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Add example RtspToWebRTCAudioAndVideo (sipsorcery-org#922)
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‎examples/WebRTCExamples/RtspToWebRTCAudioAndVideo/.gitignore

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bin/
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obj/
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**/.vs/
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‎examples/WebRTCExamples/RtspToWebRTCAudioAndVideo/DemuxerConfig.cs

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using System;
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using System.Collections.Generic;
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using System.Linq;
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using System.Text;
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using System.Threading.Tasks;
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namespace RtspToWebRtcRestreamer
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{
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internal class DemuxerConfig
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{
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private Dictionary<StreamsEnum, string> commandTemplateDic = new Dictionary<StreamsEnum, string>()
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{
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{StreamsEnum.video, "-re -i {0} -an -vcodec {1} -ssrc {2} -f rtp rtp://{3}:{4} -sdp_file {5}" },
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{StreamsEnum.audio, "-re -i {0} -vn -acodec {1} -ssrc {2} -f rtp rtp://{3}:{4} -sdp_file {5}" },
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{StreamsEnum.videoAndAudio, "-use_wallclock_as_timestamps 1 -i {0} -map 0:v -c:v {1} -ssrc {2} -f rtp rtp://{3}:{4} -map 0:a -c:a {5} -ssrc {6} -f rtp rtp://{3}:{7} -sdp_file {8} -y"},
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{StreamsEnum.videoAndAudioUdp, "-use_wallclock_as_timestamps 1 -rtsp_transport udp -i {0} -map 0:v -c:v {1} -ssrc {2} -f rtp rtp://{3}:{4} -map 0:a -c:a {5} -ssrc {6} -f rtp rtp://{3}:{7} -sdp_file {8} -y" }
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};
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// exe path
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public string SdpFolder { get; private set; } = "A:\\temp\\sdp";
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public string SdpFileName { get; private set; } = "stream.sdp";
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public string FfmpegBinaryFolder { get; private set; } = "C:\\Program Files\\ffmpeg\\bin";
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// ffmpeg command settings
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public string rtspUrl = "rtsp://admin:[email protected]:554/ISAPI/Streaming/Channels/101";
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public string vcodec = "h264";
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public string acodec = "pcm_alaw";
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public int audioPort = 5204;
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public int videoPort = 5202;
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public uint audioSsrc = 50;
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public uint videoSsrc = 60;
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public string serverIP = "127.0.0.1";
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public StreamsEnum outputStream = StreamsEnum.videoAndAudio;
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public string sdpPath
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{
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get { return Path.Combine(SdpFolder, SdpFileName); }
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}
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public string Args
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{
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get
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{
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switch (outputStream)
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{
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case StreamsEnum.videoAndAudio:
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return String.Format(commandTemplateDic[StreamsEnum.videoAndAudio],
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rtspUrl,
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vcodec,
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videoSsrc,
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serverIP,
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videoPort,
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acodec,
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audioSsrc,
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audioPort,
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sdpPath
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);
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case StreamsEnum.videoAndAudioUdp:
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return String.Format(commandTemplateDic[StreamsEnum.videoAndAudioUdp],
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rtspUrl,
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vcodec,
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videoSsrc,
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serverIP,
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videoPort,
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acodec,
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audioSsrc,
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audioPort,
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sdpPath
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);
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case StreamsEnum.audio:
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return String.Format(commandTemplateDic[StreamsEnum.audio],
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rtspUrl,
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acodec,
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audioSsrc,
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serverIP,
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audioPort,
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sdpPath
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);
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case StreamsEnum.video:
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return String.Format(commandTemplateDic[StreamsEnum.video],
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rtspUrl,
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vcodec,
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videoSsrc,
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serverIP,
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videoPort,
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sdpPath
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);
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default:
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return "";
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}
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}
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}
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}
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}

‎examples/WebRTCExamples/RtspToWebRTCAudioAndVideo/FFmpegDemuxer.cs

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using SIPSorcery.Net;
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using System;
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using System.Collections.Generic;
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using System.Diagnostics;
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using System.Net;
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using System.Linq;
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using System.Text;
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using System.Threading.Tasks;
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namespace RtspToWebRtcRestreamer
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{
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/// <summary>
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/// Create FFmpeg process that split rtsp stream into two RTP stream (audio+video)
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/// </summary>
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internal class FFmpegDemuxer
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{
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private Process _ffmpegProcess;
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private DemuxerConfig _dc;
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public DemuxerConfig GetConfig()
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{
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return _dc;
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}
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public FFmpegDemuxer(DemuxerConfig config)
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{
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_dc = config;
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}
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public void Run()
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{
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// delete old sdp file if exist
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if (File.Exists(_dc.sdpPath))
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{
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File.Delete(_dc.sdpPath);
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}
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// configure and run ffmpeg process
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SetupAndRunProcess(ref _ffmpegProcess, _dc.Args);
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// Verification
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// wait until sdp file created and satisfy condition
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// in my case sdp file need have two track audio and video
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// if we do not check desire condition it may lead to errors due to not fully writed sdp file
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var ready = false;
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while (!ready)
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{
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if(IsOk(_dc.sdpPath) == true)
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{
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ready = true;
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break;
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}
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Task.Delay(77);
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}
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}
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private bool IsOk(string sdpFilePath)
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{
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try
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{
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if (File.Exists(sdpFilePath))
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{
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var sdp = SDP.ParseSDPDescription(File.ReadAllText(sdpFilePath));
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switch (_dc.outputStream)
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{
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case StreamsEnum.videoAndAudio:
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{
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var videoAnn = sdp.Media.First(x => x.Media == SDPMediaTypesEnum.video);
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var audioAnn = sdp.Media.First(x => x.Media == SDPMediaTypesEnum.audio);
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if (videoAnn != null && audioAnn != null)
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return true;
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return false;
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}
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case StreamsEnum.video:
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{
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var videoAnn = sdp.Media.First(x => x.Media == SDPMediaTypesEnum.video);
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if (videoAnn != null)
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return true;
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return false;
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}
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case StreamsEnum.audio:
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{
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var audioAnn = sdp.Media.First(x => x.Media == SDPMediaTypesEnum.audio);
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if (audioAnn != null)
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return true;
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return false;
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}
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default:
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return false;
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}
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}
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else
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return false;
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}
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catch(Exception ex) { return false; }
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}
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void SetupAndRunProcess(ref Process proc, string arguments)
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{
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proc = new Process();
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proc.StartInfo.FileName = Path.Combine(_dc.FfmpegBinaryFolder, "ffmpeg");
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proc.StartInfo.CreateNoWindow = true;
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proc.StartInfo.UseShellExecute = false;
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proc.StartInfo.RedirectStandardError = true;
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proc.StartInfo.RedirectStandardInput = true;
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proc.StartInfo.RedirectStandardOutput = true;
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proc.OutputDataReceived += FFMpegOutputLog;
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proc.ErrorDataReceived += FFMpegOutputError;
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proc.StartInfo.Arguments = arguments;
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proc.StartInfo.WorkingDirectory = _dc.FfmpegBinaryFolder;
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if (!Directory.Exists(_dc.SdpFolder)) Directory.CreateDirectory(_dc.SdpFolder);
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proc.Start();
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proc.BeginErrorReadLine();
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proc.BeginOutputReadLine();
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}
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private void FFMpegOutputError(object sender, DataReceivedEventArgs e)
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{
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Console.WriteLine(e.Data);
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}
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private void FFMpegOutputLog(object sender, DataReceivedEventArgs e)
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{
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Console.WriteLine(e.Data);
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}
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}
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}

‎examples/WebRTCExamples/RtspToWebRTCAudioAndVideo/FFmpegListener.cs

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using Org.BouncyCastle.Security;
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using SIPSorcery.Net;
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using SIPSorcery.SIP;
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using System;
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using System.Collections.Generic;
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using System.Linq;
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using System.Net;
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using System.Runtime.Intrinsics.Arm;
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using System.Text;
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using System.Threading.Tasks;
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namespace RtspToWebRtcRestreamer
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{
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/// <summary>
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/// Listne RTP streams created by ffmpeg process
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/// </summary>
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internal class FFmpegListener
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{
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private Thread _videoThread;
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private Thread _audioThread;
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private RTPSession _videoRTP;
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private RTPSession _audioRTP;
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private DemuxerConfig _dc;
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public bool ready = false;
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public MediaStreamTrack? videoTrack { get; private set; }
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public MediaStreamTrack? audioTrack { get; private set; }
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public SDPAudioVideoMediaFormat videoFormatRTP { get; private set; }
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public SDPAudioVideoMediaFormat audioFormatRTP { get; private set; }
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public event Action<IPEndPoint, SDPMediaTypesEnum, RTPPacket> OnAudioRtpPacketReceived;
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public event Action<IPEndPoint, SDPMediaTypesEnum, RTPPacket> OnVideoRtpPacketReceived;
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public FFmpegListener( DemuxerConfig demuxConfig)
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{
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_dc = demuxConfig;
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}
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public async void Run(CancellationToken token)
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{
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switch (_dc.outputStream)
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{
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case StreamsEnum.videoAndAudio:
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case StreamsEnum.none:
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{
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ListenAudio();
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ListenVideo();
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break;
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}
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case StreamsEnum.audio:
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{
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ListenAudio();
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break;
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}
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case StreamsEnum.video:
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{
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ListenVideo();
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break;
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}
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}
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ready = true;
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}
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private void ListenAudio()
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{
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var sdpAudio = SDP.ParseSDPDescription(File.ReadAllText(_dc.sdpPath));
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var videoAnn = sdpAudio.Media.Find(x => x.Media == SDPMediaTypesEnum.video);
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var audioAnn = sdpAudio.Media.Find(x => x.Media == SDPMediaTypesEnum.audio);
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sdpAudio.Media.Remove(videoAnn);
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// configure audio listener
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audioFormatRTP = audioAnn.MediaFormats.Values.First();
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audioTrack = new MediaStreamTrack(
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SDPMediaTypesEnum.audio,
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false,
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new List<SDPAudioVideoMediaFormat> { audioFormatRTP },
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MediaStreamStatusEnum.SendRecv);
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audioTrack.Ssrc = _dc.audioSsrc;
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_audioRTP = new RTPSession(false, false, false, IPAddress.Loopback, _dc.audioPort);
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_audioRTP.AcceptRtpFromAny = true;
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_audioRTP.SetRemoteDescription(SIPSorcery.SIP.App.SdpType.answer, sdpAudio);
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_audioRTP.addTrack(audioTrack);
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_audioRTP.OnRtpPacketReceived += HndlAudioPacketReceived;
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_audioThread = new Thread(() => _audioRTP.Start());
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_audioThread.Start();
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}
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private void ListenVideo()
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{
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// create sdpVideo
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var sdpVideo = SDP.ParseSDPDescription(File.ReadAllText(_dc.sdpPath));
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var videoAnn = sdpVideo.Media.Find(x => x.Media == SDPMediaTypesEnum.video);
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// !- its necessary to delete audio announcment from whole sdp file otherwise RTP session will not catch frames
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var audioAnn = sdpVideo.Media.Find(x => x.Media == SDPMediaTypesEnum.audio);
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sdpVideo.Media.Remove(audioAnn);
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// configure video listener
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videoFormatRTP = videoAnn.MediaFormats.Values.First();
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videoTrack = new MediaStreamTrack(
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SDPMediaTypesEnum.video,
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false,
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new List<SDPAudioVideoMediaFormat> { videoFormatRTP },
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MediaStreamStatusEnum.RecvOnly);
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videoTrack.Ssrc = _dc.videoSsrc;
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_videoRTP = new RTPSession(false, false, false, IPAddress.Loopback, _dc.videoPort);
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_videoRTP.AcceptRtpFromAny = true;
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_videoRTP.SetRemoteDescription(SIPSorcery.SIP.App.SdpType.answer, sdpVideo);
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_videoRTP.addTrack(videoTrack);
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_videoRTP.OnRtpPacketReceived += HndlVideoPacketReceived;
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_videoThread = new Thread(() => _videoRTP.Start());
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_videoThread.Start();
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}
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private void HndlVideoPacketReceived(IPEndPoint arg1, SDPMediaTypesEnum arg2, RTPPacket arg3)
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{
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if (OnVideoRtpPacketReceived == null) return;
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OnVideoRtpPacketReceived.Invoke(arg1, arg2, arg3);
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}
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private void HndlAudioPacketReceived(IPEndPoint arg1, SDPMediaTypesEnum arg2, RTPPacket arg3)
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{
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if (OnAudioRtpPacketReceived == null) return;
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OnAudioRtpPacketReceived.Invoke(arg1, arg2, arg3);
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}
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}
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}
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