webrtc broadcast

Continue with the unfinished part of the previous article, including the following three sections:1, extension: WEBRTC multiparty calls.2,mcu Multipoint Control Unit.2, Extension: VOIP, telephone, message communication.Note: Translation is not verbatim, but in accordance with their own understanding of the translation, at the same time, in order to facilitate understanding, but also to join some of their own organization language.Reprint please indica

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/

service received your request.In our lab, it's convenient to have a transit person come to you with a bell, and then put the information you want to tell and which lab to record. Very convenient.In the Android world, the way is a little different.The start of the service, found on the internet, there are two methods, one is in the use case of the setup to start, a boot on the start of the service. Here we choose the first type, the second one has not been tested, so dare not to be able to be su

Reprint: Android Summary series: Android broadcast mechanism1.Android Broadcast Mechanism overviewAndroid broadcasts are divided into two areas: the broadcast sender and the broadcast receiver, usually broadcastreceiver refers to the broadcast recipient (

Android Broadcast Mechanism Analysis and android Broadcast Mechanism1.1. Broadcast OverviewAndroid broadcast is different from broadcast in life. It refers to notifications generated after events in the system. Android broadcast i

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 combination of gstreamer and webrtc is a little breakthrough, gstreamerwebrtc Today, I found a fork killer in gstreamer, and quickly came up with a general framework and solution plan, using the gst-inspector to perform object introspection attribute detection first, then, the gst-launcher tool is used for Pipeline Test. Finally, the channel Logic Source Code is implemented using c to implement webrtc-

WEBRTC Technology Group: 234795279Original Address: http://blog.csdn.net/temotemo/article/details/7530504 1, WebRTC purpose WebRTC (Web real-time communication) The ultimate purpose of the project The main is to allow web developers to be based on the browser (chrome\firefox\ ... Fast and easy to develop rich real-time multimedia applications, without the need to

Recently the major live sites are compared to fire, want to explore how to play. But read a few Daniel's answer, feel there are too many unfamiliar things, try to get up a little higher cost. Found that there is a thing called WEBRTC, someone has analyzed he is not suitable for the flow of large numbers of live. But I'm just playing with it and feeling the video connectivity.The first thing I saw on GitHub was a demo of all the APIs, and an example of

based on 3GPP Ims/rcs and can be used in embedded and desktop systems. The framework is written using ansci-c and is very portable. and has been designed to be very lightweight and effective in embedded systems with low memory and low processing power. The Idoubs feature on the Apple system is based on this framework. Most of the audio and video encoding formats are supported (H264 (VIDEO), VP8 (video), ILBC (audio), pcma,pcmu,g722,g729). NAT supports Ice (Stun+turn)2) Effect measurementTest en

Recently experiment how to let WEBRTC support H264 code, record, for people who need reference. To illustrate, I was compiling the WebRTC under Ubuntu Server 14.04, using the native (c + +) API to develop WebRTC applications. So my adjustments are based on the native code. The end result is that the browser can send a video with H264 or receive H264 video. Note t

Compile WebRTC For Android code in Ubuntu 14.04 Recently, a real-time communication project for audio and video chats was developed based on Google's open-source WebRTC project. Some problems were encountered during the download of WebRTC code, which was recorded here, we also hope to help the children's shoes who encounter similar problems while downloading and

Said Nat before the penetration of a few about the concept of WEBRTC, may have been the same as the author of the WEBRTC concept of the wrong understanding of the classmate. WebRTC (Network real-time communication) It is a Web browser to support real-time voice dialogue or video dialogue technology, it provides us with video conferencing core technology, includin

The greatest feature of real-time streaming media applications is real-time, while latency is the biggest enemy of real-time sex. The processing speed of media data is the important reason of delay, and the network congestion is the main cause of delay from the point of transmission. Network congestion can cause packet loss, and may result in longer data transfer times and increased latency.Congestion control is one of the important methods in real-time streaming media application quality assura

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) In this article we continue to learn about WebRTC 's related Api,rtcpeerconnectiont and Rtcdatachannel.RtcpeerconnectionRtcpeerconnection is a

reproduced in the original: http://blog.csdn.net/u012931018/article/details/17045077 thank Bo Master. WEBRTC Echo Cancellation (Acoustic ECHOCANCELLATION,AEC, acoustic echocancellation for MOBILE,AECM) algorithm mainly includes the following important modules: echo Time delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC

WEBRTC reply content: I am in development and have a basic understanding of the WebRTC source code stack. It mainly consists of two key technologies: 1. webRTC Video/Voice Engine, including camera microphone operations, Video preprocessing, VP8 coding/decoding, and streaming media transmission (RTP/RTCP); 2. implement the P2P channel and use libjingle to complete

Recently experiment how to let WEBRTC support H264 code, record, for people who need reference.To illustrate, I was compiling the WebRTC under Ubuntu Server 14.04, using the native (c + +) API to develop WebRTC applications. So my adjustments are based on the native code.The end result is that the browser can send a video with H264 or receive H264 video.Note that