signaling

WebRTC datachannel with manual signaling, example please?

由 |▌冷眼眸甩不掉的悲伤 提交于 2020-06-11 07:53:26
问题 I'm really struggling to get a complete example of a WebRTC datachannel example that I can copy/paste and it works. I would like a Javascript example of WebRTC datachannel with manual signaling i.e. When the example loads, it provides the Signaling data in one text box. I copy data manually (highlight, copy) and paste it in the peer's window which has a text box to accept that signaling data. I believe there needs to be an "answer" in the signaling data, so there need to be corresponding text

Thread Signaling

由 倾然丶 夕夏残阳落幕 提交于 2020-03-02 10:50:02
本文大概意思都是都下边链接文章转换过来的，没有进行一字一句的翻译，只是把大概意思整里出来。 http://tutorials.jenkov.com/java-concurrency/thread-signaling.html 线程信号的目的是为了线程之间相互通信。线程信号可以使线程等待另其他线程信号，例如thread B 或许等待从thread A 发出的数据准备处理信号。 1 Signaling via Shared Objects 线程之间可以通过共享对象来相互发送信号。Thread A 可以通过在synchronized同步块里设置变量 hasDataToProcess为true ，线程B同样在synchronized同步块里读取hasDataToProcess的值来确定是否有数据可读。 下面是一个简单的信号对象 public class MySignal{ protected boolean hasDataToProcess = false; public synchronized boolean hasDataToProcess(){ return this.hasDataToProcess; } public synchronized void setHasDataToProcess(boolean hasData){ this.hasDataToProcess =

Failed to set local answer sdp: Called in wrong state: kStable

由 十年热恋 提交于 2020-01-01 08:56:12
问题 for a couple of days I'm now stuck with trying to get my webRTC client to work and I can't figure out what I'm doing wrong. I'm trying to create multi peer webrtc client and am testing both sides with Chrome. When the callee receives the call and create the Answer, I get the following error: Failed to set local answer sdp: Called in wrong state: kStable The receiving side correctly establishes both video connnections and is showing the local and remote streams. But the caller seems not to

how to customise AppRTC to receive call

由 限于喜欢 提交于 2019-12-23 03:23:14
问题 I have integrated AppRTC in my project and I am able to join the WebRTC call in the room of https://apprtc.appspot.com using RoomName appClient = ARDAppClient(delegate: self) appClient?.createLocalMediaStream() appClient?.connectToRoomWithId(String(roomId), options: nil) I am able to hardcode a RoomName in my app and install the app in two devices A & B and if I make a call from both the devices at same time in A & B then I am able to successfully have a WebRTC call between them . Now I'd

WebRTC Firefox to Chrome video call not working

由 给你一囗甜甜゛ 提交于 2019-12-20 04:52:07
问题 My WebRTC app works fine when I connect two of the same browsers, but when I try a combination neither respond to each others signaling messages. Something probably worth mentioning is that I have not implemented TURN, however I don't see why that should make a difference so I'm not going to change that unless I'm fairly certain it will. I don't have much of a clue where the error lies, so I will just add code on request for the sake of readability. 回答1: Make sure you enable DTLS-SRTP

How to deploy Apprtc's collider into Google App Engine?

由 扶醉桌前 提交于 2019-12-11 12:25:51
问题 I have deployed the apprtc's google app engine code on my own GAE server. I don't find instructions on how to deploy the Go based collider for websocket signaling. Here is the collider project: https://github.com/webrtc/apprtc/tree/master/src/collider Any insights would help. 回答1: Collider needs to be deployed in Google Compute Engine or equivalent services by Amazon. Here is the list of steps that I had go through: 1) Install go from: https://storage.googleapis.com/golang/go1.5.1.linux-amd64

webrtc after signaling on LAN

由 大憨熊 提交于 2019-12-11 05:33:11
问题 After webrtc passes by a signaling server on Internet, how it works when two machines are running in the same network? The data will be exchanged only on the network or will it still use internet ? I am asking this because of our internet is not good, it's too slow. but our local network speed is very fast.. So I would like to know if the internet signal will affect the audio and video conversation. Thanks a lot! 回答1: Depending on the network configuration, the devices should connect directly

how to customise AppRTC to receive call

由 荒凉一梦 提交于 2019-12-09 08:13:26
I have integrated AppRTC in my project and I am able to join the WebRTC call in the room of https://apprtc.appspot.com using RoomName appClient = ARDAppClient(delegate: self) appClient?.createLocalMediaStream() appClient?.connectToRoomWithId(String(roomId), options: nil) I am able to hardcode a RoomName in my app and install the app in two devices A & B and if I make a call from both the devices at same time in A & B then I am able to successfully have a WebRTC call between them . Now I'd like to do a real VOIP call that is make a call from my app in device A and receive the call at my app in

Failed to set local answer sdp: Called in wrong state: kStable

由 喜欢而已 提交于 2019-12-04 03:23:17
for a couple of days I'm now stuck with trying to get my webRTC client to work and I can't figure out what I'm doing wrong. I'm trying to create multi peer webrtc client and am testing both sides with Chrome. When the callee receives the call and create the Answer, I get the following error: Failed to set local answer sdp: Called in wrong state: kStable The receiving side correctly establishes both video connnections and is showing the local and remote streams. But the caller seems not to receive the callees answer. Can someone hint me what I am doing wrong here? Here is the code I am using

Native Android WebRTC application development

由 半腔热情 提交于 2019-12-03 18:34:27
问题 I am trying to create an android application for video chat and messaging by using WebRtc Native apis. I have been through several links and found out that most of the documentation for android is vague, specially if you dont know where to start from. I followed the following links, https://webrtc.org/native-code/android/# https://www.chromium.org/developers/how-tos/android-build-instructions But the above mentioned links doesnt make any sense because I want to develop an android application
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