diff --git a/g3doc/sitemap.md b/g3doc/sitemap.md
index 45d78fb8a9..aa9f09dc23 100644
--- a/g3doc/sitemap.md
+++ b/g3doc/sitemap.md
@@ -32,6 +32,7 @@
     *   [PeerConnection](/pc/g3doc/peer_connection.md)
     *   Desktop capture
     *   Stats
+    *   [Logging](/logging/g3doc/rtc_event_log.md)
 *   Testing
     *   Media Quality and performance
         *   [PeerConnection Framework](/test/pc/e2e/g3doc/index.md)
diff --git a/logging/g3doc/rtc_event_log.md b/logging/g3doc/rtc_event_log.md
new file mode 100644
index 0000000000..c7996e0b42
--- /dev/null
+++ b/logging/g3doc/rtc_event_log.md
@@ -0,0 +1,85 @@
+# RTC event log
+
+<?% config.freshness.owner = 'terelius' %?>
+<?% config.freshness.reviewed = '2021-06-02' %?>
+
+## Overview
+
+RTC event logs can be enabled to capture in-depth inpformation about sent and
+received packets and the internal state of some WebRTC components. The logs are
+useful to understand network behavior and to debug issues around connectivity,
+bandwidth estimation and audio jitter buffers.
+
+The contents include:
+
+*   Sent and received RTP headers
+*   Full RTCP feedback
+*   ICE candidates, pings and responses
+*   Bandwidth estimator events, including loss-based estimate, delay-based
+    estimate, probe results and ALR state
+*   Audio network adaptation settings
+*   Audio playout events
+
+## Binary wire format
+
+No guarantees are made on the wire format, and the format may change without
+prior notice. To maintain compatibility with past and future formats, analysis
+tools should be built on top of the provided
+[rtc_event_log_parser.h](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.h)
+
+In particular, an analysis tool should *not* read the log as a protobuf.
+
+## Visualization
+
+Since the logs contain a substantial amount of data, it is usually convenient to
+get an overview by visualizing them as a set of plots. Use the command:
+
+```
+out/Default/event_log_visualizer /path/to/log_file | python
+```
+
+This visualization requires matplotlib to be installed. The tool is capable of
+producing a substantial number of plots, of which only a handful are generated
+by default. You can select which plots are generated though the `--plot=`
+command line argument. For example, the command
+
+```
+out/Default/event_log_visualizer \
+  --plot=incoming_packet_sizes,incoming_stream_bitrate \
+  /path/to/log_file | python
+```
+
+plots the sizes of incoming packets and the bitrate per incoming stream.
+
+You can get a full list of options for the `--plot` argument through
+
+```
+out/Default/event_log_visualizer --list_plots /path/to/log_file
+```
+
+You can also synchronize the x-axis between all plots (so zooming or
+panning in one plot affects all of them), by adding the command line
+argument `--shared_xaxis`.
+
+
+## Viewing the raw log contents as text
+
+If you know which format version the log file uses, you can view the raw
+contents as text. For version 1, you can use the command
+
+```
+out/Default/protoc --decode webrtc.rtclog.EventStream \
+  ./logging/rtc_event_log/rtc_event_log.proto < /path/to/log_file
+```
+
+Similarly, you can use
+
+```
+out/Default/protoc --decode webrtc.rtclog2.EventStream \
+  ./logging/rtc_event_log/rtc_event_log2.proto < /path/to/log_file
+```
+
+for logs that use version 2. However, note that not all of the contents will be
+human readable. Some fields are based on the raw RTP format or may be encoded as
+deltas relative to previous fields. Such fields will be printed as a list of
+bytes.