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Add tools to support RTP stream saving/playback #252

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maxhawkins opened this issue Mar 25, 2019 · 3 comments
Open

Add tools to support RTP stream saving/playback #252

maxhawkins opened this issue Mar 25, 2019 · 3 comments

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@maxhawkins
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Summary

Add libraries that make it easier to play and record RTP streams.

Motivation

It's useful to play pre-recorded streams directly to a PeerConnection to support playback of audio/video files without transcoding.

maxhawkins referenced this issue in pion/webrtc Mar 25, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
maxhawkins referenced this issue in pion/webrtc Mar 25, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
maxhawkins referenced this issue in pion/webrtc Mar 25, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
maxhawkins referenced this issue in pion/webrtc Mar 25, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
maxhawkins referenced this issue in pion/webrtc Mar 30, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
maxhawkins referenced this issue in pion/webrtc Mar 31, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
maxhawkins referenced this issue in pion/webrtc Apr 4, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
maxhawkins referenced this issue in pion/webrtc Apr 4, 2019
RTPDump is a widely-implemented file format for saving RTP
packet dumps without the overhead of UDP and IP headers found
in pcap dumps. libWebRTC, Wireshark, and RTPTools all have
an implementation.

For more information see:
https://www.cs.columbia.edu/irt/software/rtptools

Relates to #549
@Sean-Der
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Sean-Der commented Apr 7, 2019

@maxhawkins This all landed right? It would be cool to have an example for this, currently we don't have a way to do easy 'playback'. I think this could be it though!

also maybe taking in a webm or something easy for users to play with.

@maxhawkins
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I'm leaving this open to remind myself to add a Player/Recorder to rtpdump. I merged the reader/writer for the file format but I haven't done playback/time tracking.

@maxhawkins
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One nice thing the rtpdump utility does is provide a way to pipe in RTP streams and save as a file.

You can hook it up to gstreamer, for instance.

rtpdump -Fdump -o example.rtpdump 0.0.0.0/5000
gst-launch filesrc location=example.wav ! decodebin \
    ! audioconvert ! audioresample ! opusenc ! rtpopuspay \
    ! application/x-rtp,media=audio,encoding-name=OPUS,payload=111 \
    ! udpsink host=0.0.0.0 port=5000

Maybe we could add a gstreamer-based utility for converting input audio/video into output opus/webm streams saved as rtpdump files.

@Sean-Der Sean-Der transferred this issue from pion/webrtc Apr 29, 2024
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