[webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)

tonyherre has just created a new issue for https://github.com/w3c/webrtc-rtptransport:

== Unsignalled streams aren't needed by any existing usecase ==
As discussed in last week's discussion meeting, we've had a way to add unsignalled/negotiated RTCRtpSendStreams/RTCRtpReceiveStreams via the `RTCRtpTransport. addRtpSendStream()` and `RTCRtpTransport. addRtpReceiveStream()` methods for quite a while, but none of the existing usecases actually need them. Everything we have can be conducted by first adding a transceiver and negotiating:

```
let t = pc.addTransceiver('video');
await negotiate(); // Custom stuff, using custom codec negotiation API if required
let sendStream = await t.sender.replaceSendStream();
```

I suggest we delete the add methods until we have a fleshed out usecase that needs such unsignalled streams and works out the corner cases.

Please view or discuss this issue at https://github.com/w3c/webrtc-rtptransport/issues/66 using your GitHub account


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Received on Tuesday, 13 August 2024 11:13:34 UTC