Question about PlainTransportOptions's comedia option

The v3 documentation says, Whether remote IP:port should be auto-detected based on first RTP/RTCP packet received. If enabled, connect() must only be called if SRTP is enabled by providing the remote srtpParameters and nothing else.

Consider 1:n video streaming, when there is 1 producer but many consumers, if server side’s RTP port in udp can be set to listen mode like in tcp, then sdk-based client(not webrtc client) can directly pull H264 stream from this RTP port, without need to ask for a dynamic port or calling plainTransport.connect at server side.

I don’t really follow what you mean.

I mean, as to 1 specific stream producer, multiple consumers can connect to the same RTP port at server-side, they do not need to do any negotiation since they’re custom sdk client, not webrtc client, and then get 0-RTT connect latency…

No, sorry. That’s not gonna happen. Each plain or WebRTC transport requires its own port(s) in server side.

I know, i’m just wondering~ maybe could put a HAProxy to do server-side load balance…

UDP Loadbalancing · Issue #62 · haproxy/haproxy · GitHub HAProxy seems not supporting UDP LB, but i guess it’s possible~

Hi ibc,

when use comedia:true option, how can client send a UDP packet to server’s RTP listen port? Does RTP specify this case? Maybe to combine RTSP + RTP: client first send a RTSP PLAY packet to server’s rtp port, but mediasoup doesn’t support RTSP…