WebRTC/RTP Server

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<!--TITLE:WebRTC/RTP Server DirectShow filter-->
 
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<!--DESCRIPTION:DirectShow Filter for sending media streams from the DS graph to the network.-->
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<!--DESCRIPTION:DirectShow Filter for streaming media in the network.-->
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<!--KEYWORDS:WebRTC RTP DirectShow filter H264 VP8 Opus-->
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{{WebRTC/RTP Server: Description}}
 
{{WebRTC/RTP Server: Description}}
  
 
==Overview==
 
==Overview==
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* {{WebRTC/RTP Server: Download}}
 
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* {{Contact Support}}
 
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* {{Place Order!}}
 
  
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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 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.
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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).
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* [http://avobjects.com:8050 Live stream from our server]
  
 
==Features==
 
==Features==
  
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* Compatible with most popular browsers such as Google Chrome and Microsoft Edge
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* 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.
 
* Compatibility with most mobile and desktop devices running on operating systems Windows, iOS, MacOS, Android.
 
* Supports secure connections using SRTP and DTLS.  
 
* Supports secure connections using SRTP and DTLS.  
 
* Supports video encoders H.264/AVC, VP8, VP9.
 
* Supports video encoders H.264/AVC, VP8, VP9.
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* Supports audio encoders Opus, PCMa(G.711a), PCMa(G.711u).
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* Supports audio encoders Opus, PCMA(G.711a), PCMU(G.711u).
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* Contains built-in NTTP signaling server for exchanging SDP information with WebRTC clients.
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* Contains built-in HTTP signaling server for exchanging SDP information with WebRTC peers.
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===Supported Formats===
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* Received streams can be displayed in browsers without Adobe Flash object.
  
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*
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==Supported Formats==
  
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===Supported MXF Standards===
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* H.264/AVC video
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* VP8 video
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* VP9 video
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* Opus audio
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* a-Law/PCMA/G.711a audio
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* u-Law/PCMU/G.711u audio
  
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* IETF RFC 3550(2003): RTP: A Transport Protocol for Real-Time Applications
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==Supported Standards==
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* IETF RFC 6184(2011): RTP Payload Format for H.264 Video
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* SMPTE 377M: Material Exchange Format (MXF) - File Format Specification
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* SMPTE 378M: Operational pattern 1A (Single Item, Single Package)
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* SMPTE 379M: MXF Generic Container
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* SMPTE 381M: Mapping MPEG Streams into the MXF Generic Container
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* SMPTE 382M: Mapping AES3 and Broadcast Wave Audio into the MXF Generic Container
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* SMPTE 383M: Mapping DV-DIF Data to the MXF Generic Container
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* SMPTE 386M: Mapping Type D-10 Essence Data to the MXF Generic Container
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* SMPTE 390M: Specialized Operational Pattern "Atom" (Simplified Representation of a Single Item)
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* SMPTE 422M: Mapping JPEG 2000 Codestreams into the MXF Generic Container
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* SMPTE 429-6 / ISO 26429-6:2008: MXF Track File Essence Encryption
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* SMPTE 434:  XML Encoding for Metadata and File Structure Information
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* Matsushita Electric Industrial Co.,Ltd. Specification of Content Data Structure on P2 Card.
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===See Also===
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* [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/rfc3550 IETF RFC 3550(2003)]: RTP: A Transport Protocol for Real-Time Applications
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* [http://tools.ietf.org/html/rfc3551 IETF RFC 3551(2003)]: RTP Profile for Audio and Video Conferences with Minimal Control
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* [http://tools.ietf.org/html/rfc3711 IETF RFC 3711(2004)]: The Secure Real-time Transport Protocol (SRTP)
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* [http://tools.ietf.org/html/rfc4566 IETF RFC 4566(2006)]: SDP: Session Description Protocol
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* [http://tools.ietf.org/html/rfc5389 IETF RFC 5389(2008)]: Session Traversal Utilities for NAT (STUN)
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* [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)
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* [http://tools.ietf.org/html/rfc6184 IETF RFC 6184(2011)]: RTP Payload Format for H.264 Video
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* [http://tools.ietf.org/html/rfc6347 IETF RFC 6347(2012)]: Datagram Transport Layer Security Version 1.2
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* [http://tools.ietf.org/html/rfc7231 IETF RFC 7231(2014)]: Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content
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* [http://tools.ietf.org/html/rfc7587 IETF RFC 7587(2015)]: RTP Payload Format for the Opus Speech and Audio Codec
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* [http://tools.ietf.org/html/rfc7741 IETF RFC 7741(2016)]: RTP Payload Format for VP8 Video
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==See Also==
  
 
* [[WebRTC/RTP Server: Release Notes]]
 
* [[WebRTC/RTP Server: Release Notes]]
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==You Might Also Need==
 
==You Might Also Need==
  
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* {{LinkDescription|H.264/AVC Encoder}}
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* {{LinkDescription|QuickSync Encoder}}
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* {{LinkDescription|Audio Generator}}
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* {{LinkDescription|Opus Encoder}}
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* {{LinkDescription|Video Generator}}
  
 
==Related Products==
 
==Related Products==
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* {{LinkDescription|AAC Encoder}}
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* {{LinkDescription|RTMP Server}}
 
* {{LinkDescription|H.264/AVC Decoder}}
 
* {{LinkDescription|H.264/AVC Decoder}}
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* {{LinkDescription|Audio Mixer}}
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* {{LinkDescription|Audio Mixer}} It can be useful when working with third-party audio codecs
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* {{LinkDescription|Audio Level}}
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===Other Codecs We Made===
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{{Other Codecs}}
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==Action Items==
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* {{WebRTC/RTP Server: Download}}
 
* {{WebRTC/RTP Server: Download}}
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* {{Place Order!}}
 
* {{Place Order!}}
  
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[[Category:Network filters]]
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[[Category:Special Filters]]
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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>
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