awskvswebrtcsink: choppy audio with rtp source.
i am trying to use awskvswebrtcsink with an rtp audio source, i am able to hear the audio on the kvs webrtc demo page, but the resulting audio is choppy. if i use osxaudiosrc or pulsesrc to get audio, the audio heard on the webpage is fine, but the audio is choppy with rtp for some reason. the rtp stream is of type opus, i have tried multiple things with it such as decoding it and sending raw audio to awskvswebrtcsink, re-encoding before sending it to awskvswebrtcsink, i've also tried this with aac audio rtp stream, but that also doesn't work. i've also tried enabled logs but the relevant audio elements doesn’t split out any errors.
pipelines:
rtp_sender: gst-launch-1.0 osxaudiosrc ! "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)48000, channels=(int)1" ! queue ! opusenc ! rtpopuspay ! udpsink host=127.0.0.1 port=5001
webrtc:
gst-launch-1.0 awskvswebrtcsink name=ws signaller::channel-name=test udpsrc port=5001 caps="application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS, payload=(int)96" ! rtpopusdepay ! opusparse ! ws.
any help is appreciated.