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Re: What would you like to see in WebRTC next? A low-level API?

From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
Date: Wed, 21 Feb 2018 17:10:56 +0100
To: public-webrtc@w3.org
Message-ID: <13c72577-3b91-fa3b-4fb3-99eeb97e88fb@gmail.com>
On 21/02/2018 7:37, Randell Jesup wrote:
> On 1/24/2018 12:59 PM, Sergio Garcia Murillo wrote:
>> On 24/01/2018 18:50, Michael Tuexen wrote:
>>> Are you referring to a non-congestion controlled way of sending 
>>> unordered, unreliable datagrams?
>>> You can right now use and unordered delivery, limit the number of 
>>> retransmissions to 0 and
>>> get a congestion controlled unordered unreliable datagram service...
>> Except that it can be buffered, chunked, there is no way of knowing 
>> the MTU and last time I checked didn't work as expected on any 
>> browser.. ;) 
> There is a transmit buffer, and it does hide the MTU from you, true - 
> it provides an abstracted UDP API, not a raw socket API. If you could 
> ask the current apparent MTU as seen by the protocol, you could 
> restrict your messages to that.   But of course that same issue exists 
> for UDP - you can send larger messages (ok, max 64K) and it will 
> fragment and reassemble (maybe) the messages for delivery, just like 
> DataChannels.
> So that leaves exposing the (current) MTU, and the fact it can be 
> buffered (though normal network IO can be buffered as well). Would you 
> be happy with exposing MTU?

Yes, I think that exposing the MTU, or current "maxChunkSize" would be 
enough for me so far.

Best regards
Received on Wednesday, 21 February 2018 16:11:20 UTC

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