Gstreamer -> PlainTransport send opus

Integration
#1

hello everyone

gst-launch-1.0 audiotestsrc ! audioresample ! audio/x-raw,channels=2,rate=48000 ! opusenc bitrate=20000 ! rtpopuspay pt=101 ssrc=1112223 ! udpsink host=127.0.0.1 port=20157

gstreamer sending rtp packet to mediasoup

MediaSoup PlainTransport =>

 this.AudioProducer = await this.AudioPlainTransPort.produce(
                  {
                    kind          : 'audio',
                    rtpParameters :
                    {
                      codecs :
                      [
                        {
                          mimeType     : 'audio/opus',
                          clockRate    : 48000,
                          payloadType  : 101,
                          channels     : 2,
                          rtcpFeedback : [ ],
                          parameters   : { 'sprop-stereo': 1 }
                        }
                      ],
                      encodings : [ { ssrc: 1112223 } ]
                    },
                    paused: false
                  });

but AudioProducer stats is empty

 this.AudioProducer.getStats()
                          .then(stats => console.warn("this.AudioProducer => "+JSON.stringify(stats)));

output
this.AudioProducer => []

#2

There are tons of debugs and log tags in mediasoup to diagnose what is wrong with received RTP (if any). Use them.

#3

mediasoup:Router createPlainTransport() +60ms
mediasoup:Channel request() [method:router.createPlainTransport, id:10] +17ms
mediasoup:Channel request succeeded [method:router.createPlainTransport, id:10] +1ms
mediasoup:Transport constructor() +49ms
mediasoup:PlainTransport constructor() +72ms
mediasoup:PlainTransport connect() +0ms
mediasoup:Channel request() [method:transport.connect, id:11] +0ms
mediasoup:Transport consume() +0ms
mediasoup:Channel request() [method:transport.consume, id:12] +0ms
mediasoup:Channel request succeeded [method:transport.connect, id:11] +1ms
mediasoup:Channel request succeeded [method:transport.consume, id:12] +0ms
mediasoup:Consumer constructor() +19ms
mediasoup:Consumer resume() +0ms
mediasoup:Channel request() [method:consumer.resume, id:13] +0ms
mediasoup:Channel request succeeded [method:consumer.resume, id:13] +1ms
mediasoup:Router createWebRtcTransport() +3ms
mediasoup:Channel request() [method:router.createWebRtcTransport, id:12] +1ms
mediasoup:Channel request succeeded [method:router.createWebRtcTransport, id:12] +1ms
mediasoup:Transport constructor() +5ms
mediasoup:WebRtcTransport constructor() +37ms
mediasoup:Transport consume() +2ms
mediasoup:Channel request() [method:transport.consume, id:13] +4ms
mediasoup:Channel request succeeded [method:transport.consume, id:13] +1ms
mediasoup:Consumer constructor() +8ms
mediasoup:WebRtcTransport connect() +9ms
mediasoup:Channel request() [method:transport.connect, id:14] +5ms
mediasoup:Channel request succeeded [method:transport.connect, id:14] +0ms
mediasoup:Producer getStats() +2s
mediasoup:Channel request() [method:producer.getStats, id:15] +986ms
mediasoup:Channel request succeeded [method:producer.getStats, id:15] +1ms
:warning: this.AudioProducer => []
mediasoup:Consumer getStats() +2s
mediasoup:Channel request() [method:consumer.getStats, id:16] +1s
mediasoup:Channel request succeeded [method:consumer.getStats, id:16] +1ms
:warning: this.audioConsumer => [{“bitrate”:0,“byteCount”:0,“firCount”:0,“fractionLost”:0,“kind”:“audio”,“mimeType”:“audio/opus”,“nackCount”:0,“nackPacketCount”:0,“packetCount”:0,“packetsDiscarded”:0,“packetsLost”:0,“packetsRepaired”:0,“packetsRetransmitted”:0,“pliCount”:0,“score”:10,“ssrc”:327253245,“timestamp”:1351609,“type”:“outbound-rtp”}]
mediasoup:Producer getStats() +2s
mediasoup:Channel request() [method:producer.getStats, id:17] +990ms
mediasoup:Channel request succeeded [method:producer.getStats, id:17] +0ms
:warning: this.AudioProducer => []

#4

export DEBUG=“mediasoup*” && npm run startserver

  "worker": {
        "rtcMinPort": 20000,
        "rtcMaxPort": 65535,
        "logLevel": "debug",
        "logTags:": [
            "info",
            "ice",
            "dtls",
            "rtp",
            "srtp",
            "rtcp",
            "rtx",
            "bwe",
            "score",
            "simulcast",
            "svc"
        ]
    }

no error in logs

#5 
#!/usr/bin/env bash

AUDIO_SSRC=1112223
AUDIO_PT=100
MEDIA_FILE='/home/hd/Downloads/hd2.mp4'
audioTransportPort=55517
audioTransportRtcpPort=55928
gst-launch-1.0 rtpbin name=rtpbin filesrc location=${MEDIA_FILE} ! qtdemux name=demux demux.audio_0 ! queue ! decodebin ! audioresample ! audioconvert ! opusenc ! rtpopuspay pt=100 ssrc=1112223 ! rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=${audioTransportPort} rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=${audioTransportRtcpPort} sync=false async=false

this.AudioPlainTransPort = await this.Router.createPlainTransport(
              { 
                listenIp : '127.0.0.1',
                rtcpMux  : false,
                comedia  : true
                                                  
              });
              const audioRtpPort = this.AudioPlainTransPort.tuple.localPort;
              const audioRtcpPort = this.AudioPlainTransPort.rtcpTuple.localPort;
              const ip = this.AudioPlainTransPort.tuple.localIp;
              
              
              console.log("audio ip =>>>"+ip);
              console.log("audio rtp port =>>>"+audioRtpPort);
              console.log("audio rtcp port =>>>"+audioRtcpPort);

              var random_ssrc = Date.now();
              this.AudioProducer = await this.AudioPlainTransPort.produce(
                  {
                    kind          : 'audio',
                    rtpParameters :
                    {
                      codecs :
                      [
                        {
                          mimeType     : 'audio/opus',
                          clockRate    : 48000,
                          payloadType  : 100,
                          channels     : 2,
                          rtcpFeedback	: [ ],
						            	parameters	: { 'sprop-stereo': 1}
                        }
                      ],
                      encodings : [ { ssrc: 1112223 } ]
                    }
                  });
                  console.log( "id "+this.AudioProducer.id+" audio :"+JSON.stringify(this.AudioProducer.rtpParameters ));
                  setInterval(() =>
                  {
                      if(this.AudioProducer != null)
                      {
                          this.AudioProducer.getStats()
                          .then(stats => console.warn("this.AudioProducer => "+JSON.stringify(stats)));
                      }
                      
                  }, 2000);

output

audio ip =>>>127.0.0.1
audio rtp port =>>>55517
audio rtcp port =>>>55928
id 492fd75c-c160-47f1-9aa7-1178e59c8180 audio :{"codecs":[{"mimeType":"audio/opus","clockRate":48000,"payloadType":100,"channels":2,"rtcpFeedback":[],"parameters":{"sprop-stereo":1}}],"encodings":[{"ssrc":1112223,"dtx":false}],"headerExtensions":[],"rtcp":{"reducedSize":true,"cname":"768b7cdf"}}

It working this way