Appendix A. WebRTC 1.0 APIs
This Appendix provides a summary of the W3C WebRTC APIs.
RTCPeerConnection API
An RTCPeerConnection
allows two users to communicate directly, browser to browser.
Configuration
Name | Type | Default | Description |
iceServers | sequence<RTCIceServer> | An array containing URIs of servers available to be used by ICE, such as STUN and TURN servers. | |
iceTransports | RTCIceTransports | “all” | Indicates which candidates the ICE engine is allowed to use. |
requestIdentity | RTCIdentityOption | “ifconfigured” | See the requestIdentity member of the RTCOfferAnswerOptions dictionary. |
Name | Type | Description |
credential | DOMString | If this RTCIceServer object represents a TURN server, then this attribute specifies the credentials to use with that TURN server. |
urls | (DOMString or sequence<DOMString>) | STUN or TURN URI(s) as defined in [STUN-URI] and [TURN-URI] or other URI types. |
username | DOMString | If this RTCIceServer object represents a TURN server, then this attribute specifies the username to use with that TURN server. |
Value | Description |
none | The ICE engine must not send or receive any packets at this point. |
relay | The ICE engine must only use media relay candidates such as candidates passing through a TURN server. This can be used to reduce leakage of IP addresses in certain use cases. |
all | The ICE engine may use any type of candidates when this value is specified. |
Constructor
This is the RTCPeerConnection ...
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