Update README.md

README.md

@@ -9,15 +9,145 @@
 Setup Webrtc for video and audio streaming from local media devices
 
 # Build and Install
+# Prerequisites
+
+Install the following packages: **libgstreamer1.0-dev**, **libgstreamer-plugins-base1.0-dev**, **libunwind-dev**.
+
+Download latest go lang from the following [link](https://go.dev/dl/). Specifically for Raspbian OS download: go1.22.5.linux-arm64.tar.gz.
+
+# Build
+
 To build the code execute command `make build`. This will build executable: `bin/gowebrtc`. Executing command `./bin/gowebrtc` will print the command line usage instructions.
 
+# Installation
+
 To build deb installer package (which can be installed on Debian, Ubuntu, Raspbian OS among others) execute the command `make debian`. The package gets built in the directory `build/debian/gowebrtc.deb`.
 
 To install the above package using **sudo** execute the command `make debian-install`.
 
 # Configuring
 
-The default configuration file resides in `/etc/gowebrtc/config.yaml`. Sample configuration file is:
+The default configuration file resides in `/etc/gowebrtc/config.yaml`. 
+
+## General configuration option
+
+| Option | Type | Default | Required | Description |
+| -- | -- | -- | -- | -- |
+| image_width | number | 640 | No | Specifies the video width |
+| image_height | number | 480 | No | Specifies the video height |
+| framerate | number | 30 | No | Specifies the fps value |
+| log_file | string | none | No | Log file location. Needs to be writable |
+| audio_device | string | | Yes | Gstream pipeline to be used for audio stream |
+| video_device | string | | Yes | Gstream pipeline to be used for video stream |
+
+## Signalling configuration
+Either REST HTTP API or websockets can be used for exchanging SDP and Candidates.
+
+| Option | Type | Default | Required | Description |
+| -- | -- | -- | -- | -- |
+| port | number | 8080 | No | Port to be used for running signalling service. |
+| url | string | /stream | No | Url path to be used by signalling service. |
+| signalling | string | websocket | No | The value can be one of http or websocket. |
+| signalling_uses_tls | bool | false | No | If you want to run signalling server in TLS mode. Currently only websocket supports this. |
+| signalling_tls_cert | string | | No | Server certificate |
+| signalling_tls_key | string | | No | Server certificate key |
+
+### Websocket specific configuration options
+
+| Option | Type | Default | Required | Description |
+| -- | -- | -- | -- | -- |
+| signalling_origin | string | | No | If provided allows requests from specific origin. Requests from other origina are denied. |
+| SignallingCredentials | array | | No | Array of credentials. Any of the given credentials must be provided. If no credentials is specified in configuration file, credential are not required. |
+
+#### Specifying credentials
+
+| Option | Type | Default | Required | Description |
+| -- | -- | -- | -- | -- |
+| user | string | | Yes | Credential user name |
+| password | string | | Yes | Credential user password |
+
+## Turn server configuration
+Webrtc requires turn servers to function in some scenarios. Gowebrtc service has internal turn server. Also it supports using Open Relay and other turn servers.
+
+| Option | Type | Default | Required | Description |
+| -- | -- | -- | -- | -- |
+| use_internal_turn | bool | false | No | Use internal turn server of external turn server |
+
+### Internal turn server options
+
+Following is sample configuration portion for internal turn server configuration:
+
+```yaml
+use_internal_turn: true
+turn_configuration:
+  public_ip: <public-ip>
+  port: <turn-port>
+  users:
+    - user: <turn-user>
+      password: <turn-password>
+```
+
+### Open Relay turn server options
+
+If **open_relay_config** attribute is defined, Open relay will be used as turn server.
+
+```yaml
+open_relay_config:
+  app_name: <app-name>
+  api_key: <api-key>
+```
+
+### Other turn servers
+
+Using **ice_servers** any other ice server(s) can be used.
+
+```yaml
+ice_servers:
+  - urls:
+    - "stun:<server>"
+  - urls:
+    - "turn:<server>:80"
+    username: <username>
+    credential: <password>
+  - urls:
+    - "turn:<server>:80?transport=tcp"
+    username: <username>
+    credential: <password>
+  - urls:
+    - "turn:<server>:443"
+    username: <username>
+    credential: <password>
+  - urls:
+    - "turns:<server>:443?transport=tcp"
+    username: <username>
+    credential: <password>
+```
+
+## Samples
+### Running the service with internal turn server
+If you have public and IP and would like to host your own turn server, the following configuration can be used:
+
+```yaml
+port: 8080
+url: /ws
+image_width: 640
+image_height: 480
+framerate: 30
+log_file: /var/log/gowebrtc.log
+audio_device: alsasrc device=plughw:CARD=I930,DEV=0
+video_device: autovideosrc
+use_internal_turn: true
+signalling: websocket
+turn_configuration:
+  public_ip: <public-ip>
+  port: <turn-port>
+  users:
+    - user: <turn-user>
+      password: <turn-password>
+```
+
+### Runnig the service with external turn server
+External turn server configuration is specified by providing relevant information in the config file. Relevant section is as follows:
 
 ```yaml
 port: 8080
@@ -28,6 +158,7 @@ framerate: 30
 log_file: /var/log/gowebrtc.log
 audio_device: alsasrc device=plughw:CARD=I930,DEV=0
 video_device: autovideosrc
+signalling: http
 open_relay_config:
   app_name: <app-name>
   api_key: <api-key>
@@ -50,8 +181,8 @@ ice_servers:
     - "turns:global.relay.metered.ca:443?transport=tcp"
     username: <username>
     credential: <password>
-
 ```
+
 # Running and Enabling Service
 `sudo systemctl start gowebrtc` will start the service.
 `sudo systemctl enable gowebrtc` will configure service startup on each reboot.