# RTP In Through FFmpeg Works On Localhost But Not In Production

**URL:** <https://mediasoup.discourse.group/t/rtp-in-through-ffmpeg-works-on-localhost-but-not-in-production/5891>\
**Category:** Integration\
**Created:** [February 29, 2024, 2:04pm UTC](https://mediasoup.discourse.group/t/rtp-in-through-ffmpeg-works-on-localhost-but-not-in-production/5891 "2024-02-29T14:04:38Z")\
**Posts on this page:** 3\
**Page:** 1

<div class="post-metadata">

**Author:** ![HGB467](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/hgb467/32/1895_2.png) [@HGB467](https://mediasoup.discourse.group/u/HGB467)\
**Post date:** [February 29, 2024, 2:04pm UTC](https://mediasoup.discourse.group/t/rtp-in-through-ffmpeg-works-on-localhost-but-not-in-production/5891/1 "2024-02-29T14:04:38Z")

</div>

Hi, I am trying to re-encode an rtsp stream through ffmpeg and then sending it to mediasoup by utilizing plain transport. It works perfectly fine on localhost but in production, the remote track is always muted.

Here is the ffmpeg command i am using:

```auto
            const ffmpegArgs = [
              '-rtsp_transport', 'tcp','-i','rtsp://demo:rtspwebrtc@103.217.176.234:552', '-c:v', 'libx264', 
              '-preset', 'ultrafast',
              // '-tune', 'zerolatency',
              '-ssrc', '22222222', '-payload_type', '102',
             '-f','rtp', `rtp://37.27.0.12:${videoRtpPort}?rtcpport=${videoRtcpPort}`
            ]

```

I am creating the plain transport as below:

```auto
      const videoTransport = await router.createPlainTransport(
        { 
          listenIp : {
            ip:'0.0.0.0',
            announcedIp:'37.27.0.12'
        },
          rtcpMux : false,
          comedia : true
        });

```

and producer on it as:

```auto
        videoProducer = await videoTransport.produce(
            {
              kind : 'video',
              rtpParameters :
              {
                codecs :
                [
                  {
                    kind : "video",
                    mimeType : "video/H264",
                    payloadType: 102,
                    clockRate : 90000,
                    parameters: {
                      "packetization-mode": 1,
                      "profile-level-id": "4d0032",
                      "level-asymmetry-allowed": 1
                  },
                    rtcpFeedback: []
                  }
                ],
                encodings : [{ ssrc: 22222222 }]
              }
            });

```

I am creating webRTC transport and consumer as:

```auto
const transport = await mediasoupRouter.createWebRtcTransport({
    listenIps: [
        {
            ip:'0.0.0.0',
            announcedIp:'37.27.0.12'
        }
    ],
    enableUdp:true,
    enableTcp:true,
    preferUdp:true,
    enableSctp:true,
    initialAvailableOutgoingBitrate: 1000000
   })

  const VideoConsumer = await transport.consume({
            producerId:videoProducer?.id,
            rtpCapabilities:data?.rtpCapabilities,
            paused:false
        })

```

If i get the stats of plain transport, it shows that bytes are received regularly but while consuming it shows the track is muted. Any help will be much appreciated.

---

<div class="post-metadata">

**Author:** ![HGB467](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/hgb467/32/1895_2.png) [@HGB467](https://mediasoup.discourse.group/u/HGB467)\
**Post date:** [March 2, 2024, 3:19am UTC](https://mediasoup.discourse.group/t/rtp-in-through-ffmpeg-works-on-localhost-but-not-in-production/5891/2 "2024-03-02T03:19:11Z")

</div>

So i was able to solve it myself. Issue was related to ffmpeg and rtcp feedback. As in ffmpeg, there was no mechanism for NACK/PLI And Fir. I shifted to gstreamer and everything worked like a breeze. Gstreamer commad for anyone interested:

```auto
pipeline = Gst.parse_launch(f'''
    rtpbin name=r
    rtspsrc location=rtsp://demo:rtspwebrtc@103.217.176.234:552 !
    rtph265depay ! avdec_h265 ! videoconvert ! x264enc speed-preset=ultrafast !
    rtph264pay ! capssetter caps=application/x-rtp,payload=(int)102,clock-rate=(int)90000,ssrc=(uint)22222222,rtcp-fb-nack-pli=(int)1 !
    rtprtxqueue max-size-time=3000 ! 
    r.send_rtp_sink_0 r.send_rtp_src_0 !
    udpsink host=127.0.0.1 port={sys.argv[1]} r.send_rtcp_src_0 ! udpsink host=127.0.0.1 port={sys.argv[2]} sync=false async=false
    udpsrc ! r.recv_rtcp_sink_0
''')

```

New video producer setup:

```auto
        videoProducer = await videoTransport.produce(
            {
              kind : 'video',
              rtpParameters :
              {
                codecs :
                [
                  {
                    kind : "video",
                    mimeType : "video/H264",
                    payloadType: 102,
                    clockRate : 90000,
                    parameters: {
                      "packetization-mode": 1,
                      "profile-level-id": "4d0032",
                      "level-asymmetry-allowed": 1
                  },
                    rtcpFeedback: [
                      { type: 'nack' },
                      { type: 'nack', parameter: 'pli' },
                      { type: 'cm', parameter: 'fir' },
                      { type: 'goog-remb' }
                    ]
                  }
                ],
                encodings : [{ ssrc: 22222222 }]
              }
            });

```

Note: I have used Gstreamer bindings for python

---

<div class="post-metadata">

**Author:** ![vaibhav666-glitch](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/vaibhav666-glitch/32/2819_2.png) [@vaibhav666-glitch](https://mediasoup.discourse.group/u/vaibhav666-glitch)\
**Post date:** [March 31, 2025, 5:18am UTC](https://mediasoup.discourse.group/t/rtp-in-through-ffmpeg-works-on-localhost-but-not-in-production/5891/3 "2025-03-31T05:18:20Z")

</div>

how did you solve it, even i have tried with both ffmpeg and gstreamer but still unable to solve it, and how can we find correct ssrc value?