WebRTC/RTP Server

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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 and 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), PCMa(G.711u).
  • Contains built-in NTTP signaling server for exchanging SDP information with WebRTC clients.

Supported Formats

Supported MXF Standards

  • W3C(CR) 2009-06-21: WebRTC 1.0: Real-time Communication Between Browsers
  • 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
  • IETF RFC 3711(2004): The Secure Real-time Transport Protocol (SRTP)
  • IETF RFC 4566(2006): SDP: Session Description Protocol
  • IETF RFC 5389(2008): Session Traversal Utilities for NAT (STUN)
  • IETF RFC 6184(2011): RTP Payload Format for H.264 Video
  • IETF RFC 6347(2012): Datagram Transport Layer Security Version 1.2
  • 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
  • IETF RFC 7741(2016): RTP Payload Format for VP8 Video

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.
  • Audio Level - Provides support for level processing of an audio stream.

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