gowebrtc

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. 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.

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:

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.

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.

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:

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:

port: 8080
url: /stream
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
signalling: http
open_relay_config:
  app_name: <app-name>
  api_key: <api-key>
ice_servers:
  - urls:
    - "stun:stun.relay.metered.ca:80"
  - urls:
    - "turn:global.relay.metered.ca:80"
    username: <username>
    credential: <password>
  - urls:
    - "turn:global.relay.metered.ca:80?transport=tcp"
    username: <username>
    credential: <password>
  - urls:
    - "turn:global.relay.metered.ca:443"
    username: <username>
    credential: <password>
  - urls:
    - "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.

APIS

URL	Method	Payload	Description	Response	Error Response
/stream	POST	{"sdp": "localSessionDescription"}	This API call initiates SDP exchange more generally known as Signalling. In response the API returns the Remote SDP. In case of error error message is returned with status code as 500.	{"sdp": "remoteSessionDescription"}	{"error": "error message"}
/stream	DELETE	none	This API call terminates any existing streaming session.	none	{"error": "error message"}

Remember at any given time only one webrtc streaming is possible. An attempt to initiate another streaming will result in an error.