diff --git a/guides/advanced/simulcast.md b/guides/advanced/simulcast.md
new file mode 100644
index 0000000..0a3ff8c
--- /dev/null
+++ b/guides/advanced/simulcast.md
@@ -0,0 +1,142 @@
+# Simulcast
+
+Simulcast is a technique where a client sends multiple encodings of the same video to the server, which is then responsible for dynamically choosing the appropraite encoding for every peer (other client).
+Encodings differ between each other in resolution and/or frame rate.
+The selection of the encoding is based on:
+* Receiver available bandwidth.
+* Receiver preferences (e.g. explicit request to receive video at HD resolution instead of FHD).
+* UI layout (e.g. videos displayed in smaller tiles will be sent at a lower resolution).
+
+Simulcast is not utilized in direct client-client connections (no intermediate server) because in such cases, 
+the sender can adjust its resolution or frame rate based on a feedback from the receiver.
+
+Elixir WebRTC comes with:
+* Support for inbound simulcast - it allows to receive multiple incoming resolutions
+* RTP munger and keyframe detectors, which can be used for implementing encoding switching on the server side
+
+Currently there is no support for:
+* Outbound simulcast
+* Bandwidth estimation
+* Automatic encoding switching
+
+```mermaid
+flowchart LR
+  WB1((Web Browser1)) -->|low| Server
+  WB1((Web Browser1)) -->|medium| Server
+  WB1((Web Browser1)) -->|high| Server
+  Server -->|low| WB2((WebBrowser 2))
+  Server -->|high| WB3((WebBrowser 3))
+```
+
+## Turning simulcast on
+
+### Elixir WebRTC
+
+Elixir WebRTC automatically accepts incoming simulcast tracks so there are no extra steps required.
+
+### JavaScript 
+
+Simulcast can be enabled when adding a new track. For example:
+
+```js
+const pc = new RTCPeerConnection();
+
+const localStream = await navigator.mediaDevices.getUserMedia({
+  video: {
+    width: { ideal: 1280 },
+    height: { ideal: 720 },
+  },
+});
+
+pc.addTransceiver(localStream.getVideoTracks()[0], {
+  streams: [localStream],
+  sendEncodings: [
+    { rid: 'h', maxBitrate: 1500 * 1024 },
+    { rid: 'm', scaleResolutionDownBy: 2, maxBitrate: 600 * 1024 },
+    { rid: 'l', scaleResolutionDownBy: 4, maxBitrate: 300 * 1024 },
+  ],
+});
+```
+
+> #### Minimal starting resolution {: .warning}
+> To run 3 simulcast encodings, the minimal starting resolution
+> must be 960x540. See more [here](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=79?q=simulcast.cc).
+
+
+## Receiving simulcast packets
+
+When simulcast is enabled, packets are labeled with an `rid`, which denotes simulcast
+encoding that a packet belongs to:
+
+```elixir
+{:ex_webrtc, input_pc_pid, {:rtp, input_track_id, rid, packet}}
+```
+
+## Switching between simulcast encodings
+
+Switching between simulcast encodings requires some modifications to RTP packets.
+Every encoding starts with a random RTP sequence number and a random RTP timestamp.
+Because client that receives our stream is never aware of simulcast (they always receive
+a single encoding), we have to rewrite those sequence numbers and timestamps to be continuous and increasing.
+This process is known as munging.
+
+1. Create munger with codec sample rate
+
+```elixir
+alias ExWebRTC.PeerConnection
+alias ExWebRTC.RTP.{H264, Munger}
+
+m = Munger.new(90_000)
+```
+
+2. When a packet from an encoding that we want to forward arrives, rewrite its sequnce number and timestamp:
+
+```elixir
+receive do
+  {:ex_webrtc, input_pc, {:rtp, _input_track_id, "m", packet}} ->
+    {packet, munger} = Munger.munge(munger, packet)
+    :ok = PeerConnection.send_rtp(output_pc, output_track_id, packet)
+  {:ex_webrtc, input_pc, {:rtp, _input_track_id, _rid, packet}} ->
+    # ignore other packets
+end
+```
+
+3. To switch to another encoding, request a keyframe for this encoding.
+Once the keyframe arrives, update the munger and start forwarding new packets.
+For example, transitioning from encoding `m` to `h`:
+
+
+```elixir
+# assume we have the following state
+state = %{
+  current_encoding: "m", 
+  munger: munger, 
+  input_pc: input_pc,
+  input_track_id: input_track_id,
+  output_pc: output_pc, 
+  output_track_id: output_track_id
+}
+
+:ok = PeerConnection.send_pli(state.input_pc, state.input_track_id, "h")
+
+# ...
+
+receive do
+  {:ex_webrtc, input_pc, {:rtp, _input_track_id, rid, packet}} ->
+    cond  do
+      rid == state.current_encoding -> 
+        {munger, packet} = Munger.munge(munger, packet)
+        :ok = PeerConnection.send_rtp(state.output_pc, state.output_track_id, packet)
+        %{state | munger: munger}
+      rid == "h" and H264.keyframe?(packet) ->
+        munger = Munger.update(munger)
+        {munger, packet} = Munger.munge(munger, packet)
+        :ok = PeerConnection.send_rtp(state.output_pc, state.output_track_id, packet)
+        %{state | munger: munger, current_encoding: "h"}
+      true ->
+        state
+    end
+end
+```
+
+See our [Broadcaster](https://github.com/elixir-webrtc/apps/blob/master/broadcaster/lib/broadcaster/forwarder.ex) app source code for more.
diff --git a/mix.exs b/mix.exs
index 7667177..b18d684 100644
--- a/mix.exs
+++ b/mix.exs
@@ -74,7 +74,7 @@ defmodule ExWebRTC.MixProject do
 
   defp docs do
     intro_guides = ["intro", "negotiation", "forwarding", "consuming"]
-    advanced_guides = ["modifying", "mastering_transceivers", "debugging"]
+    advanced_guides = ["simulcast", "modifying", "mastering_transceivers", "debugging"]
 
     [
       main: "readme",