webrtc

问题 I want to build a mobile application where one of the participating users can broadcast audio at a time to other participants can only listen audio, Ant Media, Jitsi and Janus Which one is best to start live voice stream one to many in Mobile Applications? 回答1: In my opinion [as a stream engineer] Before trying to choose a webrtc sfu, decide why you need webrtc technology ? For one to many stream, hls is better and cheaper, easy to use with 3rd party cdn. If you need to publish stream via

问题 I want to build a mobile application where one of the participating users can broadcast audio at a time to other participants can only listen audio, Ant Media, Jitsi and Janus Which one is best to start live voice stream one to many in Mobile Applications? 回答1: In my opinion [as a stream engineer] Before trying to choose a webrtc sfu, decide why you need webrtc technology ? For one to many stream, hls is better and cheaper, easy to use with 3rd party cdn. If you need to publish stream via

问题 Overview I would like to use a custom video source to live stream video via WebRTC Android implementation. If I understand correctly, existing implementation only supports front and back facing cameras on Android phones. The following classes are relevant in this scenario: Camera1Enumerator.java VideoCapturer.java PeerConnectionFactory VideoSource.java VideoTrack.java Currently for using front facing camera on Android phone I'm doing the following steps: CameraEnumerator enumerator = new

问题 Overview I would like to use a custom video source to live stream video via WebRTC Android implementation. If I understand correctly, existing implementation only supports front and back facing cameras on Android phones. The following classes are relevant in this scenario: Camera1Enumerator.java VideoCapturer.java PeerConnectionFactory VideoSource.java VideoTrack.java Currently for using front facing camera on Android phone I'm doing the following steps: CameraEnumerator enumerator = new

问题 Now I test webrtc communicate with SIP Client(sx20) I send invite message with webrtc sdp. but sip client answer has not finger print, and sip client answer is not SRTP just RTP. So I need to turn off SRTP in WEBRTC. Can I do it? and one more question... I think, finger print is encrypt key, so, sender and receiver need to each key? we need two key? sender's key, receive's key? or we need just sender' key? 回答1: You cannot turn off SRTP. It is part of the standard and it will probably never be

问题 Now I test webrtc communicate with SIP Client(sx20) I send invite message with webrtc sdp. but sip client answer has not finger print, and sip client answer is not SRTP just RTP. So I need to turn off SRTP in WEBRTC. Can I do it? and one more question... I think, finger print is encrypt key, so, sender and receiver need to each key? we need two key? sender's key, receive's key? or we need just sender' key? 回答1: You cannot turn off SRTP. It is part of the standard and it will probably never be

问题 Now I test webrtc communicate with SIP Client(sx20) I send invite message with webrtc sdp. but sip client answer has not finger print, and sip client answer is not SRTP just RTP. So I need to turn off SRTP in WEBRTC. Can I do it? and one more question... I think, finger print is encrypt key, so, sender and receiver need to each key? we need two key? sender's key, receive's key? or we need just sender' key? 回答1: You cannot turn off SRTP. It is part of the standard and it will probably never be

问题 Now I test webrtc communicate with SIP Client(sx20) I send invite message with webrtc sdp. but sip client answer has not finger print, and sip client answer is not SRTP just RTP. So I need to turn off SRTP in WEBRTC. Can I do it? and one more question... I think, finger print is encrypt key, so, sender and receiver need to each key? we need two key? sender's key, receive's key? or we need just sender' key? 回答1: You cannot turn off SRTP. It is part of the standard and it will probably never be

问题 I am creating a webRTC video chat that shows a caller all active members when initiating a call from firefox and the receiver is using chrome this error is displayed "Uncaught (in promise) DOMException: Failed to execute 'addIceCandidate' on 'RTCPeerConnection': Error processing ICE candidate". And when a call is initiated from firefox and receiver uses firefox I get two errors Invalidstate: cannot add ICE candidate when there is no remote SDP and ICE failed, add a STUN and see about:webrtc

问题 I'm trying to follow this example by Dan Ristic for RTCDataChannel browser p2p communication with Google's Channel API for signaling. It seems to be failing silently - I can't get the RTCDataChannel.onopen , RTCPeerConnection.onicecandidate , or RTCPeerConnection.ondatachannel events to fire. Client JS/HTML: <html> <head> <script src="https://code.jquery.com/jquery-1.11.2.min.js"></script> <script type="text/javascript" src="/_ah/channel/jsapi"></script> <script> $(document).ready(function(){