WebRtcTransport
- Mediasoup 内部有多种 Transport 实现,其中 WebRtcTransport 的实现是基于 ICE Lite(精简版ICE)。
- WebRtcTransport 主要用来进行 client 端 与 mediasoup server 端 Router 进行通讯。
- ICE Lite: meaning that it does not initiate ICE connections but expects ICE Binding Requests from endpoints.
- FULL ICE:是双方都要进行连通性检查,完成的走一遍流程。
- Lite ICE: 在FULL ICE和Lite ICE互通时,只需要FULL ICE一方进行连通性检查, Lite一方只需回应response消息。这种模式对于部署在公网的设备比较常用。
WebRtcTransportOptions
WebRtcTransport 类型的选项参数
Field | Type | Description | Required | Default |
---|---|---|---|---|
listenIps | Array<TransportListenIp|String> | Listening IP address or addresses in order of preference (first one is the preferred one). | Yes | |
enableUdp | Boolean | Listen in UDP. | No | true |
enableTcp | Boolean | Listen in TCP. | No | false |
preferUdp | Boolean | Listen in UDP. | No | false |
preferTcp | Boolean | Listen in TCP. | No | false |
initialAvailableOutgoingBitrate | Number | Initial available outgoing bitrate (in bps). | No | 600000 |
enableSctp | Boolean Create a SCTP association. | No | false | |
numSctpStreams | NumSctpStreams | SCTP streams number. | No | |
maxSctpMessageSize | Number | Maximum allowed size for |