コード例 #1
0
        private static RTCPeerConnection Createpc(WebSocketContext context, SDPAudioVideoMediaFormat videoFormat)
        {
            var pc = new RTCPeerConnection(null);

            MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
                videoFormat
            }, MediaStreamStatusEnum.SendOnly);

            pc.addTrack(videoTrack);

            pc.onicecandidateerror        += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
            pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
            //pc.OnReceiveReport += (type, rtcp) => logger.LogDebug($"RTCP {type} report received.");
            pc.OnRtcpBye   += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
            pc.OnRtpClosed += (reason) => logger.LogDebug($"Peer connection closed, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");

            pc.onicecandidate += (candidate) =>
            {
                if (pc.signalingState == RTCSignalingState.have_local_offer ||
                    pc.signalingState == RTCSignalingState.have_remote_offer)
                {
                    context.WebSocket.Send($"candidate:{candidate}");
                }
            };

            pc.onconnectionstatechange += (state) =>
            {
                logger.LogDebug($"Peer connection state changed to {state}.");

                if (state == RTCPeerConnectionState.connected)
                {
                    logger.LogDebug("Creating RTP session to receive ffmpeg stream.");

                    _ffmpegListener.OnRtpPacketReceived += (ep, media, rtpPkt) =>
                    {
                        if (media == SDPMediaTypesEnum.video && pc.VideoDestinationEndPoint != null)
                        {
                            //logger.LogDebug($"Forwarding {media} RTP packet to webrtc peer timestamp {rtpPkt.Header.Timestamp}.");
                            pc.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
                        }
                    };
                }
            };

            return(pc);
        }
コード例 #2
0
        public async Task <RTCSessionDescriptionInit> GotOffer(RTCSessionDescriptionInit offer)
        {
            _logger.LogDebug($"SDP offer received.");
            _logger.LogTrace($"Offer SDP:\n{offer.sdp}");

            var pc = new RTCPeerConnection();

            if (_publicIPv4 != null)
            {
                var rtpPort             = pc.GetRtpChannel().RTPPort;
                var publicIPv4Candidate = new RTCIceCandidate(RTCIceProtocol.udp, _publicIPv4, (ushort)rtpPort, RTCIceCandidateType.host);
                pc.addLocalIceCandidate(publicIPv4Candidate);
            }

            if (_publicIPv6 != null)
            {
                var rtpPort             = pc.GetRtpChannel().RTPPort;
                var publicIPv6Candidate = new RTCIceCandidate(RTCIceProtocol.udp, _publicIPv6, (ushort)rtpPort, RTCIceCandidateType.host);
                pc.addLocalIceCandidate(publicIPv6Candidate);
            }

            MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
                                                               new List <SDPAudioVideoMediaFormat> {
                new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
            }, MediaStreamStatusEnum.SendRecv);

            pc.addTrack(audioTrack);
            MediaStreamTrack videoTrack = new MediaStreamTrack(new VideoFormat(VideoCodecsEnum.VP8, VP8_PAYLOAD_ID), MediaStreamStatusEnum.SendRecv);

            pc.addTrack(videoTrack);

            pc.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
            {
                pc.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
                //_logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}, SeqNum {rtpPkt.Header.SequenceNumber}.");
            };
            //peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
            //peerConnection.OnSendReport += RtpSession_OnSendReport;

            pc.OnTimeout += (mediaType) => _logger.LogWarning($"Timeout for {mediaType}.");
            pc.onconnectionstatechange += (state) =>
            {
                _logger.LogDebug($"Peer connection state changed to {state}.");

                if (state == RTCPeerConnectionState.failed)
                {
                    pc.Close("ice failure");
                }
            };

            var setResult = pc.setRemoteDescription(offer);

            if (setResult == SetDescriptionResultEnum.OK)
            {
                var offerSdp = pc.createOffer(null);
                await pc.setLocalDescription(offerSdp);

                var answer = pc.createAnswer(null);
                await pc.setLocalDescription(answer);

                _logger.LogTrace($"Answer SDP:\n{answer.sdp}");

                return(answer);
            }
            else
            {
                return(null);
            }
        }
コード例 #3
0
ファイル: Program.cs プロジェクト: shinyoshiaki/webrtc-echoes
        public async Task <RTCSessionDescriptionInit> GotOffer(RTCSessionDescriptionInit offer)
        {
            logger.LogTrace($"SDP offer received.");
            logger.LogTrace(offer.sdp);

            var pc = new RTCPeerConnection();

            if (_presetIceAddresses != null)
            {
                foreach (var addr in _presetIceAddresses)
                {
                    var rtpPort             = pc.GetRtpChannel().RTPPort;
                    var publicIPv4Candidate = new RTCIceCandidate(RTCIceProtocol.udp, addr, (ushort)rtpPort, RTCIceCandidateType.host);
                    pc.addLocalIceCandidate(publicIPv4Candidate);
                }
            }

            MediaStreamTrack audioTrack = new MediaStreamTrack(SDPWellKnownMediaFormatsEnum.PCMU);

            pc.addTrack(audioTrack);
            MediaStreamTrack videoTrack = new MediaStreamTrack(new VideoFormat(VideoCodecsEnum.VP8, VP8_PAYLOAD_ID));

            pc.addTrack(videoTrack);

            pc.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
            {
                pc.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
            };

            pc.OnTimeout += (mediaType) => logger.LogWarning($"Timeout for {mediaType}.");
            pc.oniceconnectionstatechange += (state) => logger.LogInformation($"ICE connection state changed to {state}.");
            pc.onsignalingstatechange     += () => logger.LogInformation($"Signaling state changed to {pc.signalingState}.");
            pc.onconnectionstatechange    += (state) =>
            {
                logger.LogInformation($"Peer connection state changed to {state}.");
                if (state == RTCPeerConnectionState.failed)
                {
                    pc.Close("ice failure");
                }
            };

            var setResult = pc.setRemoteDescription(offer);

            if (setResult == SetDescriptionResultEnum.OK)
            {
                var offerSdp = pc.createOffer(null);
                await pc.setLocalDescription(offerSdp);

                var answer = pc.createAnswer(null);

                logger.LogTrace($"SDP answer created.");
                logger.LogTrace(answer.sdp);

                return(answer);
            }
            else
            {
                logger.LogWarning($"Failed to set remote description {setResult}.");
                return(null);
            }
        }