This is a sink element. It is used to stream audio via WebRTC (Web Real-Time Communication). That is an established communication protocol used for bidirectional media transport via web browser and apps.
| ! | Integrations |
|
Many applications that use the WebRTC standard use bespoke implementations of it. Because of this, an adaption of our WebRTC source or sink may be necessary to successfully integrate with some third-party solutions. Please contact support@ferncast.de if you want to check availability. Ferncast has its own WebRTC room hosting and aixtream includes a full integration with SessionLinkPRO services. |
| ! | WebRTC Listen-in Player |
| The inbuilt media player for listening into active PIPEs, supports HTTP streaming by default. Low-latency WebRTC streaming can be enabled by activating the WebRTC license. Reach out via info@ferncast.de if you wish to purchase the WebRTC license module. |
| Element | Description |
|---|---|
| config | |
| dataChannel | Toggles the use of data channel transmissions “On” or “Off”. |
| Method | Configuration of the specific WebRTC method to use. See the table below for the different options for each method. |
| Method dropdown |
Dropdown: Selects the method scheme.
|
| servers | Configures what WebRTC server to use. Only active when “ferncastRoom” or “gstSession” are chosed as Method. |
| websocket | Sets the web socket address of the server. |
| stun | Sets the STUN server address to use. |
| checkCertificates | Toggles whether certificates are checked before communication begins ("On") or not ("Off"). |
| Method | Description |
| ferncastRoom | |
| uid | Sets the Unique Identifer name of the WebRTC session. |
| create | Toggles whether the configured room is created anew ("On") or not ("Off"). |
| gstSession | |
| peerID | Sets the identifier of the remote communication partner. |
| slave | |
| local | |
| webAudio |
Dropdown: Selects the WebAudio source to connect with. Note that this requires existing WebAudio PIPEs to exist. |