Kurento is a low-level platform to create WebRTC applications from scratch. You will be responsible of managing STUN/TURN servers, networking, scalability, etc. If you are new to WebRTC, we recommend using OpenVidu instead.

OpenVidu is an easier to use, higher-level, Open Source platform based on Kurento.

Browser Details

This page is a compendium of information that can be useful to work with or configure different web browsers, for tasks that are common to WebRTC development.

Example commands are written for a Linux shell, because that’s what Kurento developers use in their day to day. But most if not all of these commands should be easily converted for use on Windows or Mac systems.


Quickstart commands

Basic command

Runs a new Firefox instance with a clean profile:

/usr/bin/firefox \
    --no-remote \
    --profile "$(mktemp --directory)"

Extended command for WebRTC testing

Requires to first write some useful settings in form of an initial user.js file:

PROFILE_DIR="$(mktemp --directory)"

tee "$PROFILE_DIR/user.js" >/dev/null <<'EOF'
// Disable first-time screens and pop-ups.
user_pref("browser.aboutwelcome.enabled", false);
user_pref("browser.newtabpage.activity-stream.feeds.topsites", false);
user_pref("browser.newtabpage.activity-stream.showSponsoredTopSites", false);
user_pref("browser.translations.enable", false);
user_pref("datareporting.policy.dataSubmissionEnabled", false);
user_pref("datareporting.policy.firstRunURL", "");
// Enable fake synthetic audio and video for media capture.
user_pref("permissions.default.camera", 1);
user_pref("permissions.default.microphone", 1);
// Mozilla prefs for testing. Taken directly from Mozilla source code:
// https://searchfox.org/mozilla-central/source/testing/profiles/web-platform/user.js
// Don't use the new tab page but about:blank for opened tabs
user_pref("browser.newtabpage.enabled", false);
// Don't restore the last open set of tabs if the browser has crashed
user_pref("browser.sessionstore.resume_from_crash", false);
// Disable session restore infobar.
user_pref("browser.startup.couldRestoreSession.count", -1);
// Don't show the Bookmarks Toolbar on any tab
user_pref("browser.toolbars.bookmarks.visibility", "never");
// Expose TestUtils interface
user_pref("dom.testing.testutils.enabled", true);
// Don't open a dialog to show available add-on updates
user_pref("extensions.update.notifyUser", false);
// Enable fake media streams for getUserMedia
user_pref("media.navigator.streams.fake", true);
// Disable permission prompt for getUserMedia
user_pref("media.navigator.permission.disabled", true);
// Enable direct connection
user_pref("network.proxy.type", 0);
// Suppress automatic safe mode after crashes
user_pref("toolkit.startup.max_resumed_crashes", -1);
// Run the font loader task eagerly for more predictable behavior
user_pref("gfx.font_loader.delay", 0);
// Disable antiphishing popup
user_pref("network.http.phishy-userpass-length", 255);
// Disable safebrowsing components
user_pref("browser.safebrowsing.blockedURIs.enabled", false);
user_pref("browser.safebrowsing.downloads.enabled", false);
user_pref("browser.safebrowsing.malware.enabled", false);
user_pref("browser.safebrowsing.phishing.enabled", false);
user_pref("browser.safebrowsing.update.enabled", false);
// Turn off update
user_pref("app.update.disabledForTesting", true);

And then you can just launch a standalone Firefox instance that points to that directory:

/usr/bin/firefox \
    --no-remote \
    --profile "$PROFILE_DIR"

Debug logging


Debug logging can be enabled with the parameters MOZ_LOG and MOZ_LOG_FILE. These are controlled either with environment variables, or command-line flags. You can also open the about:networking page, and selecting the Logging section, change MOZ_LOG / MOZ_LOG_FILE options to have them applied without restarting the browser.

You can also open about:config and set any log option into the profile preferences, by adding (right-click -> New) a variable named logging.<NoduleName>, and setting it to an integer value of 0-5. For example, setting logging.foo to 3 will set the module foo to start logging at level 3 (”Info”).

The special pref logging.config.LOG_FILE can be set at runtime to change the log file being output to, and the special booleans logging.config.sync and logging.config.add_timestamp can be used to control the sync and timestamp properties:

  • sync: print each log synchronously, this is useful to check behavior in real time or get logs immediately before crash.

  • timestamp: insert timestamp at start of each log line.

Logging Levels:

  • (0) DISABLED: indicates logging is disabled. This should not be used directly in code.

  • (1) ERROR: an error occurred, generally something you would consider asserting in a debug build.

  • (2) WARNING: a warning often indicates an unexpected state.

  • (3) INFO: an informational message, often indicates the current program state. and rare enough to be logged at this level.

  • (4) DEBUG: a debug message, useful for debugging but too verbose to be turned on normally.

  • (5) VERBOSE: a message that will be printed a lot, useful for debugging program flow and will probably impact performance.

Log categories:

  • Multimedia:

    • AudioStream:5

    • MediaCapabilities:5

    • MediaControl:5

    • MediaEncoder:5

    • MediaManager:5

    • MediaRecorder:5

    • MediaStream:5

    • MediaStreamTrack:5

    • MediaTimer:5

    • MediaTrackGraph:5

    • Muxer:5

    • PlatformDecoderModule:5

    • PlatformEncoderModule:5

    • TrackEncoder:5

    • VP8TrackEncoder:5

    • VideoEngine:5

    • VideoFrameConverter:5

    • cubeb:5

  • WebRTC:

    • Autoplay:5

    • GetUserMedia:5

    • webrtc_trace:5

    • signaling:5

    • MediaPipeline:5

    • RtpLogger:5

    • RTCRtpReceiver:5

    • sdp:5

Debug logging examples

General logging of various modules:

export MOZ_LOG="timestamp,rotate:200,nsHttp:5,cache2:5,nsSocketTransport:5,nsHostResolver:5"
export MOZ_LOG_FILE="/tmp/firefox.log"

ICE candidates / STUN / TURN:

export R_LOG_DESTINATION=stderr
export R_LOG_LEVEL=7
export R_LOG_VERBOSE=1

WebRTC dump example (see https://blog.mozilla.org/webrtc/debugging-encrypted-rtp-is-more-fun-than-it-used-to-be/):

export MOZ_LOG="timestamp,signaling:5,jsep:5,RtpLogger:5"
export MOZ_LOG_FILE="/tmp/firefox"

# Later, the resulting logs can be converted into Packet Capture files:
grep -E "(RTP_PACKET|RTCP_PACKET)" firefox.*.moz_log \
    | cut -d "|" -f 2 \
    | cut -d " " -f 5- \
    | text2pcap -D -n -l 1 -i 17 -u 1234,1235 -t "%H:%M:%S." - firefox-rtp.pcap

Media decoding (audio sandbox can be enabled or disabled with the user preference media.cubeb.sandbox):

export MOZ_LOG="timestamp,sync,MediaPipeline:5,MediaStream:5,MediaStreamTrack:5,webrtc_trace:5"

Security sandboxes

Firefox has several sandboxes that can affect the logging output. For troubleshooting and development, it is recommended that you learn which sandbox might be getting in the way of the logs you need, and disable it:

For example:

  • To get logs from MOZ_LOG="signaling:5", first set security.sandbox.content.level to 0.

  • To inspect audio issues, disable the audio sandbox by setting media.cubeb.sandbox to false.


Quickstart commands

Basic command

Runs a new Chrome instance with a clean profile:

# Depending on your system, you'll want to use either of these:
# /usr/bin/chromium
# /usr/bin/chromium-browser
# /usr/bin/google-chrome

/usr/bin/chromium \
    --user-data-dir="$(mktemp --directory)"

Extended command for WebRTC testing

/usr/bin/chromium \
    --user-data-dir="$(mktemp --directory)" \
    --guest \
    --no-default-browser-check \
    --auto-accept-camera-and-microphone-capture \
    --use-fake-device-for-media-stream \
    --enable-logging=stderr \
    --log-level=0 \
    --v=0 \


  • --guest: activate “browse without sign-in” (guest session) mode, disabling extensions, sync, bookmarks, and password manager pop-ups.

  • --no-default-browser-check: disable “set as default browser” prompt.

  • --auto-accept-camera-and-microphone-capture: automatically accept all requests to access the camera and microphone.

    This flag deprecates the older --use-fake-ui-for-media-stream, which had a negative effect on screen/tab capture.

  • --use-fake-device-for-media-stream: use synthetic audio and video media to simulate capture devices (camera, microphone, etc).

    Alternatively, a local file can be provided to be used instead:

  • --unsafely-treat-insecure-origin-as-secure="URL,...": allow insecure origins to use features that would require a Secure Context (such as getUserMedia(), WebRTC, etc.) when served from localhost or over HTTP.

    A better approach is to serve the origins over HTTPS, but this flag can be useful for one-off testing.

Debug logging


Debug logging is enabled with --enable-logging=stderr --log-level=0. With that, the maximum log level for all modules is given by --v=N (with N = 0, 1, 2, etc, higher is more verbose, default 0), and per-module levels can be set with --vmodule="<categories>".

Log categories:

  • WebRTC:

    • */webrtc/*=2: everything related to the WebRTC stack.

      It’s strongly suggested to disable some modules that would otherwise flood the logs:

      • basic_ice_controller=0

      • connection=0

      • encoder_bitrate_adjuster=0

      • goog_cc_network_control=0

      • pacing_controller=0

      • video_stream_encoder=0

    • */media/*=2: logs from the user media and device capture.

    • tls*=1: establishment of SSL/TLS connections.

    See below for a full example command that can be copy-pasted.

How to find the module names for --vmodule:

  • Run with a very verbose general logging level, such as --v=9.

  • Start with --vmodule="compositor=0,display=0,layer_tree_*=0,segment_*=0,*/metrics/*=0" (these are very noisy modules that would otherwise flood the log).

  • Search the log for the lines you are interested in. For example:

    [VERBOSE2:video_capture_metrics.cc(158)] Device supports PIXEL_FORMAT_I420 at 96x96 (0)
  • Open the Google Chromium code search page: https://source.chromium.org/chromium/chromium/src

  • Search for the desired module name. In the example, this search term would match exactly:

    file:video_capture_metrics.cc content:"Device supports"

    Take note of the module path: media/capture/video/video_capture_metrics.cc.

  • Add either the module name or path with wildcards to the --vmodule list. In the example, any of these would enable the given log message:


Packet Loss

A command line for 3% sent packet loss and 5% received packet loss is:


H.264 codec

Chrome uses OpenH264 (same lib as Firefox uses) for encoding, and FFmpeg (which is already used elsewhere in Chrome) for decoding.



To enable the Debug menu in Safari, run this command in a terminal:

defaults write com.apple.Safari IncludeInternalDebugMenu 1

Browser MTU

The default Maximum Transmission Unit (MTU) in the official libwebrtc implementation is 1200 Bytes (source). All browsers base their WebRTC implementation on libwebrtc, so this means that all use the same MTU:

Bandwidth Estimation

WebRTC bandwidth estimation (BWE) was implemented first with Google REMB, and later with Transport-CC. Clients need to start “somewhere” with their estimations, and the official libwebrtc implementation chose to do so at 300 kbps (kilobits per second) (source). All browsers base their WebRTC implementation on libwebrtc, so this means that all use the same initial BWE:

Video Encoding

Video Bitrate

Web browsers will try to estimate the real performance of the network, and with this information they adapt their video output quality. Most browsers are able to adjust the video bitrate; in addition, Chrome also dynamically adapts the resolution and framerate of its video output.

The maximum video bitrate is calculated for WebRTC by following a simple rule based on the dimensions of the video source:

  • 600 kbps if width * height <= 320 * 240.

  • 1700 kbps if width * height <= 640 * 480.

  • 2000 kbps (2 Mbps) if width * height <= 960 * 540.

  • 2500 kbps (2.5 Mbps) for bigger video sizes.

  • Never less than 1200 kbps, if the video is a screen capture.

Source: the GetMaxDefaultVideoBitrateKbps() function in libwebrtc source code.

To verify what is exactly being sent by your web browser, check its internal WebRTC stats. For example, to check the outbound stats in Chrome:

  1. Open this URL: chrome://webrtc-internals/.

  2. Look for the stat name “Stats graphs for RTCOutboundRTPVideoStream (outbound-rtp)”.

  3. You will find the effective output bitrate in [bytesSent_in_bits/s], and the output resolution in frameWidth and frameHeight.

You can also check what is the network bandwidth estimation in Chrome:

  1. Look for the stat name “Stats graphs for RTCIceCandidatePair (candidate-pair)”. Note that there might be several of these, but only one will be active.

  2. Find the output network bandwidth estimation in availableOutgoingBitrate. Chrome will try to slowly increase its effective output bitrate, until it reaches this estimation.

H.264 profile

By default, Chrome uses this line in the SDP Offer for an H.264 media:

a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f

profile-level-id is an SDP attribute, defined in RFC 6184 as the hexadecimal representation of the Sequence Parameter Set (SPS) from the H.264 Specification. The value 42e01f decomposes as the following parameters:

  • profile_idc = 0x42 = 66

  • profile-iop = 0xE0 = 1110_0000

  • level_idc = 0x1F = 31

These values translate into the Constrained Baseline Profile, Level 3.1.

Source Code URLs

Here is where you can find URLs to the different web browser source code repositories. Also, for linking to specific lines of code, it’s always a good idea to use permalinks such that future visitors find the exact same source code that was linked, and not a newer version of it which might have changed.