webrtc

What is the role of SFU., Janus, mediasoup or medooze. on a webRTC application

由 天涯浪子 提交于 2021-02-09 04:23:10
问题 I'm using a webRTC application with a simple-peer npm package. I want to know what is the purpose of all these topics (SFU., Janus, mediasoup or medooze.) and how can I integrate them to make my application performance greater? PS: I'm using a node.js server the bundle the requesting and signaling between peers on my architecture. are those servers and services required to make my application performance well? Hope I could find an answer here ... 回答1: With regular webrtc every peer needs to

iOS — How to change video resolution in webRTC?

由 浪子不回头ぞ 提交于 2021-02-08 15:16:27
问题 I am trying to change local video resolution in webRTC. I used following method to create local video tracker: -(RTCVideoTrack *)createLocalVideoTrack { RTCVideoTrack *localVideoTrack = nil; RTCMediaConstraints *mediaConstraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:nil optionalConstraints:nil]; RTCAVFoundationVideoSource *source = [self.factory avFoundationVideoSourceWithConstraints:mediaConstraints]; localVideoTrack = [self.factory videoTrackWithSource:source trackId:@

How to run a websocket server on ws and wss at same time that they both communicate or sync data with each other? Or WSS on HTTP and WS on HTTPS?

由 早过忘川 提交于 2021-02-08 08:21:33
问题 My requirement is this that if some users connect through WS or WSS they can communicate with each other.Now if i run node server for WSS it does not run over HTTP and if run for WS then it does not Connect on HTTPS .Any solution? 回答1: After a long research at last i find this solution and is working for me as i was requiring.This is my sever.js file. /** Before running: > npm install ws Then: > node server.js > open http://localhost:8080 in the browser */ const http = require('http'); const

Remote video is black screen or blank in WebRTC

由 雨燕双飞 提交于 2021-02-08 07:58:26
问题 I have signaling server in java and websocket. It works well with local video. but Remote video is black screen or blank But it is not always a blank. If you turn off the server and turn it on again, the remote video will show up on your remote. Why does not it always come out sometimes, and sometimes it does not come out? this is my code... navigator.getUserMedia = navigator.getUserMedia || navigator.mozGetUserMedia || navigator.webkitGetUserMedia; window.RTCPeerConnection = window

webrtc编译

由 故事扮演 提交于 2021-02-08 05:31:32
在国外的win10上远程编译的，代码同步速度飞快。 set DEPOT_TOOLS_UPDATE=0 set DEPOT_TOOLS_WIN_TOOLCHAIN=0 set GYP_MSVS_VERSION=2017 set GYP_MSVS_OVERRIDE_PATH=C:\Program Files (x86)\Microsoft Visual Studio\2017\Community set GYP_GENERATORS=msvs-ninja,ninja mkdir webrtc_checkout cd webrtc_checkout fetch --nohooks webrtc cd src git branch -r git checkout branch-heads/72 gclient sync git rebase-update gclient sync gn gen out/Default --ide=vs2017 pause ninja -C out/Default ninja -C out/Debug ninja -C out/Release 来源： oschina 链接： https://my.oschina.net/u/104197/blog/3075922

Error : LNK1104 cannot open file 'pthread.lib'

由 北城以北 提交于 2021-02-08 03:46:56
问题 I am trying to compile a native Linux C++ application in Windows using Visual Studio 2017. The app uses WebRtc's Acoustic Echo Cancellation(AEC) APIs to negate echo on wav files. Following is the CmakeLists.txt file: cmake_minimum_required(VERSION 2.8) project(wav-aec) set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS}") set(CMAKE_INSTALL_RPATH_USE_LINK_PATH TRUE) add_subdirectory(gflags) add_definitions("-DWEBRTC_NS_FLOAT -DWEBRTC_WIN -DNOMINMAX") #-DWEBRTC_UNTRUSTED_DELAY -DWEBRTC_LINUX -DWEBRTC_POSIX

WebRTC over HTTPs Issue

由 久未见 提交于 2021-02-08 03:39:26
问题 I have created a simple WebRTC application that works fine in testing overlocal host; However, WEBRTC isn't much use unless you have a secure connection, as browsers now will not run GetUserMedia unless you have HTTPs, so I am 'trying' to upgrade it for SSL-TLS. Below is a screen shot of my two applications side by side, one secure ( not working ) the other non secure ( works ) As you can see above, localhost 'connects' while HTTPs 'can't establish connection'. I am new to SSL, so this may be

What is this data at the end of WebRTC candidate info?

由 不问归期 提交于 2021-02-08 03:31:51
问题 I set up a basic video chat app using the WebRTC APIs in Chrome along with a WebSocket script I wrote myself following the W3C specs and other questions here on SO. Sometimes though, when one PC sends ICE candidate info to the other PC via the WebSocket connection, a bunch of garbled text is attached to the end of the JSON-stringified candidate info. This problem only happens sometimes though, and it never happens with the SDP info sent via the createOffer and createAnswer methods. Please see

How to use the constant resolution during webrtc video transmission?

由 柔情痞子 提交于 2021-02-08 01:22:57
问题 I am using janus to build my webrtc SFU server. I need the chrome browser to send the video resolution from a start to a fixed value and remain unchanged during the transfer. Where should I set it? I tried setting the degradationPreference in the js code, but it didn't work, the resolution will still change, it seems that chrome does not support this parameter. var senderList = config.pc.getSenders(); var sender = config.pc.getSenders().find(function(s) {return s.track.kind == "video"}); if

How to use the constant resolution during webrtc video transmission?

由 喜欢而已 提交于 2021-02-08 01:15:28
问题 I am using janus to build my webrtc SFU server. I need the chrome browser to send the video resolution from a start to a fixed value and remain unchanged during the transfer. Where should I set it? I tried setting the degradationPreference in the js code, but it didn't work, the resolution will still change, it seems that chrome does not support this parameter. var senderList = config.pc.getSenders(); var sender = config.pc.getSenders().find(function(s) {return s.track.kind == "video"}); if
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