jitsi webrtc

based on WebRTC of the MCU Open Source Projects Licode the environment to buildDue to the needs of the project, we need to build multi-person communication and investigate three common structures of multi-person communication:1. The previous blog has been based on Codelab for three people chatting, a multi-person system based on Mesh structure. Specifically, the fake has n+1 client, then for each client needs to establish peerconnection with other N o

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

Recently in the docking WebRTC to Android phone, there is a demand is the mobile phone horizontal screen when the other side of the image rotation, study the code of WEBRTC Video_render found that the remote video rendering using OPENGLES20 or Surfaceview implementation, Where OPENGLES20 uses hardware rendering, so performance is better, so simply add the VideoRenderOpenGles20 class in the video_render_open

1. Introduction It is well known that the browser itself does not support the direct establishment of communication channels between each other through the server to relay. For example, there are now two clients, A and B, they want to communicate, first need a and server, B and server to establish a channel between. A to send a message to B, a first send the message to the server, the server to a message to relay, sent to B, the reverse is the same. So a message between A and b through

WEBRTC FEC (forward error-correcting code) is an important part of its QoS, which can be used to recover original data packets when packet loss is lost, reduce retransmission times, reduce latency and improve video quality. It is an implementation of the RFC 5109 standard. Below, we will delve into its rationale. Redundant coding To understand the FEC in WEBRTC, you need to first understand red Packet. The

Quality Scaler is WEBRTC in accordance with the video quality, adaptive adjustment resolution of the scheme, the idea is generally: to observe the video encoding loss frame rate and QP changes, determine whether the capturer in the adaptor to adjust the encoding resolution. Its code is located at:$ (ROOT)/src/webrtc/modules/video_coding/utilityThe implementation is fairly simple,Observe the average QP and d

The main reference in this document is [1], which takes the code from the reference article. But [1] did not upload the complete code. Environment configuration can refer to the previous article [2] The main realization of the video in the local transmission. It took a day. As for WEBRTC video capture, codec, please refer to the online blog, here is not mentioned. Compile using CMake to generate makefile, step cmake. Note the points that follow, an

The Audio_device is a WEBRTC audio device module. Encapsulates audio device-related code for each platform Audio device encapsulates two sets of sound code in Android. 1. Use JNI to invoke Java's media. 2. Operate directly through the native C interface of the OpenSL es. The native interface is naturally more efficient, but the downside is that OpenSL requires Android 2.3+. OpenSL ES (Open sound Library for Embedded systems) is a hardware audio accel

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

When using WEBRTC on the Android layer, the UI changes are triggered by the native layer callback, such as when to draw the other's video window, when to indicate that both connections have been established, etc...I'm going to list what I know now for the memo.Onaddstream (), which indicates that the associated media stream has been initialized successfully (but does not establish a connection), usually at this time display the other side of the video

, Videocodec); ASSERT (IRet==ret_success); IRet= m_viecapture->Connectcapturedevice (Icaptureid, m_channelid); ASSERT (IRet==ret_success); IRet= m_viertp_rtcp->setrtcpstatus (M_channelid, webrtc::krtcpcompound_rfc4585); ASSERT (IRet==ret_success); IRet= m_viertp_rtcp->Setkeyframerequestmethod (M_channelid, webrtc::kviekeyframerequestplirtcp); ASSERT (IRet==ret_success); IRet= M_viertp_rtcp->settmmbrstatus (

The purpose of this article is to save you 10 hours (or more) of your life, or to waste 10 minutes. WEBRTC's compilation has been called a nightmare as a large cross-platform base library that Google has frequently updated. If you happen to want to compile WEBRTC under Windows, then you'd better evaluate your patience and IQ in advance. As of now, I have tried almost all the articles in the Chinese blog community, which can be said to have failed. I d

This article mainly introduces FFmpeg, the article comes from the blog Garden Rtc.blacker, supports the original, the reprint must explain the source, the individual public number blacker, more see Www.rtc.helpDescriptionPS1: If you start learning audio and video directly from WEBRTC, you may not have heard of ffmpeg, and you don't need it, but as you improve your personal abilities, you'll find it really useful.As far as I am currently exposed to the

Recently finally updated the PC version of the WEBRTC, summarized under what adjustments, the article from the blog Garden Rtc.blacker, support the original, reproduced please explain the source.Figure 1: Solution Engineering Structure Comparison:Description1, the biggest adjustment is to remove the Videoengine module, the relevant effects are as follows:1.1, Webrtcdemo inside removed video calls, voice calls still exist, but the removal is a matter o

;tm_min, pbacktime->tm_sec); - -Write (WT_FD, flag,1);//Reset Watchdog Feed the dog inAlarm2); - return; to } + - the intMain () * { $ CharFlag ='V';Panax Notoginseng intret; - intTimeout = the; the + if(Sig_err = =signal (SIGALRM, sigalarm)) A { thePerror ("Signal (sigalarm) Error"); + } - $WT_FD = open ("/dev/watchdog", O_RDWR); $ if(WT_FD 0) - { -printf"Fail to open watchdog device!\n"); the } - ElseWuyi

There are Licode and kurento in contact.Licode Flaw: Limited documentation support, Licode app client library only JSKurento Advantages: Complete Documentation, demo-ready, Packaging API is more complete. Its main features are: Networked streaming protocols, including HTTP, RTP and WebRTC. Group Communications (MCUs (Multipoint Conferencing Unit) and Sfus (Selective Forwarding unit.) functionality) Supporting B Oth Media mixing and media

Today let me find a gstreamer of a bull fork of the killer, the mind immediately thought of a general framework and plan, with Gst-inspector first object introspection property detection, and then sacrificed Gst-launcher Broadsword for pipeline test, and finally use C to achieve the pipeline logic source code , you can implement WEBRTC-based video surveillance and live streaming services. Real-time two-person video call or multi-person meeting, after

The WEBRTC audio and video parsing process consists of multiple threads:1. RTP Network stream receive thread (RTP stream reciever thread)2. Audio and video decode thread (decode thread)3. Render threads (render thread)RTP network stream receive thread (RTP stream reciever thread):Receive network RTP packets, parse RTP packets, get audio and video packets. The resolved RTP packet is added to the Rtpstreamreceiver::frame_buffer_ or eventually joined Vcm

The bandwidth adaptive algorithm in WEBRTC is divided into two types: 1, the originator bandwidth control, the principle is the RTCP in the packet loss statistics to dynamically increase or decrease the bandwidth, in the reduction of bandwidth using the TFRC algorithm to increase the smoothness. 2, the receiver bandwidth estimation, the principle is and by the receipt of RTP data, the estimated bandwidth, with the Kalman filter, the transmission time

output signal of the filter and the desired response, which is to ask for a gradient. The AEC algorithm in WEBRTC belongs to the piecewise fast frequency domain Adaptive filtering algorithm, partioned block Frequeney Domain Adaptive filter (PBFDAF).To determine whether the distal and proximal end of the conversation, also known as double-ended detection, the following four conditions need to be monitored:1. Only the far end to speak, at this time the