RTCDtlsTransport - Web APIs 编辑
The RTCDtlsTransport
interface provides access to information about the Datagram Transport Layer Security (DTLS) transport over which a RTCPeerConnection
's RTP and RTCP packets are sent and received by its RTCRtpSender
and RTCRtpReceiver
objects.
A DTLS transport is also used to provide information about SCTP packets transmitted and received by an connection's data channels.
Features of the DTLS transport include the addition of security to the underlying transport; the RTCDtlsTransport
interface can be used to obtain information about the underlying transport and the security added to it by the DTLS layer.
<div id="interfaceDiagram" style="display: inline-block; position: relative; width: 100%; padding-bottom: 11.666666666666666%; vertical-align: middle; overflow: hidden;"><svg style="display: inline-block; position: absolute; top: 0; left: 0;" viewbox="-50 0 600 70" preserveAspectRatio="xMinYMin meet"><a xlink:href="/wiki/en-US/docs/Web/API/RTCDtlsTransport" target="_top"><rect x="1" y="1" width="160" height="50" fill="#F4F7F8" stroke="#D4DDE4" stroke-width="2px" /><text x="81" y="30" font-size="12px" font-family="Consolas,Monaco,Andale Mono,monospace" fill="#4D4E53" text-anchor="middle" alignment-baseline="middle">RTCDtlsTransport</text></a></svg></div>
a:hover text { fill: #0095DD; pointer-events: all;}
Properties
iceTransport
undefined- The read-only
RTCDtlsTransport
propertyiceTransport
contains a reference to the underlyingRTCIceTransport
. state
undefinedWebRTC
Methods
This interface has no methods.
Description
Allocation of DTLS transports
RTCDtlsTransport
objects are created when an app calls either setLocalDescription()
or setRemoteDescription()
. The number of DTLS transports created and how they're used depends on the bundling mode used when creating the RTCPeerConnection
.
Whether bundling is used depends on what the other endpoint is able to negotiate. All browsers support bundling, so when both endpoints are browsers, you can rest assured that bundling will be used.
Some non-browser legacy endpoints, however, may not support bundle. To be able to negotiate with such endpoints (or to exclude them entirely), the RTCConfiguration
property bundlePolicy
may be provided when creating the connection. The bundlePolicy
lets you control how to negotiate with these legacy endpoints. The default policy is "balanced"
, which provides a balance between performance and compatibility.
For example, to create the connection using the highest level of bundling:
const rtcConfig = {
bundlePolicy: "max-bundle"
};
const pc = new RTCPeerConnection(rtcConfig);
Bundling lets you use one RTCDtlsTransport
to carry the data for multiple higher-level transports, such as multiple RTCRtpTransceiver
s.
When not using BUNDLE
When the connection is created without using BUNDLE, each RTP or RTCP component of each RTCRtpTransceiver
has its own RTCDtlsTransport
; that is, every RTCRtpSender
and RTCRtpReceiver
, has its own transport, and all RTCDataChannel
objects share a transport dedicated to SCTP.
When using BUNDLE
When the connection is using BUNDLE, each RTCDtlsTransport
object represents a group of RTCRtpTransceiver
objects. If the connection was created using max-compat
mode, each transport is responsible for handling all of the communications for a given type of media (audio, video, or data channel). Thus, a connection that has any number of audio and video channels will always have exactly one DTLS transport for audio and one for video communications.
Because transports are established early in the negotiation process, it's likely that it won't be known until after they're created whether or not the remote peer supports bundling or not. For this reason, you'll sometimes see separate transports created at first, one for each track, then see them get bundled up once it's known that bundling is possible. If your code accesses RTCRtpSender
s and/or RTCRtpReceiver
s directly, you may encounter situations where they're initially separate, then half or more of them get closed and the senders and receivers updated to refer to the appropriate remaining RTCDtlsTransport
objects.
Data channels
RTCDataChannel
s use SCTP to communicate. All of a peer connection's data channels share a single RTCSctpTransport
, found in the connection's sctp
property.
You can, in turn, identify the RTCDtlsTransport
used to securely encapsulate the data channels' SCTP communications by looking at the RTCSctpTransport
object's transport
property.
Examples
This example presents a function, tallySenders()
, which iterates over an RTCPeerConnection
's RTCRtpSender
s, tallying up how many of them are in various states. The function returns an object containing properties whose values indicate how many of the senders are in each state.
let pc = new RTCPeerConnection({ bundlePolicy: "max-bundle" });
/* ... */
function tallySenders(pc) {
let results = {
transportMissing: 0,
connectionPending: 0,
connected: 0,
closed: 0,
failed: 0,
unknown: 0
};
let senderList = pc.getSenders();
senderList.forEach(sender => {
let transport = sender.transport;
if (!transport) {
results.transportMissing++;
} else {
switch(transport.state) {
case "new":
case "connecting":
results.connectionPending++;
break;
case "connected":
results.connected++;
break;
case "closed":
results.closed++;
break;
case "failed":
results.failed++;
break;
default:
results.unknown++;
break;
}
}
});
return results;
}
Note that in this code, the new
and connecting
states are being treated as a single connectionPending
status in the returned object.
Specifications
Specification | Status | Comment |
---|---|---|
WebRTC 1.0: Real-time Communication Between Browsers The definition of 'RTCDtlsTransport' in that specification. | Candidate Recommendation | Initial definition. |
Browser compatibility
BCD tables only load in the browser
See also
如果你对这篇内容有疑问,欢迎到本站社区发帖提问 参与讨论,获取更多帮助,或者扫码二维码加入 Web 技术交流群。
绑定邮箱获取回复消息
由于您还没有绑定你的真实邮箱,如果其他用户或者作者回复了您的评论,将不能在第一时间通知您!
发布评论