webrtc

WebRTC: SDP in Firefox with VP9 encoding

由 核能气质少年 提交于 2021-02-19 04:27:21
问题 I am not able to connect a call from Firefox to Firefox using VP9, allthough I have tried modifying the SDP in several different ways. I have a site similar to https://webrtc.github.io/samples/src/content/peerconnection/munge-sdp/, where I simply remove the unwanted codec. In Firefox, VP9 is not listed when I initiate the call. I have enabled VP9 on both sender and receiver, by setting media.mediasource.webm.enabled to true in about:config . Does anybody know how to modify it correctly to get

WebRTC with python

由 烈酒焚心 提交于 2021-02-19 00:34:35
问题 I would like to make a streaming server with python/twisted, which receives a WebRTC video stream and then applys some OpenCV algorithms to it. However I cannot find a python module for WebRTC. How can I send and receive a WebRTC video stream with python/twisted? Thanks! 回答1: What you can do is take screen shots continuously and push them to a websocket and allow your twisted server to take a gander at each one as it comes in. I have modified some common recorders and my version takes Jpeg

WebRTC with python

由 天大地大妈咪最大 提交于 2021-02-19 00:30:18
问题 I would like to make a streaming server with python/twisted, which receives a WebRTC video stream and then applys some OpenCV algorithms to it. However I cannot find a python module for WebRTC. How can I send and receive a WebRTC video stream with python/twisted? Thanks! 回答1: What you can do is take screen shots continuously and push them to a websocket and allow your twisted server to take a gander at each one as it comes in. I have modified some common recorders and my version takes Jpeg

WebRTC with python

由 ≯℡__Kan透↙ 提交于 2021-02-19 00:28:13
问题 I would like to make a streaming server with python/twisted, which receives a WebRTC video stream and then applys some OpenCV algorithms to it. However I cannot find a python module for WebRTC. How can I send and receive a WebRTC video stream with python/twisted? Thanks! 回答1: What you can do is take screen shots continuously and push them to a websocket and allow your twisted server to take a gander at each one as it comes in. I have modified some common recorders and my version takes Jpeg

WebRTC with python

由 假如想象 提交于 2021-02-19 00:28:07
问题 I would like to make a streaming server with python/twisted, which receives a WebRTC video stream and then applys some OpenCV algorithms to it. However I cannot find a python module for WebRTC. How can I send and receive a WebRTC video stream with python/twisted? Thanks! 回答1: What you can do is take screen shots continuously and push them to a websocket and allow your twisted server to take a gander at each one as it comes in. I have modified some common recorders and my version takes Jpeg

Can I re-use an “offer” in WebRTC for mulitple connections?

由 六月ゝ 毕业季﹏ 提交于 2021-02-18 21:11:51
问题 I'm starting to learn WebRTC and have a working prototype using copy/paste here: https://github.com/aerik/webrtc (the prototype is meant to be run in two browser windows, unlike many other examples that run both sides in one window) I understand that WebRTC is peer-to-peer and a I need a connection for every set of peers. However, I'm starting to think about signalling (no code yet) and I'm wondering about the "offer". In my prototype I see that clicking "create offer" multiple times results

Checking for WebRTC connectivity - reliable methods

由 夙愿已清 提交于 2021-02-18 18:15:14
问题 I have a live video chat application and I use a TURN server which supports STUN/TURN and both UPD/TCP transmission. Sometimes users can be connected to the network which blocks that much ports and protocols that WebRTC connection just cannot happen (usually those are corporate networks). I would like to check if a WebRTC connection is possible before users try to connect to each other (actually, perform a technical check ). How can I do it? Ideas I have in my head: Try to download a hosted

smart rtmpd 推流 url 和拉流 url

由 旧巷老猫 提交于 2021-02-18 12:53:33
----------------------------------------------------------------------------------------------------------------------------------------- 一分钟快速搭建 rtmpd 服务器: https://blog.csdn.net/freeabc/article/details/102880984 软件下载地址: http://www.qiyicc.com/download/rtmpd.rar github 地址：https://github.com/superconvert/smart_rtmpd ----------------------------------------------------------------------------------------------------------------------------------------- smart rtmpd 推流 url 和拉流 url smart rtmpd 支持两种推流 rtmp 和 rtsp，下面我先说说 rtmp 的相关部分： rtmp 支持两种格式的 url ，直播和录像 RTMP 直播 url 格式： // RTMP URL rtmp://您的域名或IP:端口

webRTC: How to tell Opus codec to use super wide band/full band

由 不想你离开。 提交于 2021-02-18 12:26:52
问题 I am working on a webRTC web application which works wonderfully so far. What I have not figured out yet is how to tell the Opus codec to (force) use "full band", for example. Setting the codec up for 510 kHz bit rate is easy: desc.sdp=desc.sdp.replace(/a=mid:audio\r\n/g,'a=mid:audio\r\nb=AS:510\r\n'); But is there a way to tell Opus which band to use? 回答1: Specifying the band is not that bad. With opus, you just specify the MAX rate capabilities and let it run from there. By default OPUS

Error while building WebRTC for android on Ubuntu

由 二次信任 提交于 2021-02-17 05:15:16
问题 I am trying to build WebRTC for Android on Ubuntu 16.04. I have followed the steps mentioned at: https://medium.com/@abdularis/how-to-compile-native-webrtc-from-source-for-android-d0bac8e4c933 ( I found similar steps in the other links also) I checked out the version: branch-heads/m79 I am getting the following error while building the code with the command: python tools_webrtc/android/build_aar.py ERROR at //build/config/android/internal_rules.gni:3051:7: Assertion failed. assert(_is
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