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Bitrate management when capturing WebRTC stream in browser


For optimal picture quality with available channel bandwidth video bitrate should be managed while capturing WebRTC stream in browser. WCS server allows to limit minimum and maximum video bitrate for stream published. Audio bitrate is not managed.

WCS builds before 5.2.1825 use REMB to manage a publishing bitrate. Since build 5.2.1825 TWCC is used by default.

Platforms and browsers supported

Chrome Firefox Safari Edge
Mac OS



Since build 5.2.1825 REMB support should be enabled by the following parameter


The following WCS settings are intended to limit publishing bitrate:

Parameter Browser side (JavaScript) Server side (
Minimum bitrate limit webrtc_cc_min_bitrate
Maximum bitrate limit webrtc_cc_max_bitrate

On browser side, bitrate limits are set in kilobits per second, for example

and on server side limits are set in bits per second


If the limits are set on both sides, browser settings take precedence over server settings.

If browser settings are not defined, server settings are applied.

If none of limits are set, default settings are applied


These settings work in most modern browsers and define the REMB bitrate management limits.

How it works

If maxBitrate is set, WCS server will send REMB command to decrease bitrate when this limit is reached.

If minBitrate is set, WCS server will stop sending REMB command to decrease bitrate when this limit is reached.

Therefore, the settings define 3 ranges with its own bitrate management algorithm:

Range Algorithm
1 [0, minBitrate] WCS stops bitrate management and not send any REMB commands
2 [minBitrate, maxBitrate] WCS server makes active bitrate management: depending on jitter and incoming traffic uniformity WCS decides to send REMB commands to decrease bitrate. If channel is good, WCS does nothing and bitrate is not decreasing
3 [maxBitrate, ...] WCS constantly sends bitrate decrease commands until it reduce to `maxBitrate`


Since build 5.2.1825, TWCC is enabled by default


When TWCC is used, bitrate may be limited via publishing constraints at browser side only, in kbps

TWCC bitrate settings work in Android SDK, iOS SDK 2.6.122 and WebSDK 2.0.239.

How to enforce bitrate increasing

Bitrate increasing can be enforced in Chromium browsers only by setting x-google-max-bitrate and  x-google-min-bitrate parameters via SDP hook.

It is impossible to enforce bitrate rising with client and server settings, only bitrate decreasing can be managed.

In this case, Chrome specific settings take precedence if they are set, i.e. constraints and server settings will be ignored.

In the latest Chrome builds, the option videoContentHint: "motion" should be set when enforcing bitrate on browser side, because Chrome drops publishing bitrate in prefer to resolution when one of the other values is set

    name: streamName,
    display: localVideo,
    videoContentHint: "motion"

Enforcing bitrate increasing in Chrome browser with server parameters

Chrome browser SDP bitrate settings can be adjusted on server side using the following parameters


These parameters are set in bps. In the example above, the settings have a same effect as


These settings are intended for Chromium-based browsers. Also, they are applied when Safari 12 or newer is used.

Bitrate enforcing does not work in Firefox.


Bitrate limiting in certain borders can be useful for example when publishing video to Safari browser subscribers. This browser is sensitive to bitrate jumps, in this case picture quality loses until freeze and browser hangs. It is recommended to stabilize bitrate when publishing streams for Safari viewers by setting narrow limits of bitrate change, for example

In this case picture quality in Safari browser will be acceptable depending on channel bandwidth and state.

Bitrate rising needs to be enforced when publishing HD and 4K streams. In this case, Chrome browser is recommended for publishing.

Known issues

1. In some Chrome versions (Chrome 75 for example) browser holds publishing bitrate on low limit when publishing WebRTC H264 stream


With SDP settings


the publishing bitrate is held on 3000 kbps while channel bandwidth is on 20 Mbps.


double bitrate limits, for example