github.com/pion/webrtc/v4@v4.0.1/examples/rtp-to-webrtc/README.md (about)

     1  # rtp-to-webrtc
     2  rtp-to-webrtc demonstrates how to consume a RTP stream video UDP, and then send to a WebRTC client.
     3  
     4  With this example we have pre-made GStreamer and ffmpeg pipelines, but you can use any tool you like!
     5  
     6  ## Instructions
     7  ### Download rtp-to-webrtc
     8  ```
     9  go install github.com/pion/webrtc/v4/examples/rtp-to-webrtc@latest
    10  ```
    11  
    12  ### Open jsfiddle example page
    13  [jsfiddle.net](https://jsfiddle.net/z7ms3u5r/) you should see two text-areas and a 'Start Session' button
    14  
    15  
    16  ### Run rtp-to-webrtc with your browsers SessionDescription as stdin
    17  In the jsfiddle the top textarea is your browser's SessionDescription, copy that and:
    18  
    19  #### Linux/macOS
    20  Run `echo $BROWSER_SDP | rtp-to-webrtc`
    21  
    22  #### Windows
    23  1. Paste the SessionDescription into a file.
    24  1. Run `rtp-to-webrtc < my_file`
    25  
    26  ### Send RTP to listening socket
    27  You can use any software to send VP8 packets to port 5004. We also have the pre made examples below
    28  
    29  
    30  #### GStreamer
    31  ```
    32  gst-launch-1.0 videotestsrc ! video/x-raw,width=640,height=480,format=I420 ! vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1 ! rtpvp8pay ! udpsink host=127.0.0.1 port=5004
    33  ```
    34  
    35  #### ffmpeg
    36  ```
    37  ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -vcodec libvpx -cpu-used 5 -deadline 1 -g 10 -error-resilient 1 -auto-alt-ref 1 -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'
    38  ```
    39  
    40  If you wish to send audio replace all occurrences of `vp8` with Opus in `main.go` then run
    41  
    42  ```
    43  ffmpeg -f lavfi -i 'sine=frequency=1000' -c:a libopus -b:a 48000 -sample_fmt s16p -ssrc 1 -payload_type 111 -f rtp -max_delay 0 -application lowdelay 'rtp://127.0.0.1:5004?pkt_size=1200'
    44  ```
    45  
    46  If you wish to send H264 instead of VP8 replace all occurrences of `vp8` with H264 in `main.go` then run
    47  
    48  ```
    49  ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -pix_fmt yuv420p -c:v libx264 -g 10 -preset ultrafast -tune zerolatency -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'
    50  ```
    51  
    52  ### Input rtp-to-webrtc's SessionDescription into your browser
    53  Copy the text that `rtp-to-webrtc` just emitted and copy into second text area
    54  
    55  ### Hit 'Start Session' in jsfiddle, enjoy your video!
    56  A video should start playing in your browser above the input boxes.
    57  
    58  Congrats, you have used Pion WebRTC! Now start building something cool
    59  
    60  ## Dealing with broken/lossy inputs
    61  Pion WebRTC also provides a [SampleBuilder](https://pkg.go.dev/github.com/pion/webrtc/v3@v3.0.4/pkg/media/samplebuilder). This consumes RTP packets and returns samples.
    62  It can be used to re-order and delay for lossy streams. You can see its usage in this example in [daf27b](https://github.com/pion/webrtc/commit/daf27bd0598233b57428b7809587ec3c09510413).
    63  
    64  Currently it isn't working with H264, but is useful for VP8 and Opus. See [#1652](https://github.com/pion/webrtc/issues/1652) for the status of fixing for H264.