- From: Wenbo Zhu <wenboz@google.com>
- Date: Thu, 8 Dec 2011 13:04:31 -0800
- To: Charles Pritchard <chuck@jumis.com>
- Cc: WebApps WG <public-webapps@w3.org>, Ian Hickson <ian@hixie.ch>
- Message-ID: <CAD3-0rPLr-bib=8TGMLSm8s3OsQrwj0yaJMKtmFbMOOyUJYy4g@mail.gmail.com>
On Wed, Dec 7, 2011 at 6:04 PM, Charles Pritchard <chuck@jumis.com> wrote: > ** > I think the Web Sockets spec is intended for client to server sessions > like this. > WebSocket protocol isn't yet well-supported by proxies. Besides, all we need here is a throwaway RPC-like request/response, and it's a bit heavy duty to use WebSocket for this use case. > > As for the peer-to-peer communication, I imagine the WebRTC group is where > you'll see more activity on this issue. > WebRTC is mostly peer-to-peer UDP (e.g. RTP), and the client-to-client communication has nothing to do with either HTTP or WebSocket. I don't know that arbitrary binary data is on their agenda -- I hope it is > -- for p2p communication. > > > > On 12/7/11 5:59 PM, Wenbo Zhu wrote: > > One use case that we have which is not currently handled by XMLHttpRequest > is incrementally sending data that takes a long time to generate _from the > client to the server_. For example, if we were to record data from a > microphone, we couldn't upload it in real time to the server with the > current API. > > The MediaStreaming spec also mentioned several use cases which would > require streaming request data via an API: > - Sending the locally-produced streams to remote peers and receiving > streams from remote peers. > - Sending arbitrary data to remote peers. > > > http://www.whatwg.org/specs/web-apps/current-work/multipage/video-conferencing-and-peer-to-peer-communication.html > > - Wenbo > > >
Received on Thursday, 8 December 2011 21:05:07 UTC