apprtc | 易学教程

How to get frame data in AppRTC iOS app for video modifications?

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问题 I am currently trying to make some modifications to the incoming WebRTC video stream in the AppRTC app for iOS in Swift (which in turn is based on this Objective-C version). To do so, I need access to the data which is stored in the frame objects of class RTCI420Frame (which is a basic class for the Objective-C implementation of libWebRTC). In particular, I need an array of bytes: [UInt8] and Size of the frames. This data is to be used for further processing & addition of some filters. The

Unable to Screencast to Apprtc using Replay Kit Over Webrtc

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问题 Hello There, I am trying to screen broadcast with latest webrtc libraries but I keep getting following error: iOS 13.0 & above: Live broadcast has stopped due to Attempted to start invalid broadcast session. less than iOS13 but greater than iOS 12.0: Live broadcast has stopped due to:(null) I would really appreciate if anyone can answer my question. Thanks var peerConnectionFactory: RTCPeerConnectionFactory? var localVideoSource: RTCVideoSource? var videoCapturer: RTCVideoCapturer? func

Unable to Screencast to Apprtc using Replay Kit Over Webrtc

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Websocket open error, websocket register error

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问题 I am facing web socket issue with https://appr.tc and unable to create or join the room. Is there any temporary break down at the server end? 回答1: its Apprtc signalling server issue. It will be back soon. Permament solution would be to set up your own Collider server Refer 来源： https://stackoverflow.com/questions/43752796/websocket-open-error-websocket-register-error

How to disable audio in webrtc mobile app(ios) without changing in framework

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问题 I am working with webrtc mobile(ios). I can't disable audio in webrtc(ios). I have got no flag to disable audio.By changing in framwork/library it can done easily. My purpose is that I have to disable audio without changing in framework/library. Can anyone help me?. 回答1: Update your question with code snippet, how you are creating mediaStrem or tracks(audio/video). Generally with default Native WebRTC Framework, RTCMediaStream localStream = [_factory mediaStreamWithStreamId:kARDMediaStreamId]

how to customise AppRTC to receive call

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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

Android webrtc not working with 3g, 4g

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问题 In android application i have user webrtc for video calling app. If mobile is connected with wifi then app work perfectly. When i change wifi to mobile data and start app then app not work and also not display video . I have user webrtc in my app and server side kurento. Below list of scenario with status : Browser Client + Mobile Client -> Both on same Wifi network(KMS Server on same wifi network) -> Working fine . Mobile Client + Mobile Client -> Both on same Wifi network(KMS Server on same

Send Replay Kit Audio samples(not Microphone) Over Webrtc

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问题 I am able to send Replaykit video samples to the remote peer but with no Audio. What to use instead of RTCVideoFrame? Is there a way I can send a Audio Video data together? Below is the snippet: var peerConnectionFactory: RTCPeerConnectionFactory? var localVideoSource: RTCVideoSource? var videoCapturer: RTCVideoCapturer? func setupVideoCapturer(){ // localVideoSource and videoCapturer will use localVideoSource = self.peerConnectionFactory!.videoSource() videoCapturer = RTCVideoCapturer()

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

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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

how to customise AppRTC to receive call

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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