.ietf.org/html/rfc6347 IETF RFC 6347(2012)]: Datagram Transport Layer Security Version 1.2
* IETF RFC 7231(2014): Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content
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* [http://tools.ietf.org/html/rfc7231 IETF RFC 7231(2014)]: Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content
* IETF RFC 7587(2015): RTP Payload Format for the Opus Speech and Audio Codec
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* [http://tools.ietf.org/html/rfc7587 IETF RFC 7587(2015)]: RTP Payload Format for the Opus Speech and Audio Codec
* IETF RFC 7741(2016): RTP Payload Format for VP8 Video
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* [http://tools.ietf.org/html/rfc7741 IETF RFC 7741(2016)]: RTP Payload Format for VP8 Video
  
 
===See Also===
 
===See Also===

Revision as of 14:02, 10 September 2018

This page is a copy of the original page on the AVObjects' web site and can also be viewed here.

DirectShow Filter for streaming media in the network

Overview

The WebRTC/RTP Server is a DirectShow Filter for sending media streams from the graph to the network. It gets encoded video or audio streams and send it by the network to peers wia WebRTC compatible protocols. The filter performs the function of render and should be the sink filter in a graph.

Features

  • Compatible with most popular browsers such as Google Chrome, Safari, Microsoft Edge
  • Compatibility with most mobile and desktop devices running on operating systems Windows, iOS, MacOS, Android.
  • Supports secure connections using SRTP and DTLS.
  • Supports video encoders H.264/AVC, VP8, VP9.
  • Supports audio encoders Opus, PCMA(G.711a), PCMU(G.711u).
  • Contains built-in HTTP signaling server for exchanging SDP information with WebRTC peers.
  • Display video in browsers without Adobe Flash

Supported Formats

  • H.264/AVC video
  • VP8 video
  • VP9 video
  • Opus audio
  • a-Law/PCMA/G.711a audio
  • u-Law/PCMU/G.711u audio

Supported Standards

See Also

You Might Also Need

Related Products

  • H.264/AVC Decoder - H.264/AVC DirectShow video decoder.
  • Audio Mixer - DirectShow filter for real-time mixing of multiple mono, stereo or multichannel audio streams.

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