RTMP - Increase quality output

[RafaelRibeiro]

Hello!

We are using WEBRTC to RTMP streaming.
We capture áudio, video and screen from user and re-transmit it over RTMP by default from flashphoner server.

How to increase bitrate and audio / video quality?

 

[Max]

Good day.
RTMP republished stream will always have the same resolution/bitrate as incoming WebRTC stream. You can manage WebRTC publishing bitrate as described here. Please note that incoming resolution/bitrate depemds on channel quality and bandwidth from publisher to server.
To increase republished audio quality, you can use custom SDP file media_transponder.sdp

v=0
o=- 1988962254 1988962254 IN IP4 0.0.0.0
c=IN IP4 0.0.0.0
t=0 0
a=sdplang:en
m=video 0 RTP/AVP 95 96
a=rtpmap:95 H264/90000
a=fmtp:95 profile-level-id=42e01f;packetization-mode=0
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42e01f;packetization-mode=1
a=recvonly
m=audio 0 RTP/AVP 108 103
a=rtpmap:108 mpeg4-generic/48000/1
a=rtpmap:96 mpeg4-generic/8000/1
a=rtpmap:97 mpeg4-generic/11025/1
a=rtpmap:98 mpeg4-generic/12000/1
a=rtpmap:99 mpeg4-generic/16000/1
a=rtpmap:100 mpeg4-generic/22050/1
a=rtpmap:104 mpeg4-generic/24000/1
a=rtpmap:102 mpeg4-generic/32000/1
a=rtpmap:103 mpeg4-generic/44100/1
a=recvonly

This file should be placed to /usr/local/FlashphonerWebCallServer/conf folder, WCS should be restarted to apply changes.