Add deploying tutorial (#159)

guides/advanced/debugging.md

@@ -161,3 +161,5 @@ You can also configure PeerConnection to use a specific port range by doing
 ```
 
 Otherwise, the connection won't be established at all, or just in one direction.
+
+Read more in our [Deploying tutorial](./deploying.md#allow-udp-traffic-in-your-firewall)!

guides/advanced/deploying.md

@@ -0,0 +1,91 @@
+# Deploying
+
+Deploying WebRTC applications can be cumbersome.
+Here are a few details you should keep in mind when trying to push your project into production.
+
+## Allow UDP traffic in your firewall
+
+In most cases, WebRTC uses UDP to exchange audio and video data.
+Therefore, you have to allow UDP traffic in your firewall.
+In Linux-based systems that use `ufw`, you can do this with the following command:
+
+```sh
+ufw allow 50000:60000/udp
+```
+
+Our ICE implementation, by default, uses an ephemeral port range, so it might vary depending on your operating system.
+However, you can specify an exact port range that ICE will use when creating a new peer connection, e.g.:
+
+```elixir
+PeerConnection.start_link(ice_port_range: 50_000..60_000)
+```
+
+## Allow TCP traffic in your firewall
+
+In some cases, when ICE really cannot find a UDP path, it may fall back to a TCP connection.
+However, since our ICE implementation does not support TCP yet, you don't need to take any extra steps here :)
+
+## Export ports in your Docker container
+
+If you are running your application using Docker, we recommend using the `--network host` option.
+If that's not possible (e.g. you are running on macOS), you have to manually export the ports used by ICE, e.g.:
+
+```
+docker run -p 50000-50010/udp myapp
+```
+
+Keep in mind that exporting a lot of ports might take a lot of time or even cause the Docker daemon to timeout.
+That's why we recommend using host's network.
+
+## Choose your cloud provider wisely
+
+Many cloud providers do not offer good support for UDP traffic.
+In such cases, deploying a WebRTC-based application might be impossible.
+We recommend using bare machines that you can configure as you need.
+
+## Enable HTTPS in your frontend
+
+The server hosting your frontend site must have HTTPS enabled.
+This is a requirement for accessing the user's microphone and camera devices.
+Not using HTTPS on addresses different than localhost will result in `navigator.mediaDevices` being `null`.
+
+## Proxy WebSocket connections
+
+WebSockets are a common option for the signalling channel.
+If you are using a reverse-proxy like nginx, to make your WebSocket connections work,
+you have to preserve the original (client) request headers.
+In other words, you need to add the following lines to your endpoint handling websocket connections configuration:
+
+```
+proxy_http_version 1.1;
+proxy_set_header Upgrade $http_upgrade;
+proxy_set_header Connection "upgrade";
+```
+
+Read more [here](https://nginx.org/en/docs/http/websocket.html).
+
+## Configure STUN servers
+
+If you are deploying your application behind a NAT, you have to configure a STUN 
+server that will allow it to discover its public IP address.
+In Elixir WebRTC this will be:
+
+```elixir
+PeerConnection.start_link(ice_servers: [%{urls: "stun:stun.l.google.com:19302"}])
+```
+
+Google's STUN server is publicaly available, but keep in mind that you depend on
+someone else's infrastructure.
+If it goes down, you can do nothing about it.
+To avoid that, you would need to host your own STUN server.
+Keep in mind, that TURN servers are also STUN servers so if you have already TURN deployed,
+you don't need to specify additional STUN servers.
+And as a TURN server, you can always use our [Rel](https://github.com/elixir-webrtc/rel)!
+
+## Configure TURN servers
+
+If your application is deployed behind a very restrictive NAT, which should be very rare (e.g. a symmetric NAT),
+you will need to configure a TURN server.
+In most cases, TURN servers are needed on the client side as you don't have any control 
+over a network your clients connect from.
+For testing and experimental purposes, you can use our publicly available TURN called [Rel](https://github.com/elixir-webrtc/rel)!

mix.exs

@@ -74,7 +74,14 @@ defmodule ExWebRTC.MixProject do
 
   defp docs do
     intro_guides = ["intro", "negotiation", "forwarding", "consuming"]
-    advanced_guides = ["simulcast", "modifying", "mastering_transceivers", "debugging"]
+
+    advanced_guides = [
+      "simulcast",
+      "modifying",
+      "mastering_transceivers",
+      "deploying",
+      "debugging"
+    ]
 
     [
       main: "readme",