Incoming payloads are silently truncated to 8kB

[jech]

If incoming payloads are larger than 8kB, they are silently truncated, which causes loss of data. I believe that there are two issues here:

• there is no error indication when a packet is truncated;
• 8kB is too small in any cases (the protocol that I'm trying to implement uses slightly over 16kB).

A simple workaround is to increase the value of receiveMTU in rtcpeerconnection.go¸ but that's obviously not satisfactory.

[backkem]

This is a known problem with the WebRTC API itself. It doesn't provide a good way to handle large messages. All implementations have some limitation. We can increase the default payload size but it would still be an arbitrary limit. A possible solution would be to make this configurable.
Alternatively, You can check out the detach example. This shows off an alternative way to interact with data channels. This API allows you to pass your own buffers for reading and writing, allowing you to determine their size. Note that we've probably done limited testing with large buffers so you may uncover some issues.

[jech]

This is a known problem with the WebRTC API itself. It doesn't provide a good way to handle large messages

I still think there should be some sort of indication given to the application that the message has been truncated.

[jech]

This is a known problem with the WebRTC API itself.

I'm confused — your link speaks of a protocol problem with SCTP, not an API problem. Couldn't the acceptSRTCP function reallocate the buffer when it is found to overflow, and then continue reading? Alternatively, couldn't the OnMessage callback return an io.Reader, as is done in gorilla.websocket?

[jech]

I'm glad to report that Detach solved my problem quite nicely. Thanks.

[backkem]

I agree we should look into adding a warning. We'll keep the issue open untill then.

[backkem]

FYI, I did indeed link the wrong info above. For truly sending/receiving any size messages via the WebRTC API, it would have to be extended as suggested in w3c/webrtc-pc#1732.