Re: [dispatch] Dispatching WebTransport
Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com> Mon, 17 June 2019 14:52 UTC
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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Date: Mon, 17 Jun 2019 16:52:38 +0200
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Subject: Re: [dispatch] Dispatching WebTransport
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From the draft: WebTransport [OVERVIEW <https://tools.ietf.org/html/draft-vvv-webtransport-quic-00#ref-OVERVIEW>] is a protocol framework that enables clients constrained by the Web security model to communicate with a remote server using a secure multiplexed transport. Although while I agree with your analysis regarding current webrtc dc and sctp status, if this is a pure client-to-server transport, why is webrtc mentioned in the background at all? shouldn't this be a replacement of websockets and NOT of the webrtc datachannels? Best regards Sergio On 17/06/2019 15:35, Victor Vasiliev wrote: > Hello friendly IETF dispatchers, > > I am writing about new work I want to bring to IETF. The proposal is > called WebTransport. It’s a combination of a Web API currently under > development in W3C WICG [0], a protocol framework and some protocols > that fit into that framework. Combined, they would allow web > applications to establish WebSocket-like connections that instead of > ordered reliable messages use multiple streams and datagrams > (datagrams are unreliable and streams do not have head-of-line > blocking). This is highly useful for real-time and other latency > sensitive applications. > > # Background > > Historically, the only networking operations available to the Web > applications were sending HTTP requests and receiving HTTP responses. > That model does not fit all applications well, so over time, more > mechanisms were added. The two most relevant here are WebSockets (RFC > 6455) and RTC Data Channels (draft-ietf-rtcweb-data-channel). > WebSockets are a way for Web applications to do bidirectional > communication over a TCP connection; they work great if TCP fits your > transport needs, but perform poorly if your application is latency > sensitive and would, in non-Web context, use a UDP-based protocol. > There are many different kinds of applications like that, but I would > like to highlight two major categories which I to some extent surveyed > when coming up with this proposal: > > 1. Custom client-server chat/multimedia protocols (faster-than-DASH > video streaming, game streaming, etc). Those are usually developed > by teams with a good amount of resources, and they are interested > in tailoring the setup for their use case. > 2. Game developers. Online games are commonly real-time in nature > and benefit dramatically from ability to give up on transmitting > old information. They usually use some in-house UDP-based > protocol, and often need to run on unusual platforms. > > WebRTC Data Channels are a mechanism that provides a WebSocket-like > interface with unreliable delivery features. On the wire, it’s > SCTP-over-DTLS, established using ICE and SDP. In theory, this > provides users with enough functionality to build anything they need, > since SCTP messages can be unreliable and unordered. In practice, > while RtcDataChannel is fairly straightforward to use for > browser-to-browser peer-to-peer communication, it has seen much lower > adoption than WebSockets in the client-server scenario, even > considering the fact that its use cases is naturally more niche. > > The main reason for this is the incredible complexity of the WebRTC > stack. WebSockets are a fairly straightforward overlay on top of TCP > and TLS; there is a wide variety of implementations out there, and > it's fairly easy to write a new one (I wrote on myself in less than > 1,000 lines of C++). With data channels, however, once there is no > browser to abstract all of the complexity away, the web developers are > required to understand and implement (or at least integrate) SDP, ICE, > STUN, DTLS and userspace SCTP. While a lot of those have > simplifications for this use case (ICE Lite) and some protocols listed > have a variety of implementations widely available (DTLS), the entire > system still requires going through hundreds of pages of RFCs in order > to understand it well enough to implement. This complexity barrier > has precluded Data Channel adoption by communities of smaller > developers who don’t have resources to implement them, notably game > developers (see [1] and [2] for some discussion). > > Even among the people who got past the complexity barrier, the > feedback I heard almost universally is that WebRTC Data Channels are > hard to work with. From the feedback I gathered, the main problem is > usually around the transport protocol itself. Userspace SCTP is > essentially a monoculture: virtually all implementations use > libusrsctp, a 80,000-line adaptation of FreeBSD SCTP implementation. > This lack of tool choice is fairly painful since latency-sensitive > real-time applications often require quite a bit of tuning on the > transport side to get the best performance (custom congestion control, > etc). In addition, the limitations on the message size stemming from > both the API itself and the lack of widespread support for message > interleaving (RFC 8260) means that the developers have to roll their > own framing on top of SCTP messages if they want to avoid > head-of-line-blocking (this is particularly bad because the framing > overhead in data channels is already large as-is). > > In summary, we have a system that technically provides what everyone > wants, but that nobody is happy with, and that is not usable by all > but the most well-resourced users. > > # Proposal > > Our initial idea for fixing this was to take QUIC and do what > WebSocket did to TCP: add security features that would make it safe to > expose on the Web (by adding origin checks, etc), but otherwise expose > it as-is. This would get us out of libusrsctp monoculture (QUIC is > not yet finished, but it already has a fairly diverse implementation > ecosystem, see [3]), and remove all P2P-related complexity involving > SDP and ICE. The original proposal for that was called QuicTransport; > we showed it to various people, and the feedback we got is that (1) > the API should not be tied to a particular transport (since we already > switched once from SCTP to QUIC, tying it to QUIC specificially would > not be wise), and (2) it shouldn’t fail hard when QUIC is unavailable. > > As a result of that feedback, we abstracted it into a general-purpose > framework called WebTransport. The overview draft, > > https://tools.ietf.org/html/draft-vvv-webtransport-overview-00 > > describes the framework itself, mainly the requirements the transport > protocols have to satisfy to be usable on the web through the API. > Within this framework, we propose the following protocols: > > * QuicTransport -- a simple WebSocket-like adaptation of QUIC, > described in > https://tools.ietf.org/html/draft-vvv-webtransport-quic-00 > * Http3Transport -- a mechanism that allows creating custom non-HTTP > streams within an HTTP/3 session, described in > https://tools.ietf.org/html/draft-vvv-webtransport-http3-00. This > is sort of a version of RFC 8441 for QuicTransport. > * FallbackTransport -- a TCP-based transport with multiplexed > streams that can be used when QUIC is not available (e.g. on > network that require CONNECT proxy). We don’t have a draft > specifically for this, and there are at least two approaches we > could take here: either reusing HTTP/2 as a transport (i.e. just > use draft-kinnear-httpbis-http2-transport), or building a protocol > with QUIC-like semantics on top of WebSockest. The earlier is a > more straightforward way; the latter has the advantage of being > fully polyfillable in JavaScript. > > > # Discussion > > At this point, I am fairly certain that there is a problem here that > needs to be addressed. I am formally requesting ART area to take this > problem on. > > I believe the drafts above would be a good starting point for > discussion. The design that they describe went through several > iterations based on the feedback I got when I discussed this work > within a more narrow audience (mostly people in QUIC working group), > so we’re hopefully at least looking in the right direction here. I am > requesting feedback on this proposal, both on the overall plan and the > specifics described in the drafts. I hope to discuss this in depth in > Montreal, both at dispatch and (in more depth) at a side-meeting. > > Thanks, > Victor. > > [0] https://github.com/WICG/web-transport > [1] https://discourse.wicg.io/t/webtransport-proposal/3508/9 > [2] https://news.ycombinator.com/item?id=13266692 > [3] https://github.com/quicwg/base-drafts/wiki/Implementations > > _______________________________________________ > dispatch mailing list > dispatch@ietf.org > https://www.ietf.org/mailman/listinfo/dispatch
- [dispatch] Dispatching WebTransport Victor Vasiliev
- Re: [dispatch] Dispatching WebTransport Paul Kyzivat
- Re: [dispatch] Dispatching WebTransport T H Panton
- Re: [dispatch] Dispatching WebTransport Sergio Garcia Murillo
- Re: [dispatch] Dispatching WebTransport Iñaki Baz Castillo
- Re: [dispatch] Dispatching WebTransport Victor Vasiliev
- Re: [dispatch] Dispatching WebTransport Peter Thatcher
- Re: [dispatch] Dispatching WebTransport Peter Thatcher
- Re: [dispatch] Dispatching WebTransport westhawk
- Re: [dispatch] Dispatching WebTransport westhawk
- Re: [dispatch] Dispatching WebTransport Peter Thatcher