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Since build 5.2.801, WCS is running from 'flashphoner' user for security reasons. Threrfore, when WCS is started manually
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cd /usr/local/FlashphonerWebCallServer/bin
sudo ./webcallserver start |
RTSP server cannot Therefore RTSP server may not be launched because TCP ports in range 0-1000 are ub=navailable privileged and unavailable to non-root users. In In this case RTSP port should be changed, for example
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To adjust RTSP playback parameters, for example, to change audio or video codec, SDP setting file rtsp_server.sdp should be used. Note that this file should contain WCS server IP address.
Playing H265 without transcoding
Since build 5.2.1577 it is possible to play MPEG-TS H265 stream via RTSP. To do this, H265 codec must be set in rtsp_server.sdp
file:
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Warning |
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Streams published in H264, VP8, or MPV codecs may not be played as H265! Use this codec to play MPEG-TS H265 streams only |
Dynamic codec detection
Since build 5.2.1592 codecs are detected dynamically for RTSP playback
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v=0
o=- 1988962254 1988962254 IN IP4 0.0.0.0
c=IN IP4 0.0.0.0
t=0 0
a=sdplang:en
a=range:npt=now-
a=control:*
m=audio 0 RTP/AVP
m=video 0 RTP/AVP |
In this case, if a stream is published as H264, VP8 or H265, and RTSP client supports the codec, the stream will be played via RTSP without transcoding. Audio codecs are detected by the same way.
Interleaved mode support
Before build 5.2.1609, WCS supported interleaved mode only, in this case both RTSP signaling and RTP traffic go via TCP, therefore some player applications (including VLC) may not play RTSP from WCS with default settings. Since build 5.2.1609, non-interleaved mode is also supported, in this case RTSP signaling goes via TCP, and RTP traffic flows via UDP. Note that non-interleaved mode is less packet loss proof.
Known issues
1. Frame loss and picture artefacts can occur when HD stream is played via RTSP
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