WebRTC/RTP Server

From AVObjects Knowledge Base
(Difference between revisions)
Jump to: navigation, search
 
(24 intermediate revisions by 3 users not shown)
Line 1: Line 1:
 
<!--RM_ID:143-->
 
<!--RM_ID:143-->
 
<!--TITLE:WebRTC/RTP Server DirectShow filter-->
 
<!--TITLE:WebRTC/RTP Server DirectShow filter-->
−
<!--DESCRIPTION:DirectShow Filter or streaming media in the network.-->
+
<!--DESCRIPTION:DirectShow Filter for streaming media in the network.-->
 
<!--KEYWORDS:WebRTC RTP DirectShow HTTP streaming H264 Opus -->
 
<!--KEYWORDS:WebRTC RTP DirectShow HTTP streaming H264 Opus -->
−
{{This|products/special_filters/webrtc_server.html}}
+
{{This|products/network_filters/webrtc_server.html}}
  
 
{{WebRTC/RTP Server: Description}}
 
{{WebRTC/RTP Server: Description}}
  
 
==Overview==
 
==Overview==
−
* {{WebRTC/RTP Server: Download}}
 
−
* {{Contact Support}}
 
−
* {{Place Order!}}
 
  
−
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.
+
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==
Line 22: Line 20:
 
* Supports audio encoders Opus, PCMA(G.711a), PCMU(G.711u).
 
* Supports audio encoders Opus, PCMA(G.711a), PCMU(G.711u).
 
* Contains built-in HTTP signaling server for exchanging SDP information with WebRTC peers.
 
* Contains built-in HTTP signaling server for exchanging SDP information with WebRTC peers.
−
* Display video in browsers without Adobe Flash
+
* Received streams can be displayed in browsers without Adobe Flash object.
  
−
===Supported Formats===
+
==Supported Formats==
  
 
* H.264/AVC video
 
* H.264/AVC video
Line 33: Line 31:
 
* u-Law/PCMU/G.711u audio
 
* u-Law/PCMU/G.711u audio
  
−
===Supported Standards===
+
==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
Line 40: Line 38:
 
* [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/rfc5766 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
Line 47: Line 46:
 
* [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
  
−
===See Also===
+
==See Also==
  
 
* [[WebRTC/RTP Server: Release Notes]]
 
* [[WebRTC/RTP Server: Release Notes]]
−
<!--* [[WebRTC/RTP Server: Technical Specs]]-->
+
* [[WebRTC/RTP Server: Technical Specs]]
  
 
==You Might Also Need==
 
==You Might Also Need==
  
−
* {{LinkDescription|H.264/AVC Encoder}}
+
* {{LinkDescription|QuickSync Encoder}}
−
* [http://avobjects.com/downloads/121 Video Generator]
+
* {{LinkDescription|Opus Encoder}}
 +
* {{LinkDescription|Video Generator}}
  
 
==Related Products==
 
==Related Products==
 +
 +
* {{LinkDescription|AAC Encoder}}
 +
* {{LinkDescription|RTMP Server}}
 
* {{LinkDescription|H.264/AVC Decoder}}
 
* {{LinkDescription|H.264/AVC Decoder}}
−
* {{LinkDescription|Audio Mixer}}
+
* {{LinkDescription|Audio Mixer}} It can be useful when working with third-party audio codecs
−
<!--
+
 
−
===Other Codecs We Made===
+
===Action Items===
−
{{Other Codecs}}
+
−
-->
+
−
==Action Items==
+
  
 
* {{WebRTC/RTP Server: Download}}
 
* {{WebRTC/RTP Server: Download}}
Line 70: Line 70:
 
* {{Place Order!}}
 
* {{Place Order!}}
  
−
<script src="file:///D:/CVS/Dev/Filters/RTP/WRtpServer/Samples/HTML/WRtpServerTest/main1.js"></script>
+
{{Prices|1750|2950|3950|300987440|300987441|300987442}}
−
<!--
+
 
−
{{Prices|1600|2800|3950|300606409|300606424|300606425}}
+
[[Category:Network filters]]
−
-->
+
−
[[Category:Special Filters]]
+
 
__NOTOC__
 
__NOTOC__

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

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">Buy.jpg</div>
Single Developer License 2950 EUR <div class="floatright">Buy.jpg</div>
Team License 3950 EUR <div class="floatright">Buy.jpg</div>
Retrieved from "http://wiki.avobjects.com/index.php?title=WebRTC/RTP_Server&oldid=14893"
Personal tools