WebRTC/RTP Server
From AVObjects Knowledge Base
(Difference between revisions)
| Line 41: | Line 41: | ||
* [http://tools.ietf.org/html/rfc3711 IETF RFC 3711(2004)]: The Secure Real-time Transport Protocol (SRTP) | * [http://tools.ietf.org/html/rfc3711 IETF RFC 3711(2004)]: The Secure Real-time Transport Protocol (SRTP) | ||
* [http://tools.ietf.org/html/rfc4566 IETF RFC 4566(2006)]: SDP: Session Description Protocol | * [http://tools.ietf.org/html/rfc4566 IETF RFC 4566(2006)]: SDP: Session Description Protocol | ||
| − | * [http://tools.ietf.org/html/rfc5389 IETF RFC 5389(2008)]: Session Traversal Utilities for NAT (STUN) | + | * [http://tools.ietf.org/html/rfc5389 IETF RFC 5389(2008)]: Session Traversal Utilities for NAT (STUN) |
| + | * [http://tools.ietf.org/html/rfc5389 IETF RFC 5766(2010)]: Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN) | ||
* [http://tools.ietf.org/html/rfc6184 IETF RFC 6184(2011)]: RTP Payload Format for H.264 Video | * [http://tools.ietf.org/html/rfc6184 IETF RFC 6184(2011)]: RTP Payload Format for H.264 Video | ||
* [http://tools.ietf.org/html/rfc6347 IETF RFC 6347(2012)]: Datagram Transport Layer Security Version 1.2 | * [http://tools.ietf.org/html/rfc6347 IETF RFC 6347(2012)]: Datagram Transport Layer Security Version 1.2 | ||
Revision as of 16:36, 26 October 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
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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 them by the network to peers wia WebRTC compatible protocols. The filter performs a renderer function and must be placed as a sink filter in the graph (the last filter in the graph chain).
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.
- Received streams can be displayed in browsers without Adobe Flash object.
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
- 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 5766(2010): Traversal Using Relays around NAT (TURN): Relay Extensions to 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
- H.264/AVC Encoder - H.264/AVC DirectShow video encoder.
- Video Generator
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. It can be useful when working with third-party audio codecs
