Hi, is there way to record broadcast?
I was trying with createRtpStreamer, i added it inside Server.js
room.on('newpeer', (peer) => { peer.on('newproducer', (producer) => { let rtpParams = { remoteIP: '127.0.0.1', remotePort: (producer.kind === 'audio') ? 10000 : 10002}; room.createRtpStreamer(producer, rtpParams).then((streamer) => { console.log('started mirroring RTP for', producer.kind); }); }); });
So when i publish broadcast i get in console:
::ffff:127.0.0.1 59300 new peer joined the room publisher
started mirroring RTP for audio
started mirroring RTP for video
I tried to access stream with vlc but i cant connect over rtmp://127.0.0.1:1935/stream.
When i try gstreamer
command:
gst-launch-1.0 -em rtpbin name=rtpbin latency=5 udpsrc port=10000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS" ! rtpbin.recv_rtp_sink_0 rtpbin. ! queue ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! voaacenc ! mux. udpsrc port=10002 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" ! rtpbin.recv_rtp_sink_1 rtpbin. ! queue ! rtph264depay ! h264parse ! mux. flvmux name=mux streamable=true ! rtmpsink sync=false location=rtmp://127.0.0.1:1935/stream
then when i publish broadcast i get in terminal:
Got message #185 from element "rtpsession0" (element): application/x-rtp-source-sdes, cname=(string)"\{f02b34db-da4f-456b-a964-0eb4d3e9f383\}";
Do you have any idea why i cant access stream?
Im still learning webrtc/mediasoup
Hi, is there way to record broadcast?
I was trying with
createRtpStreamer, i added it inside Server.jsroom.on('newpeer', (peer) => { peer.on('newproducer', (producer) => { let rtpParams = { remoteIP: '127.0.0.1', remotePort: (producer.kind === 'audio') ? 10000 : 10002}; room.createRtpStreamer(producer, rtpParams).then((streamer) => { console.log('started mirroring RTP for', producer.kind); }); }); });So when i publish broadcast i get in console:
::ffff:127.0.0.1 59300 new peer joined the room publisherstarted mirroring RTP for audiostarted mirroring RTP for videoI tried to access stream with vlc but i cant connect over
rtmp://127.0.0.1:1935/stream.When i try gstreamer
command:
gst-launch-1.0 -em rtpbin name=rtpbin latency=5 udpsrc port=10000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS" ! rtpbin.recv_rtp_sink_0 rtpbin. ! queue ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! voaacenc ! mux. udpsrc port=10002 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" ! rtpbin.recv_rtp_sink_1 rtpbin. ! queue ! rtph264depay ! h264parse ! mux. flvmux name=mux streamable=true ! rtmpsink sync=false location=rtmp://127.0.0.1:1935/streamthen when i publish broadcast i get in terminal:
Got message #185 from element "rtpsession0" (element): application/x-rtp-source-sdes, cname=(string)"\{f02b34db-da4f-456b-a964-0eb4d3e9f383\}";Do you have any idea why i cant access stream?
Im still learning webrtc/mediasoup