diff --git a/g3doc/sitemap.md b/g3doc/sitemap.md
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--- a/g3doc/sitemap.md
+++ b/g3doc/sitemap.md
@@ -13,6 +13,7 @@
       * [SCTP](/pc/g3doc/sctp_transport.md)
     *   Congestion control and bandwidth estimation
     *   Audio
+      * [NetEq](/modules/audio_coding/neteq/g3doc/index.md)
       * AudioEngine
         * [ADM](/modules/audio_device/g3doc/audio_device_module.md)
       * [Audio Coding](/modules/audio_coding/g3doc/index.md)
diff --git a/modules/audio_coding/neteq/g3doc/index.md b/modules/audio_coding/neteq/g3doc/index.md
new file mode 100644
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@@ -0,0 +1,102 @@
+<?% config.freshness.reviewed = '2021-04-13' %?>
+<?% config.freshness.owner = 'jakobi' %?>
+
+# NetEq
+
+NetEq is the audio jitter buffer and packet loss concealer. The jitter buffer is
+an adaptive jitter buffer, meaning that the buffering delay is continuously
+optimized based on the network conditions. Its main goal is to ensure a smooth
+playout of incoming audio packets from the network with a low amount of audio
+artifacts (alterations to the original content of the packets) while at the same
+time keep the delay as low as possible.
+
+## API
+
+At a high level, the NetEq API has two main functions:
+[`InsertPacket`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/neteq/neteq.h;l=198;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+and
+[`GetAudio`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/neteq/neteq.h;l=219;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72).
+
+### InsertPacket
+
+[`InsertPacket`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/neteq/neteq.h;l=198;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+delivers an RTP packet from the network to NetEq where the following happens:
+
+1.  The packet is discarded if it is too late for playout (for example if it was
+    reordered). Otherwize it is put into the packet buffer where it is stored
+    until it is time for playout. If the buffer is full, discard all the
+    existing packets (this should be rare).
+2.  The interarrival time between packets is analyzed and statistics is updated
+    which is used to derive a new target playout delay. The interarrival time is
+    measured in the number of GetAudio ‘ticks’ and thus clock drift between the
+    sender and receiver can be accounted for.
+
+### GetAudio
+
+[`GetAudio`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/neteq/neteq.h;l=219;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+pulls 10 ms of audio from NetEq for playout. A much simplified decision logic is
+as follows:
+
+1.  If there is 10 ms audio in the sync buffer then return that.
+2.  If the next packet is available (based on RTP timestamp) in the packet
+    buffer then decode it and append the result to the sync buffer.
+    1.  Compare the current delay estimate (filtered buffer level) with the
+        target delay and time stretch (accelerate or decelerate) the contents of
+        the sync buffer if the buffer level is too high or too low.
+    2.  Return 10 ms of audio from the sync buffer.
+3.  If the last decoded packet was a discontinuous transmission (DTX) packet
+    then generate comfort noise.
+4.  If there is no available packet for decoding due to the next packet having
+    not arrived or been lost then generate packet loss concealment by
+    extrapolating the remaining audio in the sync buffer or by asking the
+    decoder to produce it.
+
+In summary, the output is the result one of the following operations:
+
+*   Normal: audio decoded from a packet.
+*   Acceleration: accelerated playout of a decoded packet.
+*   Preemptive expand: decelerated playout of a decoded packet.
+*   Expand: packet loss concealment generated by NetEq or the decoder.
+*   Merge: audio stitched together from packet loss concealment to decoded data
+    in case of a loss.
+*   Comfort noise (CNG): comfort noise generated by NetEq or the decoder between
+    talk spurts due to discontinuous transmission of packets (DTX).
+
+## Statistics
+
+There are a number of functions that can be used to query the internal state of
+NetEq, statistics about the type of audio output and latency metrics such as how
+long time packets have waited in the buffer.
+
+*   [`NetworkStatistics`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/neteq/neteq.h;l=273;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    instantaneous values or stats averaged over the duration since last call to
+    this function.
+*   [`GetLifetimeStatistics`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/neteq/neteq.h;l=280;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    cumulative stats that persist over the lifetime of the class.
+*   [`GetOperationsAndState`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/neteq/neteq.h;l=284;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    information about the internal state of NetEq (is only inteded to be used
+    for testing and debugging).
+
+## Tests and tools
+
+*   [`neteq_rtpplay`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc;drc=cee751abff598fc19506f77de08bea7c61b9dcca):
+    Simulate NetEq behavior based on either an RTP dump, a PCAP file or an RTC
+    event log. A replacement audio file can also be used instead of the original
+    payload. Outputs aggregated statistics and optionally an audio file to
+    listen to.
+*   [`neteq_speed_test`](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc;drc=2ab97f6f8e27b47c0d9beeb8b6ca5387bda9f55c):
+    Measure performance of NetEq, used on perf bots.
+*   Unit tests including bit exactness tests where RTP file is used as an input
+    to NetEq, the output is concatenated and a checksum is calculated and
+    compared against a reference.
+
+## Other responsibilities
+
+*   Dual-tone multi-frequency signaling (DTMF): receive telephone events and
+    produce dual tone waveforms.
+*   Forward error correction (RED or codec inband FEC): split inserted packets
+    and prioritize the payloads.
+*   NACK (negative acknowledgement): keep track of lost packets and generate a
+    list of packets to NACK.
+*   Audio/video sync: NetEq can be instructed to increase the latency in order
+    to keep audio and video in sync.