webrtc streaming

Switch from using WEBRTC to build front-end video chat room--Data channel ChapterIn two browsers, it is very complex to send messages for chat, games, or file transfers. Usually, we need to set up a server to forward the data, of course, the larger the size of the case, will be expanded into multiple data centers. In this case, there is a high latency and it is difficult to guarantee the privacy of the data.These issues can be addressed through the Rt

Pick Want streaming media file format plays an important role in streaming media system, so designing a reasonable file format is the most direct and effective way to improve the efficiency of streaming media server. Based on the analysis of the common streaming media system and file format, this paper In particular,

WebRTC supports the use of its own codec (limited to native development), audio, video can be. Here the video coding as an example to analyze the corresponding source code in the WebRTC.CreatepeerconnectionfactoryIn Webrtc/api/peerconnectioninterface.h there is a method Createpeerconnectionfactory, the prototype is as follows:inline rtc::scoped_refptrAs you can see, the last four parameters of Createpeercon

WebRTC supports the use of its own codec (limited to native development), audio, video can be. Here the video coding as an example to analyze the corresponding source code in the WebRTC. createpeerconnectionfactory In Webrtc/api/peerconnectioninterface.h there is a method Createpeerconnectionfactory, the prototype is as follows: Inline rtc::scoped_refptr As you

1.WebRTC Backend Service: Room server for callsThe room server is used to create and manage call session status maintenance, is the two sides call or multiparty calls, join and leave the room and so on, we temporarily follow the Google deployment on the Gae platform APPRTC this room server implementation, the Gae The app's source code can be obtained on the github.com. The implementation is a Python-based Gae app that we need to download Goog

Which framework or library is the best for use WebRTCHttp://stackoverflow.com/questions/24857637/current-state-of-javascript-webrtc-librariesI want to know the which framework or library is the best for use WebRTC. Here are a small and incomplete list of libraries/sdk out there. Any lib that I forgot, feel free to let me know:Libraries: Simplertc Rtcmulticonnection Crocodilertc Lynckia/

This article mainly introduces the multi-person video conferencing Service end architecture, the article from the blog Park Rtc.blacker, reproduced please explain the source.With the rapid development of mobile Internet, many companies want to intervene in online education, smart home, multi-person video, security monitoring and other fields, although they are video communications, but their service-side architecture and point-to-point communication big do not want the same,In most cases, single

This article is translated from WEBRTC data channelsIn two browsers, it is very complex to send messages for chat, games, or file transfers. Usually, we need to set up a server to forward the data, of course, the larger the size of the case, will be expanded into multiple data centers. In this case, there is a high latency and it is difficult to guarantee the privacy of the data.These issues can be addressed through the Rtcdatachannel API provided by

src\webrtc\examples\peerconnection\client\conductor.ccboolconductor::initializepeerconnection()1 webrtc::createpeerconnectionfactory ();src\talk\app\webrtc\peerconnectionfactory.cc1.1 New Rtc::refcountedobject1.2 bool Peerconnectionfactory::initialize ()1.2.1 cricket::mediaengineinterface* media_engine =Peerconnectionfactory::createmediaengine_w()src\talk\media\

, the receiver side decoding good performance, no mosaic phenomenon.3.2, adding the QoS module will bring a certain delay and lag, because packet retransmission is time-required.3.3, the above plan is WEBRTC inside the nack concrete realization way.The above scheme is provided by Peng Zuyuan, a senior audio and video expert from the ring, with some adjustments, and Kelly for editing and finishing.Peng has many years of audio and video codec developmen

WEBRTC IntroductionWEBRTC provides three types of APIs: MediaStream, namely Getusermedia Rtcpeerconnection Rtcdatachannel Getusermedia has been supported by Chrome, opera and Firefox.rtcpeerconnection is currently supported by Chrome, opera and Firefox. Chrome and opera provide an interface named Webkitrtcpeerconnection,firefox with the name Mozrtcpeerconnection.Rtcdatachannel is only available in Chrome, Opera 18 and Firefox 22

First, Local: Using JAVACV (Android and Java Platform recommended JAVACV), FFmpeg, OpenCV or JMF can easily get to the local camera streaming media JAVACV Series articles: JAVACV Development 1: Call native webcam video JAVACV Development in Detail 2: The implementation of the converter, push the local camera video to the streaming media server and camera recording video func