Good article, was really helpful for us.
However I have a question:
We are creating rtp streams dynamically using ffmpeg.
Is there a way to create streams dynamically, as I'm currently doing it in the streaming.cfg statically:
[stream1-rtp]
type = rtp
id = 10
description = H264 test streaming
audio = no
video = yes
videoport = 5016
videopt = 97
videortpmap = H264/90000
videofmtp = profile-level-id=42E01F\;packetization-mode=0
We will have different ports and something like 2-3 streams simultaneously.
Please provide any api examples.
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I dumped results from the admin interface when the mic was muted and then again when the mic was unmuted, yet I could not see a change in the dBov level attribute from mute to unmute. The voice stopped when I muted the mic, but the mute did not change the dBov level attribute.This is not at all important, but maybe in the future it would be nice if listeners could see when the publisher has turned off his mic.
[stream1-rtp]
type = rtp
id = 10
description = RPWC H264 test streaming
audio = no
video = yes
videoport = 5006
videopt = 97
videortpmap = H264/90000
videofmtp = profile-level-id=42E01F\;packetization-mode=0
and send video by ffmpeg
Create an audio room and send audio.