WebRTC/RTP Server
From AVObjects Knowledge Base
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==Overview== | ==Overview== | ||
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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). | 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). | ||
| + | * [http://avobjects.com:8050 Live stream from our server] | ||
==Features== | ==Features== | ||
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* Received streams can be displayed in browsers without Adobe Flash object. | * Received streams can be displayed in browsers without Adobe Flash object. | ||
| − | + | ==Supported Formats== | |
* H.264/AVC video | * H.264/AVC video | ||
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* u-Law/PCMU/G.711u audio | * u-Law/PCMU/G.711u audio | ||
| − | + | ==Supported Standards== | |
* [https://www.w3.org/TR/webrtc W3C(CR) 2009-06-21]: WebRTC 1.0: Real-time Communication Between Browsers | * [https://www.w3.org/TR/webrtc W3C(CR) 2009-06-21]: WebRTC 1.0: Real-time Communication Between Browsers | ||
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* [http://tools.ietf.org/html/rfc7741 IETF RFC 7741(2016)]: RTP Payload Format for VP8 Video | * [http://tools.ietf.org/html/rfc7741 IETF RFC 7741(2016)]: RTP Payload Format for VP8 Video | ||
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* [[WebRTC/RTP Server: Release Notes]] | * [[WebRTC/RTP Server: Release Notes]] | ||
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==You Might Also Need== | ==You Might Also Need== | ||
| − | * {{LinkDescription| | + | * {{LinkDescription|QuickSync Encoder}} |
| + | * {{LinkDescription|Opus Encoder}} | ||
* {{LinkDescription|Video Generator}} | * {{LinkDescription|Video Generator}} | ||
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* {{LinkDescription|H.264/AVC Decoder}} | * {{LinkDescription|H.264/AVC Decoder}} | ||
* {{LinkDescription|Audio Mixer}} It can be useful when working with third-party audio codecs | * {{LinkDescription|Audio Mixer}} It can be useful when working with third-party audio codecs | ||
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| − | = | + | ===Action Items=== |
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* {{WebRTC/RTP Server: Download}} | * {{WebRTC/RTP Server: Download}} | ||
Latest revision as of 11:43, 22 April 2023
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
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
- QuickSync Encoder - DirectShow filter for encoding 8-bit 4:2:0 progressive or interlaced video frames in HEVC, H264 or MPEG-2 formats.
- Opus Encoder - Opus Encoder for encode PCM stream to Opus format.
- Video Generator - DirectShow filter for generating a video stream with the necessary parameters, for creating video frames for checking chroma key, for outputting an audio stream synchronized with video, and much more.
Related Products
- AAC Encoder - A high-performance AAC audio encoder.
- RTMP Server - DirectShow Filter for streaming media in the network
- 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
Action Items
Prices
| Single Application License | 1750 EUR <div class="floatright"> </div>
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| Single Developer License | 2950 EUR <div class="floatright"></div>
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| Team License | 3950 EUR <div class="floatright"></div>
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