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Comment by ibc | Hacker News Reader

Majority of RTP media server listen into a separate port for each connection. That's how RTP typically works. This is not TCP connections.

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rfc3550 states that it is per destination ip/port tuple. So you should be able to support multiple connections per local port. Is it possible this is an oversight in the current implementation? I appreciate this isn’t TCP, which is why I have just read through all relevant RFCs.

Why is that so important? As I said, choosing a specific port is not enough. This is not TLS. An aggressive firewall may drop those TCP connections because there is no TLS data on them.

TLS port was just a thought, as I want to reduce cases where turn server is used because of a limitation with scalability (65k connection limit per turn server due to a shared source ip). But our discussion has raised another issue regarding mediasoups limitation of one source per local port - which compounds the issue.

I’m replacing a web socket server with a data channel server. If I use mediasoup then I will need to listen over 4 ip4 addresses to support the 200k clients I can currently support on 1 ip address with web sockets. Not a huge deal right now, but if I want to support millions of user it means managing 40 or so ip addresses instead of 1 or 2.

Not knocking mediasoup at all, just now aware of a limitation that sounds like it doesn’t need to exist so seeing if we can do something about it.

This is RTP not WebSocket or HTTP. Media servers need a separate port for each RTP communication. A hack could be done to make all WebRTC endpoints to use a single port in mediasoup side. However mediasoup also support plain RTP endpoints and, in those, you need to be ready to listen for RTP from any remote IP:port (you don't know it in advance due to NATs). In WebRTC we can use ICE user/pwd (previously given to the server via signaling) but that's not possible with plain/regular RTP (no ICE).

Isn’t that what the SSRC is for? I.e you use the SSRC (sent as part of media in the SDP) to identify the stream, rather than trusting an authentic stream is the only one to send to an open port? At least, that is how I understood the (multiple) rfcs. Not an expert here by any means.

In WebRTC spec (although not super mandatory but the current way to go), the client no longer signals its sending SSRCs into the SDP but a MID and optional RID values (if simulcast is in use), and those MID and RID are not supposed to be unique across all participants (not at all, but neither SSRCs are supposed to). Those MID and RID values are signaled in the SDP and then included into RTP packets as header extensions. The remote matches RTP packets based on them and then learns the associated SSRC for a faster lookup for future packets.

Anyway, WebRTC is not just about RTP. In fact, before RTP happens, ICE and DTLS must de done.

Many Thanks. Sorry to be a bother :) Ill have a play around and see what I can get working for my usecase.

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