Gstreamer Opus/RaptorQ/RTP/UDP/localhost

gistfile1.txt

# Working:

## Transmitter

```
$ gst-launch-1.0 -v \
  rtpbin name=rtp fec-encoders='fec,0="raptorqenc\ mtu=400\ repair-packets=15\ repair-window=500\ symbol-size=192";' \
  uridecodebin uri=file:///tmp/foo.flac \
  ! audioconvert ! audioresample \
  ! opusenc ! queue ! rtpopuspay \
  ! rtp.send_rtp_sink_0 rtp.send_rtp_src_0 \
  ! udpsink host=127.0.0.1 port=5000 rtp.send_fec_src_0_0 \
  ! udpsink host=127.0.0.1 port=5002 async=false
```

## Receiver

```
$ gst-launch-1.0 \
  rtpbin latency=3000 fec-decoders='fec,0="raptorqdec";' name=rtp \
  udpsrc address=127.0.0.1 port=5002 \
  caps="application/x-rtp, payload=96, raptor-scheme-id=(string)6, repair-window=(string)500, t=(string)192" \
  ! rtp.recv_fec_sink_0_0 \
  udpsrc address=127.0.0.1 port=5000 caps="application/x-rtp, media=audio, clock-rate=48000, encoding-name=OPUS, payload=96" \
  ! queue ! netsim drop-probability=0.25 \
  ! rtp.recv_rtp_sink_0 rtp. \
  ! decodebin ! audioconvert ! queue ! autoaudiosink
```


# Not working:

Trying to change to "opusenc frame-size=60 bitrate=32000":

## Transmitter

```
$ gst-launch-1.0 -v \
  rtpbin name=rtp fec-encoders='fec,0="raptorqenc\ mtu=1000\ repair-packets=15\ repair-window=1500\ symbol-size=576";' \
  uridecodebin uri=file:///tmp/foo.flac \
  ! audioconvert ! audioresample \
  ! opusenc frame-size=60 bitrate=32000 ! queue ! rtpopuspay \
  ! rtp.send_rtp_sink_0 rtp.send_rtp_src_0 \
  ! udpsink host=127.0.0.1 port=5000 rtp.send_fec_src_0_0 \
  ! udpsink host=127.0.0.1 port=5002 async=false
```

## Receiver

```
$ gst-launch-1.0 \
  rtpbin latency=10000 fec-decoders='fec,0="raptorqdec";' name=rtp \
  udpsrc address=127.0.0.1 port=5002 \
  caps="application/x-rtp, payload=96, raptor-scheme-id=(string)6, repair-window=(string)1500, t=(string)192" \
  ! rtp.recv_fec_sink_0_0 \
  udpsrc address=127.0.0.1 port=5000 caps="application/x-rtp, media=audio, clock-rate=48000, encoding-name=OPUS, payload=96" \
  ! queue ! netsim drop-probability=0.25 \
  ! rtp.recv_rtp_sink_0 rtp. \
  ! decodebin ! audioconvert ! queue ! autoaudiosink
```