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WebRTC to RTMP media server heavily inspired by https://github.com/pion/webrtc/tree/master/examples/rtp-forwarder

License: Apache License 2.0

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lichao201626 496220530 alephcom toshiba01 zeusseo77 remymatheus rborgese bugswriter ifallah remarket-vn iq-scm lloyd-2014

hexcord-mediaserver's Issues

Error while unmarshal offerStr to offer.

I am getting an error at json.Unmarshal([]byte(offerStr), &offer). Any reasons why it does not work?

Example index.html with webcam or Screenshare

FFMPEG Disconnect

Hello friend, good job, the server generates the following error once FFMPEG starts:

libpostproc    55.  5.100 / 55.  5.100
[sdp @ 0x5577fa3e0d00] Could not find codec parameters for stream 1 (Video: h264, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, sdp, from 'rtp-forwarder.sdp':
  Metadata:
    title           : Pion WebRTC
  Duration: N/A, bitrate: N/A
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Stream #0:1: Video: h264, none, 90k tbr, 90k tbn, 180k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> aac (native))

Metadata:
    title           : Pion WebRTC
    encoder         : Lavf58.29.100
    Stream #0:0: Audio: aac (LC) ([10][0][0][0] / 0x000A), 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.54.100 aac
[flv @ 0x5577fa488700] Failed to update header with correct duration.
[flv @ 0x5577fa488700] Failed to update header with correct filesize.
size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[aac @ 0x5577fa46ec40] Qavg: -nan
Connection State has changed failed
Connection State has changed closed

Thanks for your reply.

hey i was writing something similar to ur code. How does ur code pushes packets to ffmpeg?

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