Warning

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.

Firefox

Test instance

To run a new Firefox instance with a clean profile:

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

Other options:

  • [-url] <URL>: Open URL in a new tab or window.

  • -jsconsole: Start Firefox with the Browser Console.

Debug logging

Sources:

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.

In Firefox 54 and later, you can use about:networking, and select the Logging option, to change MOZ_LOG / MOZ_LOG_FILE options on the fly (without restarting the browser).

Lastly, you can also use 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 boolean prefs 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.

These are the Mozilla 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.

Some examples:

Linux:

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

Mac:

export MOZ_LOG=timestamp,rotate:200,nsHttp:5,cache2:5,nsSocketTransport:5,nsHostResolver:5
export MOZ_LOG_FILE=~/Desktop/firefox.log
/Applications/Firefox.app/Contents/MacOS/firefox-bin

Windows:

set MOZ_LOG=timestamp,rotate:200,nsHttp:5,cache2:5,nsSocketTransport:5,nsHostResolver:5
set MOZ_LOG_FILE=%TEMP%\firefox.log
"C:\Program Files\Mozilla Firefox\firefox.exe"

With command line arguments:

/usr/bin/firefox \
    -MOZ_LOG=timestamp,rotate:200,nsHttp:5,cache2:5,nsSocketTransport:5,nsHostResolver:5 \
    -MOZ_LOG_FILE=/tmp/firefox.log

Log ICE candidates / STUN / TURN:

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

/usr/bin/firefox -no-remote -profile "$(mktemp --directory)" \
    "https://localhost:8443/"

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="$PWD/firefox"

/usr/bin/firefox -no-remote -profile "$(mktemp --directory)" \
    "https://localhost:8443/"

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

Other 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

Notes:

  • The 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

/usr/bin/firefox -no-remote -profile "$(mktemp --directory)" \
    "https://localhost:8443/"

# Equivalent code for Selenium:
# firefoxOptions.addPreference("media.cubeb.sandbox", true);
# firefoxOptions.addPreference("logging.config.add_timestamp", true);
# firefoxOptions.addPreference("logging.config.sync", true);
# firefoxOptions.addPreference("logging.cubeb", 5);
# firefoxOptions.addPreference("logging.MediaTrackGraph", 5);

Safari

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

defaults write com.apple.Safari IncludeInternalDebugMenu 1

Chrome

Test instance

To run a new Chrome instance with a clean profile:

/usr/bin/google-chrome --user-data-dir="$(mktemp --directory)"

Debug logging

Sources:

/usr/bin/google-chrome --user-data-dir="$(mktemp --directory)" \
    --enable-logging=stderr \
    --log-level=0 \
    --v=0 \
    --vmodule='*/webrtc/*=2,*/libjingle/*=2,*=-2' \
    "https://localhost:8443/"

Other options:

--use-fake-device-for-media-stream \
--use-file-for-fake-audio-capture="${HOME}/test.wav" \

H.264 codec

Chrome uses OpenH264 (same lib as Firefox uses) for encoding, and FFmpeg (which is already used elsewhere in Chrome) for decoding. Feature page: https://www.chromestatus.com/feature/6417796455989248 Since Chrome 52. Bug tracker: https://bugs.chromium.org/p/chromium/issues/detail?id=500605

Autoplay: - https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#best-practices - https://www.chromium.org/audio-video/autoplay

Command-line

Chrome

export WEB_APP_HOST_PORT="198.51.100.1:8443"

/usr/bin/google-chrome \
    --user-data-dir="$(mktemp --directory)" \
    --enable-logging=stderr \
    --no-first-run \
    --allow-insecure-localhost \
    --allow-running-insecure-content \
    --disable-web-security \
    --unsafely-treat-insecure-origin-as-secure="https://${WEB_APP_HOST_PORT}" \
    "https://${WEB_APP_HOST_PORT}"

Firefox

export SERVER_PUBLIC_IP="198.51.100.1"

/usr/bin/firefox \
    -profile "$(mktemp --directory)" \
    -no-remote \
    "https://${SERVER_PUBLIC_IP}:4443/" \
    "http://${SERVER_PUBLIC_IP}:4200/#/test-sessions"

WebRTC JavaScript API

Generate an SDP Offer.

let pc1 = new RTCPeerConnection();
navigator.mediaDevices.getUserMedia({ video: true, audio: true })
.then((stream) => {
    stream.getTracks().forEach((track) => {
        console.log("Local track available: " + track.kind);
        pc1.addTrack(track, stream);
    });
    pc1.createOffer().then((offer) => {
        console.log(JSON.stringify(offer).replace(/\\r\\n/g, '\n'));
    });
});

Browser MTU

The default Maximum Transmission Unit (MTU) in the official libwebrtc implementation is 1200 Bytes (source code). 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 code). 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 adapt their output video quality according to what they detect is the network quality. Most browsers will adapt the video bitrate; in addition, Chrome also adapts the video resolution.

The maximum video bitrate is calculated by the WebRTC stack, by following a simple rule based on the video dimensions:

  • 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.

  • 1200 kbps in any case, if the video is a screen capture.

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

Browsers offer internal stats through a special web address that you can use to verify what is really being sent by their WebRTC stack.

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 video 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 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.