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References: <CAAZdMadKgUkAwnYQ7MmSR46qQZTh8+FF5BuKmc1r33SMyF91sQ@mail.gmail.com> <98EF12E4-4BF2-4FBC-852B-776A0E0118BE@westhawk.co.uk>
In-Reply-To: <98EF12E4-4BF2-4FBC-852B-776A0E0118BE@westhawk.co.uk>
From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 1 Jul 2019 17:10:27 -0700
Message-ID: <CAJrXDUGubsbRV7_FinKr_r1m+MkBBXSVw=O-3oH31+UOJGjNtw@mail.gmail.com>
To: T H Panton <thp@westhawk.co.uk>
Cc: Victor Vasiliev <vasilvv=40google.com@dmarc.ietf.org>, dispatch@ietf.org
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Subject: Re: [dispatch] Dispatching WebTransport
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On Mon, Jun 17, 2019 at 7:13 AM T H Panton <thp@westhawk.co.uk> wrote:

>
>
> On 17 Jun 2019, at 14:35, Victor Vasiliev <
> vasilvv=3D40google.com@dmarc.ietf.org> wrote:
>
> Hello friendly IETF dispatchers,
>
> I am writing about new work I want to bring to IETF.  The proposal is
> called WebTransport.  It=E2=80=99s a combination of a Web API currently u=
nder
> 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.
>
>
> Establish connections to where ? Between eachother P2P ? Or to any server=
?
> Or just to the originating server only?
>

Any server or p2p.


>
>
> # Background
>
> Historically, the only networking operations available to the Web
> applications were sending HTTP requests and receiving HTTP responses.  Th=
at
> 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 tail=
oring
>    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 o=
ften
>    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=E2=80=99s
> 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 WebSocke=
ts
> in the client-server scenario, even considering the fact that its use cas=
es
> 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 line=
s
> of C++).  With data channels, however, once there is no browser to abstra=
ct
> 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.  Thi=
s
> complexity barrier has precluded Data Channel adoption by communities of
> smaller developers who don=E2=80=99t have resources to implement them, no=
tably game
> developers (see [1] and [2] for some discussion).
>
>
> Much, but not all, of the complexity of the webRTC stack is due to the NA=
T
> traversal and security commitments you need to do P2P.
>
If you exclude the need to do NAT traversal and P2P
> then it _could_ be much simpler. But it also gives you E2E encryption and
> low latency as side effects!
>
>
 In client/server mode, WebTransport doesn't require any of that, so it is
a lot more simple.

Work is in progress decoupling the APIs from the (dreaded) SDP and
> simplifying the developer experience.
>
>
What work?


> That said, it is by no means un-managable - I've implemented pretty much
> the whole stack from scratch for small devices and I know of 2 other
> implementations which don't use libusrsctp
>
>
Maybe not for you, but we regularly get emails from people saying things
like "It would be great to delete the mess of the WebRTC stack if there's a
simpler protocol available. ICE/DTLS/SCTP all scare me".


>
>
> 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 t=
he
> 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 congesti=
on
> 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-blocki=
ng
> (this is particularly bad because the framing overhead in data channels i=
s
> already large as-is).
>
>
> SCTP is well defined nice protocol that suffers (IMHO) from a lack of
> 'coolness' due to being used by the telcos for most of it's life.
> The head-of-line issue is a problem, but entirely fixable with small
> tweaks to SCTP protocol. (read it wasn't in the original requirements - s=
o
> it doesn't do it).
>

It's been fixable for a very long time.  I have my doubts about when it
will actually get fixed.


>
>
> 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 t=
he
> most well-resourced users.
>
>
> I'd point out that 2.5bn endpoints support it right now - including
> _every_ smartphone shipped in the last 2 years. Last I heard DataChannel
> traffic was 0.1% of chrome's traffic
> which is _astonishingly_ high
>

>
> # 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 th=
e
> 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 peopl=
e,
> 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=E2=
=80=99t
> fail hard when QUIC is unavailable.
>
>
> It would be _great_ if that abstraction could _also_ work over SCTP data
> channels. That way you could cover the P2P use cases too.
>

QUIC can be p2p.  In fact, we already have it implemented in Chrome (in an
original trial).

The WebTransport API could run on top of SCTP, and you could probably make
a JS polyfill for it.  But I'm not sure why anyone would bother.


> 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.  Withi=
n
> 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-0=
0
>    - 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 requ=
ire
>    CONNECT proxy).  We don=E2=80=99t 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.  Th=
e
>    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 need=
s
> to be addressed.  I am formally requesting ART area to take this problem =
on.
>
>
> I'd really like to be clear what that problem is.
>
>
Web developers routinely ask for an unreliable websocket, or something
close to UDP (but it needs to be secure).   It turns out that ICE/DTLS/SCTP
is too complex to fill that roll.

I believe the drafts above would be a good starting point for discussion.
> The design that they describe went through several iterations based on th=
e
> feedback I got when I discussed this work within a more narrow audience
> (mostly people in QUIC working group), so we=E2=80=99re hopefully at leas=
t 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 ho=
pe
> to discuss this in depth in Montreal, both at dispatch and (in more depth=
)
> at a side-meeting.
>
>
> Tim.
>
>
> 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=3D13266692
> [3] https://github.com/quicwg/base-drafts/wiki/Implementations
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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<div dir=3D"ltr"><div dir=3D"ltr"><br></div><br><div class=3D"gmail_quote">=
<div dir=3D"ltr" class=3D"gmail_attr">On Mon, Jun 17, 2019 at 7:13 AM T H P=
anton &lt;<a href=3D"mailto:thp@westhawk.co.uk" target=3D"_blank">thp@westh=
awk.co.uk</a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex"><div><br><div><br><blockquote type=3D"cite"><div>On 17 Jun 2019,=
 at 14:35, Victor Vasiliev &lt;<a href=3D"mailto:vasilvv=3D40google.com@dma=
rc.ietf.org" target=3D"_blank">vasilvv=3D40google.com@dmarc.ietf.org</a>&gt=
; wrote:</div><br class=3D"m_-8857436336868946234gmail-m_690526826935632728=
Apple-interchange-newline"><div><div dir=3D"ltr">Hello friendly IETF dispat=
chers,<br><br>I am writing about new work I want to bring to IETF.=C2=A0 Th=
e proposal is called WebTransport.=C2=A0 It=E2=80=99s a combination of a We=
b API currently under development in W3C WICG [0], a protocol framework and=
 some protocols that fit into that framework.=C2=A0 Combined, they would al=
low web applications to establish WebSocket-like connections that instead o=
f ordered reliable messages use multiple streams and datagrams (datagrams a=
re unreliable and streams do not have head-of-line blocking).=C2=A0 This is=
 highly useful for real-time and other latency sensitive applications.<br><=
/div></div></blockquote><div><br></div><div>Establish connections to where =
? Between eachother P2P ? Or to any server? Or just to the originating serv=
er only?=C2=A0</div></div></div></blockquote><div><br></div><div>Any server=
 or p2p.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left=
:1ex"><div><div><br><blockquote type=3D"cite"><div><div dir=3D"ltr"><br># B=
ackground<br><br>Historically, the only networking operations available to =
the Web applications were sending HTTP requests and receiving HTTP response=
s.=C2=A0 That model does not fit all applications well, so over time, more =
mechanisms were added.=C2=A0 The two most relevant here are WebSockets (RFC=
 6455) and RTC Data Channels (draft-ietf-rtcweb-data-channel).=C2=A0 WebSoc=
kets are a way for Web applications to do bidirectional communication over =
a TCP connection; they work great if TCP fits your transport needs, but per=
form poorly if your application is latency sensitive and would, in non-Web =
context, use a UDP-based protocol.=C2=A0 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:<br><ol><=
li>Custom client-server chat/multimedia protocols (faster-than-DASH video s=
treaming, game streaming, etc).=C2=A0 Those are usually developed by teams =
with a good amount of resources, and they are interested in tailoring the s=
etup for their use case.</li><li>Game developers.=C2=A0 Online games are co=
mmonly real-time in nature and benefit dramatically from ability to give up=
 on transmitting old information.=C2=A0 They usually use some in-house UDP-=
based protocol, and often need to run on unusual platforms.=C2=A0=C2=A0</li=
></ol>WebRTC Data Channels are a mechanism that provides a WebSocket-like i=
nterface with unreliable delivery features.=C2=A0 On the wire, it=E2=80=99s=
 SCTP-over-DTLS, established using ICE and SDP.=C2=A0 In theory, this provi=
des users with enough functionality to build anything they need, since SCTP=
 messages can be unreliable and unordered.=C2=A0 In practice, while RtcData=
Channel is fairly straightforward to use for browser-to-browser peer-to-pee=
r communication, it has seen much lower adoption than WebSockets in the cli=
ent-server scenario, even considering the fact that its use cases is natura=
lly more niche.<br><br>The main reason for this is the incredible complexit=
y of the WebRTC stack.=C2=A0 WebSockets are a fairly straightforward overla=
y on top of TCP and TLS; there is a wide variety of implementations out the=
re, and it&#39;s fairly=C2=A0easy to write a new one (I wrote on myself in =
less than 1,000 lines of C++).=C2=A0 With data channels, however, once ther=
e 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.=C2=A0 While a lot of those have simplificati=
ons for this use case (ICE Lite) and some protocols listed have a variety o=
f implementations widely available (DTLS), the entire system still requires=
 going through hundreds of pages of RFCs in order to understand it well eno=
ugh to implement.=C2=A0 This complexity barrier has precluded Data Channel =
adoption by communities of smaller developers who don=E2=80=99t have resour=
ces to implement them, notably game developers (see [1] and [2] for some di=
scussion).<br></div></div></blockquote><div><br></div><div>Much, but not al=
l, of the complexity of the webRTC stack is due to the NAT traversal and se=
curity commitments you need to do P2P. </div></div></div></blockquote><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:=
1px solid rgb(204,204,204);padding-left:1ex"><div><div><div>If you exclude =
the need to do NAT traversal and P2P</div><div>then it _could_ be much simp=
ler. But it also gives you E2E encryption and low latency as side effects!<=
/div><div><br></div></div></div></blockquote><div>=C2=A0</div><div>=C2=A0In=
 client/server mode, WebTransport doesn&#39;t require any of that, so it is=
 a lot more simple.=C2=A0<br></div><div><br></div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,20=
4,204);padding-left:1ex"><div><div><div></div><div>Work is in progress deco=
upling the APIs from the (dreaded) SDP and simplifying the developer experi=
ence.</div><div><br></div></div></div></blockquote><div><br></div><div>What=
 work?=C2=A0 =C2=A0</div><div>=C2=A0</div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);p=
adding-left:1ex"><div><div><div></div><div>That said, it is by no means un-=
managable - I&#39;ve implemented pretty much the whole stack from scratch f=
or small devices and I know of 2 other implementations which don&#39;t use =
libusrsctp=C2=A0</div><div><br></div></div></div></blockquote><div><br></di=
v><div>Maybe not for you, but we regularly get emails from people saying th=
ings like &quot;It would be great to delete the mess of the WebRTC stack if=
 there&#39;s a simpler protocol available. ICE/DTLS/SCTP all scare me&quot;=
.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><=
div><div><div></div><br><blockquote type=3D"cite"><div><div dir=3D"ltr"><br=
>Even among the people who got past the complexity barrier, the feedback I =
heard almost universally=C2=A0is that WebRTC Data Channels are hard to work=
 with.=C2=A0 From the feedback I gathered, the main problem is usually arou=
nd the transport protocol itself.=C2=A0 Userspace SCTP is essentially a mon=
oculture: virtually all implementations use libusrsctp, a 80,000-line adapt=
ation of FreeBSD SCTP implementation.=C2=A0 This lack of tool choice is fai=
rly painful since latency-sensitive real-time applications often require qu=
ite a bit of tuning on the transport side to get the best performance (cust=
om congestion control, etc).=C2=A0 In addition, the limitations on the mess=
age size stemming from both the API itself and the lack of widespread suppo=
rt for message interleaving (RFC 8260) means that the developers have to ro=
ll 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 dat=
a channels is already large as-is).<br></div></div></blockquote><div><br></=
div><div>SCTP is well defined nice protocol that suffers (IMHO) from a lack=
 of &#39;coolness&#39; due to being used by the telcos for most of it&#39;s=
 life.</div><div>The head-of-line issue is a problem, but entirely fixable =
with small tweaks to SCTP protocol. (read it wasn&#39;t in the original req=
uirements - so it doesn&#39;t do it).</div></div></div></blockquote><div><b=
r></div><div>It&#39;s been fixable for a very long time.=C2=A0 I have my do=
ubts about when it will actually get fixed.</div><div>=C2=A0</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px =
solid rgb(204,204,204);padding-left:1ex"><div><div><br><blockquote type=3D"=
cite"><div><div dir=3D"ltr"><br>In summary, we have a system that technical=
ly provides what everyone wants, but that nobody is happy with, and that is=
 not usable by all but the most well-resourced users.<br></div></div></bloc=
kquote><div><br></div><div>I&#39;d point out that 2.5bn endpoints support i=
t right now - including _every_ smartphone shipped in the last 2 years. Las=
t I heard DataChannel traffic was 0.1% of chrome&#39;s traffic=C2=A0</div><=
div>which is _astonishingly_ high</div></div></div></blockquote><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px so=
lid rgb(204,204,204);padding-left:1ex"><div><div><br><blockquote type=3D"ci=
te"><div><div dir=3D"ltr"><br># Proposal<br><br>Our initial idea for fixing=
 this was to take QUIC and do what WebSocket did to TCP: add security featu=
res that would make it safe to expose on the Web (by adding origin checks, =
etc), but otherwise expose it as-is.=C2=A0 This would get us out of libusrs=
ctp monoculture (QUIC is not yet finished, but=C2=A0 it=C2=A0already has a =
fairly diverse implementation ecosystem, see [3]), and remove all P2P-relat=
ed complexity involving SDP and ICE.=C2=A0 The original proposal for that w=
as called QuicTransport; we showed it to various people, and the feedback w=
e got is that (1) the API should not be tied to a particular transport (sin=
ce we already switched once from SCTP to QUIC, tying it to QUIC specificial=
ly would not be wise), and (2) it shouldn=E2=80=99t fail hard when QUIC is =
unavailable.<br><br></div></div></blockquote><div><br></div><div>It would b=
e _great_ if that abstraction could _also_ work over SCTP data channels. Th=
at way you could cover the P2P use cases too.</div></div></div></blockquote=
><div><br></div><div>QUIC can be p2p.=C2=A0 In fact, we already have it imp=
lemented in Chrome (in an original trial).</div><div><br></div><div>The Web=
Transport API could run on top of SCTP, and you could probably make a JS po=
lyfill for it.=C2=A0 But I&#39;m not sure why anyone would bother.</div><di=
v><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><div><di=
v><br></div><blockquote type=3D"cite"><div><div dir=3D"ltr">As a result of =
that feedback, we abstracted it into a general-purpose framework called Web=
Transport.=C2=A0 The overview draft,<br><br>=C2=A0 <a href=3D"https://tools=
.ietf.org/html/draft-vvv-webtransport-overview-00" target=3D"_blank">https:=
//tools.ietf.org/html/draft-vvv-webtransport-overview-00</a><br><br>describ=
es the framework itself, mainly the requirements the transport protocols ha=
ve to satisfy to be usable on the web through the API.=C2=A0 Within this fr=
amework, we propose the following protocols:<br><ul><li>QuicTransport -- a =
simple WebSocket-like adaptation of QUIC, described in <a href=3D"https://t=
ools.ietf.org/html/draft-vvv-webtransport-quic-00" target=3D"_blank">https:=
//tools.ietf.org/html/draft-vvv-webtransport-quic-00</a></li><li>Http3Trans=
port -- a mechanism that allows creating custom non-HTTP streams within an =
HTTP/3 session, described in <a href=3D"https://tools.ietf.org/html/draft-v=
vv-webtransport-http3-00" target=3D"_blank">https://tools.ietf.org/html/dra=
ft-vvv-webtransport-http3-00</a>..=C2=A0 This is sort of a version of RFC 8=
441 for QuicTransport.</li><li>FallbackTransport -- a TCP-based transport w=
ith multiplexed streams that can be used when QUIC is not available (e.g. o=
n network that require CONNECT proxy).=C2=A0 We don=E2=80=99t 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-htt=
pbis-http2-transport), or building a protocol with QUIC-like semantics on t=
op of WebSockest.=C2=A0 The earlier is a more straightforward way; the latt=
er has the advantage of being fully polyfillable in JavaScript.</li></ul><b=
r># Discussion<br><br>At this point, I am fairly certain that there is a pr=
oblem here that needs to be addressed.=C2=A0 I am formally requesting ART a=
rea to take this problem on.<br></div></div></blockquote><div><br></div><di=
v>I&#39;d really like to be clear what that problem is.=C2=A0</div><br></di=
v></div></blockquote><div><br></div><div>Web developers routinely ask for a=
n unreliable websocket, or something close to UDP (but it needs to be secur=
e).=C2=A0 =C2=A0It turns out that ICE/DTLS/SCTP is too complex to fill that=
 roll.=C2=A0=C2=A0</div><div><br></div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padd=
ing-left:1ex"><div><div><blockquote type=3D"cite"><div><div dir=3D"ltr">I b=
elieve the drafts above would be a good starting point for discussion.=C2=
=A0 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=E2=80=99re hopefully at least=
 looking in the right direction here.=C2=A0 I am requesting feedback on thi=
s proposal, both on the overall plan and the specifics described in the dra=
fts.=C2=A0 I hope to discuss this in depth in Montreal, both at dispatch an=
d (in more depth) at a side-meeting.<br></div></div></blockquote><div><br><=
/div><div>Tim.</div><br><blockquote type=3D"cite"><div><div dir=3D"ltr"><br=
>Thanks,<br>=C2=A0 Victor.<br><br>[0] <a href=3D"https://github.com/WICG/we=
b-transport" target=3D"_blank">https://github.com/WICG/web-transport</a><di=
v>[1] <a href=3D"https://discourse.wicg.io/t/webtransport-proposal/3508/9" =
target=3D"_blank">https://discourse.wicg.io/t/webtransport-proposal/3508/9<=
/a><br>[2] <a href=3D"https://news.ycombinator.com/item?id=3D13266692" targ=
et=3D"_blank">https://news.ycombinator.com/item?id=3D13266692</a><br>[3] <a=
 href=3D"https://github.com/quicwg/base-drafts/wiki/Implementations" target=
=3D"_blank">https://github.com/quicwg/base-drafts/wiki/Implementations</a><=
br></div></div>
_______________________________________________<br>dispatch mailing list<br=
><a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</=
a><br><a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br></div></block=
quote></div><br></div>_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"noreferre=
r" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div></div>

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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 1 Jul 2019 17:13:28 -0700
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Subject: Re: [dispatch] Dispatching WebTransport
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On Mon, Jun 17, 2019 at 7:52 AM Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

> From the draft:
>
>    WebTransport [OVERVIEW <https://tools.ietf.org/html/draft-vvv-webtrans=
port-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?
>

It's both a better WebSocket and a better RTCDataChannel.  Take your pick
of how you look at it.


> 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=E2=80=99s a combination of a Web API currently u=
nder
> 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.  Th=
at
> 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 tail=
oring
>    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 o=
ften
>    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=E2=80=99s
> 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 WebSocke=
ts
> in the client-server scenario, even considering the fact that its use cas=
es
> 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 line=
s
> of C++).  With data channels, however, once there is no browser to abstra=
ct
> 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.  Thi=
s
> complexity barrier has precluded Data Channel adoption by communities of
> smaller developers who don=E2=80=99t have resources to implement them, no=
tably 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 t=
he
> 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 congesti=
on
> 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-blocki=
ng
> (this is particularly bad because the framing overhead in data channels i=
s
> 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 t=
he
> 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 th=
e
> 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 peopl=
e,
> 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=E2=
=80=99t
> 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.  Withi=
n
> 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-0=
0
>    - 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 requ=
ire
>    CONNECT proxy).  We don=E2=80=99t 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.  Th=
e
>    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 need=
s
> 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 th=
e
> feedback I got when I discussed this work within a more narrow audience
> (mostly people in QUIC working group), so we=E2=80=99re hopefully at leas=
t 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 ho=
pe
> 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=3D13266692
> [3] https://github.com/quicwg/base-drafts/wiki/Implementations
>
> _______________________________________________
> dispatch mailing listdispatch@ietf.orghttps://www.ietf.org/mailman/listin=
fo/dispatch
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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<div dir=3D"ltr"><div dir=3D"ltr"><br></div><br><div class=3D"gmail_quote">=
<div dir=3D"ltr" class=3D"gmail_attr">On Mon, Jun 17, 2019 at 7:52 AM Sergi=
o Garcia Murillo &lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com">ser=
gio.garcia.murillo@gmail.com</a>&gt; wrote:<br></div><blockquote class=3D"g=
mail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204=
,204,204);padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF">
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix">From the dra=
ft:</div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix"><br>
    </div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix">
      <pre style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;co=
lor:rgb(0,0,0);font-style:normal;font-variant-ligatures:normal;font-variant=
-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-in=
dent:0px;text-transform:none;word-spacing:0px;text-decoration-style:initial=
;text-decoration-color:initial">   WebTransport [<a href=3D"https://tools.i=
etf.org/html/draft-vvv-webtransport-quic-00#ref-OVERVIEW" title=3D"&quot;Th=
e WebTransport Protocol Framework&quot;" target=3D"_blank">OVERVIEW</a>] is=
 a protocol framework that enables clients
   constrained by the Web security model to communicate with a remote
   server using a secure multiplexed transport.</pre>
    </div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix"><br>
    </div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix">Although whi=
le 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&#39;t this be a replacement of websockets
      and NOT of the webrtc datachannels? <br></div></div></blockquote><div=
><br></div><div>It&#39;s both a better WebSocket and a better RTCDataChanne=
l.=C2=A0 Take your pick of how you look at it.</div><div><br></div><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px=
 solid rgb(204,204,204);padding-left:1ex"><div bgcolor=3D"#FFFFFF"><div cla=
ss=3D"gmail-m_-4560266703682762057moz-cite-prefix"><br>
    </div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix">Best regards=
</div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix">Sergio<br>
    </div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix"><br>
    </div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix"><br>
    </div>
    <div class=3D"gmail-m_-4560266703682762057moz-cite-prefix">On 17/06/201=
9 15:35, Victor Vasiliev
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <div dir=3D"ltr">Hello friendly IETF dispatchers,<br>
        <br>
        I am writing about new work I want to bring to IETF.=C2=A0 The
        proposal is called WebTransport.=C2=A0 It=E2=80=99s a combination o=
f a Web
        API currently under development in W3C WICG [0], a protocol
        framework and some protocols that fit into that framework.=C2=A0
        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).=C2=A0 Th=
is
        is highly useful for real-time and other latency sensitive
        applications.<br>
        <br>
        # Background<br>
        <br>
        Historically, the only networking operations available to the
        Web applications were sending HTTP requests and receiving HTTP
        responses.=C2=A0 That model does not fit all applications well, so
        over time, more mechanisms were added.=C2=A0 The two most relevant
        here are WebSockets (RFC 6455) and RTC Data Channels
        (draft-ietf-rtcweb-data-channel).=C2=A0 WebSockets are a way for We=
b
        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.=C2=A0 There ar=
e
        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:<br>
        <ol>
          <li>Custom client-server chat/multimedia protocols
            (faster-than-DASH video streaming, game streaming, etc).=C2=A0
            Those are usually developed by teams with a good amount of
            resources, and they are interested in tailoring the setup
            for their use case.</li>
          <li>Game developers.=C2=A0 Online games are commonly real-time in
            nature and benefit dramatically from ability to give up on
            transmitting old information.=C2=A0 They usually use some
            in-house UDP-based protocol, and often need to run on
            unusual platforms.=C2=A0=C2=A0</li>
        </ol>
        WebRTC Data Channels are a mechanism that provides a
        WebSocket-like interface with unreliable delivery features.=C2=A0 O=
n
        the wire, it=E2=80=99s SCTP-over-DTLS, established using ICE and SD=
P.=C2=A0
        In theory, this provides users with enough functionality to
        build anything they need, since SCTP messages can be unreliable
        and unordered.=C2=A0 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.<br>
        <br>
        The main reason for this is the incredible complexity of the
        WebRTC stack.=C2=A0 WebSockets are a fairly straightforward overlay
        on top of TCP and TLS; there is a wide variety of
        implementations out there, and it&#39;s fairly=C2=A0easy to write a=
 new
        one (I wrote on myself in less than 1,000 lines of C++).=C2=A0 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.=C2=A0 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.=C2=A0 This complexity barrier has precluded Data Channel
        adoption by communities of smaller developers who don=E2=80=99t hav=
e
        resources to implement them, notably game developers (see [1]
        and [2] for some discussion).<br>
        <br>
        Even among the people who got past the complexity barrier, the
        feedback I heard almost universally=C2=A0is that WebRTC Data Channe=
ls
        are hard to work with.=C2=A0 From the feedback I gathered, the main
        problem is usually around the transport protocol itself.=C2=A0
        Userspace SCTP is essentially a monoculture: virtually all
        implementations use libusrsctp, a 80,000-line adaptation of
        FreeBSD SCTP implementation.=C2=A0 This lack of tool choice is fair=
ly
        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).=C2=A0 In additio=
n,
        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).<br>
        <br>
        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.<br>
        <br>
        # Proposal<br>
        <br>
        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.=C2=A0 This would get us out of libusrsct=
p
        monoculture (QUIC is not yet finished, but=C2=A0 it=C2=A0already ha=
s a
        fairly diverse implementation ecosystem, see [3]), and remove
        all P2P-related complexity involving SDP and ICE.=C2=A0 The origina=
l
        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=E2=80=99t fail hard when QUI=
C is
        unavailable.<br>
        <br>
        As a result of that feedback, we abstracted it into a
        general-purpose framework called WebTransport.=C2=A0 The overview
        draft,<br>
        <br>
        =C2=A0 <a href=3D"https://tools.ietf.org/html/draft-vvv-webtranspor=
t-overview-00" target=3D"_blank">https://tools.ietf.org/html/draft-vvv-webt=
ransport-overview-00</a><br>
        <br>
        describes the framework itself, mainly the requirements the
        transport protocols have to satisfy to be usable on the web
        through the API.=C2=A0 Within this framework, we propose the
        following protocols:<br>
        <ul>
          <li>QuicTransport -- a simple WebSocket-like adaptation of
            QUIC, described in <a href=3D"https://tools.ietf.org/html/draft=
-vvv-webtransport-quic-00" target=3D"_blank">https://tools.ietf.org/html/dr=
aft-vvv-webtransport-quic-00</a></li>
          <li>Http3Transport -- a mechanism that allows creating custom
            non-HTTP streams within an HTTP/3 session, described in <a href=
=3D"https://tools.ietf.org/html/draft-vvv-webtransport-http3-00" target=3D"=
_blank">https://tools.ietf.org/html/draft-vvv-webtransport-http3-00</a>.=C2=
=A0
            This is sort of a version of RFC 8441 for QuicTransport.</li>
          <li>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).=C2=A0 W=
e
            don=E2=80=99t 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.=C2=A0 T=
he
            earlier is a more straightforward way; the latter has the
            advantage of being fully polyfillable in JavaScript.</li>
        </ul>
        <br>
        # Discussion<br>
        <br>
        At this point, I am fairly certain that there is a problem here
        that needs to be addressed.=C2=A0 I am formally requesting ART area
        to take this problem on.<br>
        <br>
        I believe the drafts above would be a good starting point for
        discussion.=C2=A0 The design that they describe went through severa=
l
        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=E2=80=99re hopefully at least looking in the =
right
        direction here.=C2=A0 I am requesting feedback on this proposal, bo=
th
        on the overall plan and the specifics described in the drafts.=C2=
=A0
        I hope to discuss this in depth in Montreal, both at dispatch
        and (in more depth) at a side-meeting.<br>
        <br>
        Thanks,<br>
        =C2=A0 Victor.<br>
        <br>
        [0] <a href=3D"https://github.com/WICG/web-transport" target=3D"_bl=
ank">https://github.com/WICG/web-transport</a>
        <div>[1] <a href=3D"https://discourse.wicg.io/t/webtransport-propos=
al/3508/9" target=3D"_blank">https://discourse.wicg.io/t/webtransport-propo=
sal/3508/9</a><br>
          [2] <a href=3D"https://news.ycombinator.com/item?id=3D13266692" t=
arget=3D"_blank">https://news.ycombinator.com/item?id=3D13266692</a><br>
          [3] <a href=3D"https://github.com/quicwg/base-drafts/wiki/Impleme=
ntations" target=3D"_blank">https://github.com/quicwg/base-drafts/wiki/Impl=
ementations</a><br>
        </div>
      </div>
      <br>
      <fieldset class=3D"gmail-m_-4560266703682762057mimeAttachmentHeader">=
</fieldset>
      <pre class=3D"gmail-m_-4560266703682762057moz-quote-pre">____________=
___________________________________
dispatch mailing list
<a class=3D"gmail-m_-4560266703682762057moz-txt-link-abbreviated" href=3D"m=
ailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a>
<a class=3D"gmail-m_-4560266703682762057moz-txt-link-freetext" href=3D"http=
s://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank">https://www.i=
etf.org/mailman/listinfo/dispatch</a>
</pre>
    </blockquote>
    <p><br>
    </p>
  </div>

_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
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</blockquote></div></div>

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(Top post  - the quote level was getting unwieldy so I=E2=80=99ve =
trimmed quite a bit from the discussion - hopefully it still makes =
sense)

> On 2 Jul 2019, at 01:10, Peter Thatcher <pthatcher@google.com> wrote:
>=20
>=20
>=20
> On Mon, Jun 17, 2019 at 7:13 AM T H Panton <thp@westhawk.co.uk =
<mailto:thp@westhawk.co.uk>> wrote:
>=20
>=20
>> On 17 Jun 2019, at 14:35, Victor Vasiliev =
<vasilvv=3D40google.com@dmarc.ietf.org =
<mailto:vasilvv=3D40google.com@dmarc.ietf.org>> wrote:
>>=20
>> Hello friendly IETF dispatchers,
>>=20
>> I am writing about new work I want to bring to IETF.  The proposal is =
called WebTransport.  It=E2=80=99s 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.
>=20
> Establish connections to where ? Between eachother P2P ? Or to any =
server? Or just to the originating server only?=20
>=20
> Any server or p2p.
> =20

Ok, great, but the requirements are different, and the OP seemed to be =
claiming simplifications that to my mind only work
client - server.

>=20
>  In client/server mode, WebTransport doesn't require any of that, so =
it is a lot more simple.=20

Right. See above. But that becomes a null statement if you do want P2P =
support .

>=20
> Work is in progress decoupling the APIs from the (dreaded) SDP and =
simplifying the developer experience.
>=20
>=20
> What work?  =20

Slow, slow, adoption of the ORTC interfaces.=20

> =20
> That said, it is by no means un-managable - I've implemented pretty =
much the whole stack from scratch for small devices and I know of 2 =
other implementations which don't use libusrsctp=20
>=20
>=20
> Maybe not for you, but we regularly get emails from people saying =
things like "It would be great to delete the mess of the WebRTC stack if =
there's a simpler protocol available. ICE/DTLS/SCTP all scare me=E2=80=9D.=


Yeah, but not as much as ICE/QUIC-RT scares me. (which by the way =
isn=E2=80=99t documented _anywhere_).

> =20
>=20
>>=20
>> 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.  =46rom 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).
>=20
> SCTP is well defined nice protocol that suffers (IMHO) from a lack of =
'coolness' due to being used by the telcos for most of it's life.
> The head-of-line issue is a problem, but entirely fixable with small =
tweaks to SCTP protocol. (read it wasn't in the original requirements - =
so it doesn't do it).
>=20
> It's been fixable for a very long time.  I have my doubts about when =
it will actually get fixed.

With that being the view from a browser team you can see why people =
choose not to donate time to submit patches that fix it.

> =20
>=20
>>=20
>> 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.
>=20
> I'd point out that 2.5bn endpoints support it right now - including =
_every_ smartphone shipped in the last 2 years. Last I heard DataChannel =
traffic was 0.1% of chrome's traffic=20
> which is _astonishingly_ high
>=20
>>=20
>> # Proposal
>>=20
>> 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=E2=80=99t fail hard when QUIC is unavailable.
>>=20
>=20
> It would be _great_ if that abstraction could _also_ work over SCTP =
data channels. That way you could cover the P2P use cases too.
>=20
> QUIC can be p2p.  In fact, we already have it implemented in Chrome =
(in an original trial).

On top of ICE with a laughable security model no media and no RFC - not =
a good basis (yet) for others to implement.

>=20
> The WebTransport API could run on top of SCTP, and you could probably =
make a JS polyfill for it.  But I'm not sure why anyone would bother.

2.5 billion shipped endpoints. near 100% penetration in smartphones =
today.

>=20
> I'd really like to be clear what that problem is.=20
>=20
>=20
> Web developers routinely ask for an unreliable websocket, or something =
close to UDP (but it needs to be secure).   It turns out that =
ICE/DTLS/SCTP is too complex to fill that roll. =20

I agree. If they want unreliable websockets to their servers, then we =
can cut out most of the layers and produce something based on QUIC
very simply. If they want P2P and want it to carry media (especially =
audio) then there is a _lot_ of work to do to ensure it is usable, =
secure etc.

It seems to me the right way forward is to define a data-only interface =
and PoC implementation that:

1) works with QUIC to the originating server almost immediately on one =
or 2 of the QUIC stacks
2) has a polyfill on top of data channels so it can be used P2P =
immediately
3) a companion GOLang/python server side implementation on top of the =
existing webRTC (non usrsctp) libraries.
4) serves as a target/requiements for the necessary protocol work on =
Quic-RT

The realtime media can be added later.

T.




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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D"">(Top =
post &nbsp;- the quote level was getting unwieldy so I=E2=80=99ve =
trimmed quite a bit from the discussion - hopefully it still makes =
sense)<br class=3D""><div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 2 Jul 2019, at 01:10, Peter Thatcher =
&lt;<a href=3D"mailto:pthatcher@google.com" =
class=3D"">pthatcher@google.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><br =
class=3D"Apple-interchange-newline"><br style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
text-decoration: none;" class=3D""><div class=3D"gmail_quote" =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none;"><div dir=3D"ltr" =
class=3D"gmail_attr">On Mon, Jun 17, 2019 at 7:13 AM T H Panton &lt;<a =
href=3D"mailto:thp@westhawk.co.uk" target=3D"_blank" =
class=3D"">thp@westhawk.co.uk</a>&gt; wrote:<br =
class=3D""></div><blockquote class=3D"gmail_quote" style=3D"margin: 0px =
0px 0px 0.8ex; border-left-width: 1px; border-left-style: solid; =
border-left-color: rgb(204, 204, 204); padding-left: 1ex;"><div =
class=3D""><br class=3D""><div class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D"">On 17 Jun 2019, at 14:35, =
Victor Vasiliev &lt;<a href=3D"mailto:vasilvv=3D40google.com@dmarc.ietf.or=
g" target=3D"_blank" =
class=3D"">vasilvv=3D40google.com@dmarc.ietf.org</a>&gt; wrote:</div><br =
class=3D"m_-8857436336868946234gmail-m_690526826935632728Apple-interchange=
-newline"><div class=3D""><div dir=3D"ltr" class=3D"">Hello friendly =
IETF dispatchers,<br class=3D""><br class=3D"">I am writing about new =
work I want to bring to IETF.&nbsp; The proposal is called =
WebTransport.&nbsp; It=E2=80=99s a combination of a Web API currently =
under development in W3C WICG [0], a protocol framework and some =
protocols that fit into that framework.&nbsp; 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).&nbsp; =
This is highly useful for real-time and other latency sensitive =
applications.<br class=3D""></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Establish connections to where ? =
Between eachother P2P ? Or to any server? Or just to the originating =
server only?&nbsp;</div></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Any server or p2p.</div><div =
class=3D"">&nbsp;</div></div></div></blockquote><div><br =
class=3D""></div><div>Ok, great, but the requirements are different, and =
the OP seemed to be claiming simplifications that to my mind only =
work</div><div>client - server.</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D"gmail_quote" style=3D"caret-color: =
rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: =
normal; font-variant-caps: normal; font-weight: normal; letter-spacing: =
normal; text-align: start; text-indent: 0px; text-transform: none; =
white-space: normal; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
text-decoration: none;"><blockquote class=3D"gmail_quote" style=3D"margin:=
 0px 0px 0px 0.8ex; border-left-width: 1px; border-left-style: solid; =
border-left-color: rgb(204, 204, 204); padding-left: 1ex;"><div =
class=3D""><div class=3D""></div></div></blockquote><br class=3D""><div =
class=3D"">&nbsp;In client/server mode, WebTransport doesn't require any =
of that, so it is a lot more simple.&nbsp;<br =
class=3D""></div></div></blockquote><div><br class=3D""></div><div>Right. =
See above. But that becomes a null statement if you do want P2P support =
.</div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><div class=3D"gmail_quote" style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
text-decoration: none;"><div class=3D""><br class=3D""></div><blockquote =
class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; =
border-left-width: 1px; border-left-style: solid; border-left-color: =
rgb(204, 204, 204); padding-left: 1ex;"><div class=3D""><div =
class=3D""><div class=3D""></div><div class=3D"">Work is in progress =
decoupling the APIs from the (dreaded) SDP and simplifying the developer =
experience.</div><div class=3D""><br =
class=3D""></div></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">What work?&nbsp; =
&nbsp;</div></div></div></blockquote><div><br class=3D""></div><div>Slow, =
slow, adoption of the ORTC interfaces.&nbsp;</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
class=3D"gmail_quote" style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: normal; text-align: start; =
text-indent: 0px; text-transform: none; white-space: normal; =
word-spacing: 0px; -webkit-text-stroke-width: 0px; text-decoration: =
none;"><div class=3D"">&nbsp;</div><blockquote class=3D"gmail_quote" =
style=3D"margin: 0px 0px 0px 0.8ex; border-left-width: 1px; =
border-left-style: solid; border-left-color: rgb(204, 204, 204); =
padding-left: 1ex;"><div class=3D""><div class=3D""><div =
class=3D""></div><div class=3D"">That said, it is by no means =
un-managable - I've implemented pretty much the whole stack from scratch =
for small devices and I know of 2 other implementations which don't use =
libusrsctp&nbsp;</div><div class=3D""><br =
class=3D""></div></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Maybe not for you, but we regularly get =
emails from people saying things like "It would be great to delete the =
mess of the WebRTC stack if there's a simpler protocol available. =
ICE/DTLS/SCTP all scare me=E2=80=9D.</div></div></div></blockquote><div><b=
r class=3D""></div><div>Yeah, but not as much as ICE/QUIC-RT scares me. =
(which by the way isn=E2=80=99t documented _anywhere_).</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
class=3D"gmail_quote" style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: normal; text-align: start; =
text-indent: 0px; text-transform: none; white-space: normal; =
word-spacing: 0px; -webkit-text-stroke-width: 0px; text-decoration: =
none;"><div class=3D"">&nbsp;</div><blockquote class=3D"gmail_quote" =
style=3D"margin: 0px 0px 0px 0.8ex; border-left-width: 1px; =
border-left-style: solid; border-left-color: rgb(204, 204, 204); =
padding-left: 1ex;"><div class=3D""><div class=3D""><div =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><div dir=3D"ltr" class=3D""><br class=3D"">Even among the =
people who got past the complexity barrier, the feedback I heard almost =
universally&nbsp;is that WebRTC Data Channels are hard to work =
with.&nbsp; =46rom the feedback I gathered, the main problem is usually =
around the transport protocol itself.&nbsp; Userspace SCTP is =
essentially a monoculture: virtually all implementations use libusrsctp, =
a 80,000-line adaptation of FreeBSD SCTP implementation.&nbsp; 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).&nbsp; 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).<br class=3D""></div></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">SCTP is well defined =
nice protocol that suffers (IMHO) from a lack of 'coolness' due to being =
used by the telcos for most of it's life.</div><div class=3D"">The =
head-of-line issue is a problem, but entirely fixable with small tweaks =
to SCTP protocol. (read it wasn't in the original requirements - so it =
doesn't do it).</div></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">It's been fixable for a very long =
time.&nbsp; I have my doubts about when it will actually get =
fixed.</div></div></div></blockquote><div><br class=3D""></div><div>With =
that being the view from a browser team you can see why people choose =
not to donate time to submit patches that fix it.</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
class=3D"gmail_quote" style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: normal; text-align: start; =
text-indent: 0px; text-transform: none; white-space: normal; =
word-spacing: 0px; -webkit-text-stroke-width: 0px; text-decoration: =
none;"><div class=3D"">&nbsp;</div><blockquote class=3D"gmail_quote" =
style=3D"margin: 0px 0px 0px 0.8ex; border-left-width: 1px; =
border-left-style: solid; border-left-color: rgb(204, 204, 204); =
padding-left: 1ex;"><div class=3D""><div class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
dir=3D"ltr" class=3D""><br class=3D"">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.<br =
class=3D""></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">I'd point out that 2.5bn endpoints =
support it right now - including _every_ smartphone shipped in the last =
2 years. Last I heard DataChannel traffic was 0.1% of chrome's =
traffic&nbsp;</div><div class=3D"">which is _astonishingly_ =
high</div></div></div></blockquote><blockquote class=3D"gmail_quote" =
style=3D"margin: 0px 0px 0px 0.8ex; border-left-width: 1px; =
border-left-style: solid; border-left-color: rgb(204, 204, 204); =
padding-left: 1ex;"><div class=3D""><div class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
dir=3D"ltr" class=3D""><br class=3D""># Proposal<br class=3D""><br =
class=3D"">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.&nbsp; This would get us out of libusrsctp monoculture (QUIC is =
not yet finished, but&nbsp; it&nbsp;already has a fairly diverse =
implementation ecosystem, see [3]), and remove all P2P-related =
complexity involving SDP and ICE.&nbsp; 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=E2=80=99t fail =
hard when QUIC is unavailable.<br class=3D""><br =
class=3D""></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">It would be _great_ if that abstraction =
could _also_ work over SCTP data channels. That way you could cover the =
P2P use cases too.</div></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">QUIC can be p2p.&nbsp; In fact, we =
already have it implemented in Chrome (in an original =
trial).</div></div></div></blockquote><div><br class=3D""></div><div>On =
top of ICE with a laughable security model no media and no RFC - not a =
good basis (yet) for others to implement.</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><div class=3D"gmail_quote" =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none;"><div =
class=3D""><br class=3D""></div><div class=3D"">The WebTransport API =
could run on top of SCTP, and you could probably make a JS polyfill for =
it.&nbsp; But I'm not sure why anyone would =
bother.</div></div></div></blockquote><div><br class=3D""></div>2.5 =
billion shipped endpoints. near 100% penetration in smartphones =
today.</div><div><br class=3D""></div><div><blockquote type=3D"cite" =
class=3D""><div class=3D""><div class=3D"gmail_quote" =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none;"><blockquote =
class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; =
border-left-width: 1px; border-left-style: solid; border-left-color: =
rgb(204, 204, 204); padding-left: 1ex;"><div class=3D""><div =
class=3D""><div class=3D""><br class=3D""></div><div class=3D"">I'd =
really like to be clear what that problem is.&nbsp;</div><br =
class=3D""></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Web developers routinely ask for an =
unreliable websocket, or something close to UDP (but it needs to be =
secure).&nbsp; &nbsp;It turns out that ICE/DTLS/SCTP is too complex to =
fill that roll.&nbsp;&nbsp;</div></div></div></blockquote><div><br =
class=3D""></div><div>I agree. If they want unreliable websockets to =
their servers, then we can cut out most of the layers and produce =
something based on QUIC</div><div>very simply. If they want P2P and want =
it to carry media (especially audio) then there is a _lot_ of work to do =
to ensure it is usable, secure etc.</div><div><br class=3D""></div><div>It=
 seems to me the right way forward is to define a data-only interface =
and PoC implementation that:</div><div><br class=3D""></div><div>1) =
works with QUIC to the originating server almost immediately on one or 2 =
of the QUIC stacks</div><div>2) has a polyfill on top of data channels =
so it can be used P2P immediately</div><div>3) a companion GOLang/python =
server side implementation on top of the existing webRTC (non usrsctp) =
libraries.</div><div>4) serves as a target/requiements for the necessary =
protocol work on Quic-RT</div><div><br class=3D""></div><div>The =
realtime media can be added later.</div><div><br =
class=3D""></div><div>T.</div><div><br class=3D""></div></div><br =
class=3D""><div class=3D""><br class=3D""></div></body></html>=

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> On 2 Jul 2019, at 01:10, Peter Thatcher <pthatcher@google.com> wrote:
>=20
> Maybe not for you, but we regularly get emails from people saying =
things like "It would be great to delete the mess of the WebRTC stack if =
there's a simpler protocol available. ICE/DTLS/SCTP all scare me=E2=80=9D.=


To be clear, I utterly sympathise with this viewpoint - no one in their =
right minds wants to pull libwebrtc into their server codebase.

I took one look at it and decided to write a fresh implementation that =
better suited my needs.
The fact that I could do that is due to the existence of RFCs for all of =
the  webRTC stack layers.

When implementing the datachannel I did ask a few questions of the =
experts, their replies were pointers
to the relevant bits of the RFCs. QUIC-RT simply isn=E2=80=99t in that =
position yet.

T.


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 2 Jul 2019, at 01:10, Peter Thatcher &lt;<a =
href=3D"mailto:pthatcher@google.com" =
class=3D"">pthatcher@google.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none; float: none; =
display: inline !important;" class=3D"">Maybe not for you, but we =
regularly get emails from people saying things like "It would be great =
to delete the mess of the WebRTC stack if there's a simpler protocol =
available. ICE/DTLS/SCTP all scare =
me=E2=80=9D.</span></div></blockquote><div><br class=3D""></div>To be =
clear, I utterly sympathise with this viewpoint - no one in their right =
minds wants to pull libwebrtc into their server codebase.</div><div><br =
class=3D""></div><div>I took one look at it and decided to write a fresh =
implementation that better suited my needs.</div><div>The fact that I =
could do that is due to the existence of RFCs for all of the =
&nbsp;webRTC stack layers.</div><div><br class=3D""></div><div>When =
implementing the datachannel I did ask a few questions of the experts, =
their replies were pointers</div><div>to the relevant bits of the RFCs. =
QUIC-RT simply isn=E2=80=99t in that position yet.</div><div><br =
class=3D""></div><div>T.</div><br class=3D""></body></html>=

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From: Peter Thatcher <pthatcher@google.com>
Date: Mon, 8 Jul 2019 11:47:39 -0700
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Subject: Re: [dispatch] Dispatching WebTransport
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On Tue, Jul 2, 2019 at 1:22 AM westhawk <thp@westhawk.co.uk> wrote:

> (Top post  - the quote level was getting unwieldy so I=E2=80=99ve trimmed=
 quite a
> bit from the discussion - hopefully it still makes sense)
>
> On 2 Jul 2019, at 01:10, Peter Thatcher <pthatcher@google.com> wrote:
>
>
>
> On Mon, Jun 17, 2019 at 7:13 AM T H Panton <thp@westhawk.co.uk> wrote:
>
>>
>>
>> On 17 Jun 2019, at 14:35, Victor Vasiliev <
>> vasilvv=3D40google.com@dmarc.ietf.org> wrote:
>>
>> Hello friendly IETF dispatchers,
>>
>> I am writing about new work I want to bring to IETF.  The proposal is
>> called WebTransport.  It=E2=80=99s a combination of a Web API currently =
under
>> development in W3C WICG [0], a protocol framework and some protocols tha=
t
>> 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 an=
d
>> streams do not have head-of-line blocking).  This is highly useful for
>> real-time and other latency sensitive applications.
>>
>>
>> Establish connections to where ? Between eachother P2P ? Or to any
>> server? Or just to the originating server only?
>>
>
> Any server or p2p.
>
>
>
> Ok, great, but the requirements are different, and the OP seemed to be
> claiming simplifications that to my mind only work
> client - server.
>
>
>  In client/server mode, WebTransport doesn't require any of that, so it i=
s
> a lot more simple.
>
>
> Right. See above. But that becomes a null statement if you do want P2P
> support .
>
>
> Work is in progress decoupling the APIs from the (dreaded) SDP and
>> simplifying the developer experience.
>>
>>
> What work?
>
>
> Slow, slow, adoption of the ORTC interfaces.
>

>
>> That said, it is by no means un-managable - I've implemented pretty much
>> the whole stack from scratch for small devices and I know of 2 other
>> implementations which don't use libusrsctp
>>
>>
> Maybe not for you, but we regularly get emails from people saying things
> like "It would be great to delete the mess of the WebRTC stack if there's=
 a
> simpler protocol available. ICE/DTLS/SCTP all scare me=E2=80=9D.
>
>
> Yeah, but not as much as ICE/QUIC-RT scares me. (which by the way isn=E2=
=80=99t
> documented _anywhere_).
>
>
>
>>
>>
>> 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 o=
f
>> 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 congest=
ion
>> control, etc).  In addition, the limitations on the message size stemmin=
g
>> 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-block=
ing
>> (this is particularly bad because the framing overhead in data channels =
is
>> already large as-is).
>>
>>
>> SCTP is well defined nice protocol that suffers (IMHO) from a lack of
>> 'coolness' due to being used by the telcos for most of it's life.
>> The head-of-line issue is a problem, but entirely fixable with small
>> tweaks to SCTP protocol. (read it wasn't in the original requirements - =
so
>> it doesn't do it).
>>
>
> It's been fixable for a very long time.  I have my doubts about when it
> will actually get fixed.
>
>
> With that being the view from a browser team you can see why people choos=
e
> not to donate time to submit patches that fix it.
>
>
>
>>
>>
>> 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.
>>
>>
>> I'd point out that 2.5bn endpoints support it right now - including
>> _every_ smartphone shipped in the last 2 years. Last I heard DataChannel
>> traffic was 0.1% of chrome's traffic
>> which is _astonishingly_ high
>>
>
>>
>> # 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 t=
he
>> 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, bu=
t
>> 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 peop=
le,
>> 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=E2=
=80=99t
>> fail hard when QUIC is unavailable.
>>
>>
>> It would be _great_ if that abstraction could _also_ work over SCTP data
>> channels. That way you could cover the P2P use cases too.
>>
>
> QUIC can be p2p.  In fact, we already have it implemented in Chrome (in a=
n
> original trial).
>
>
> On top of ICE with a laughable security model no media and no RFC - not a
> good basis (yet) for others to implement.
>
>
It's the same security model as the RTCDataChannel uses, and it can
transfer media just fine (transport doesn't care if it's media or not).  As
for RFCs, there are ICE and QUIC RFCs.  What more do you need?  If you
tried to write a "ICE + QUIC" RFC, I'm pretty sure it would be all
boilerplate and no actual text.

>
> The WebTransport API could run on top of SCTP, and you could probably mak=
e
> a JS polyfill for it.  But I'm not sure why anyone would bother.
>
>
> 2.5 billion shipped endpoints. near 100% penetration in smartphones today=
.
>
>
>> I'd really like to be clear what that problem is.
>>
>>
> Web developers routinely ask for an unreliable websocket, or something
> close to UDP (but it needs to be secure).   It turns out that ICE/DTLS/SC=
TP
> is too complex to fill that roll.
>
>
> I agree. If they want unreliable websockets to their servers, then we can
> cut out most of the layers and produce something based on QUIC
> very simply. If they want P2P and want it to carry media (especially
> audio) then there is a _lot_ of work to do to ensure it is usable, secure
> etc.
>
>
When you say "a lot" of work, what do you mean?  You can already send audio
on top of the QUIC-based WebTransport we have in origin trial and it's
usable and secure.  It would be nice to have low-level audio codec APIs in
the browser (which we are working on), but it's feasible to do audio codecs
in WASM in conjunction with WebAudio.

The only thing that seems like an issue is proving out whether or not one
can implement a good audio jitter buffer in WASM/JS (or port NetEq).  That
is work left to be done, indeed.  Is that what you were referring to?



> It seems to me the right way forward is to define a data-only interface
> and PoC implementation that:
>
> 1) works with QUIC to the originating server almost immediately on one or
> 2 of the QUIC stacks
>

I'm not completely sure what you mean here by "one or two of the QUIC
stacks", but see #3 below for code that can (hopefully soon) can do the
server side.

2) has a polyfill on top of data channels so it can be used P2P immediately
>
 3) a companion GOLang/python server side implementation on top of the
existing webRTC (non usrsctp) libraries.

I have one in progress:
https://github.com/pthatcherg/quic-go/blob/gquic/example/webtransport/serve=
r.go
.
I don't quite have it working, but it's close.

And there is also this one:
https://github.com/pion/webrtc/blob/736e0bb5062c473a61d9aae0dbb3bac7f32db37=
7/quictransport.go

4) serves as a target/requiements for the necessary protocol work on Quic-R=
T
>
> The realtime media can be added later.
>
> T.
>
>
>
>

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<div dir=3D"ltr"><div dir=3D"ltr"><br></div><br><div class=3D"gmail_quote">=
<div dir=3D"ltr" class=3D"gmail_attr">On Tue, Jul 2, 2019 at 1:22 AM westha=
wk &lt;<a href=3D"mailto:thp@westhawk.co.uk">thp@westhawk.co.uk</a>&gt; wro=
te:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px =
0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=
=3D"overflow-wrap: break-word;">(Top post =C2=A0- the quote level was getti=
ng unwieldy so I=E2=80=99ve trimmed quite a bit from the discussion - hopef=
ully it still makes sense)<br><div><br><blockquote type=3D"cite"><div>On 2 =
Jul 2019, at 01:10, Peter Thatcher &lt;<a href=3D"mailto:pthatcher@google.c=
om" target=3D"_blank">pthatcher@google.com</a>&gt; wrote:</div><br class=3D=
"gmail-m_2905353180801879657Apple-interchange-newline"><div><br class=3D"gm=
ail-m_2905353180801879657Apple-interchange-newline"><br style=3D"font-famil=
y:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-=
weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-t=
ransform:none;white-space:normal;word-spacing:0px;text-decoration:none"><di=
v class=3D"gmail_quote" style=3D"font-family:Helvetica;font-size:12px;font-=
style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:nor=
mal;text-align:start;text-indent:0px;text-transform:none;white-space:normal=
;word-spacing:0px;text-decoration:none"><div dir=3D"ltr" class=3D"gmail_att=
r">On Mon, Jun 17, 2019 at 7:13 AM T H Panton &lt;<a href=3D"mailto:thp@wes=
thawk.co.uk" target=3D"_blank">thp@westhawk.co.uk</a>&gt; wrote:<br></div><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft:1px solid rgb(204,204,204);padding-left:1ex"><div><br><div><br><blockqu=
ote type=3D"cite"><div>On 17 Jun 2019, at 14:35, Victor Vasiliev &lt;<a hre=
f=3D"mailto:vasilvv=3D40google.com@dmarc.ietf.org" target=3D"_blank">vasilv=
v=3D40google.com@dmarc.ietf.org</a>&gt; wrote:</div><br class=3D"gmail-m_29=
05353180801879657m_-8857436336868946234gmail-m_690526826935632728Apple-inte=
rchange-newline"><div><div dir=3D"ltr">Hello friendly IETF dispatchers,<br>=
<br>I am writing about new work I want to bring to IETF.=C2=A0 The proposal=
 is called WebTransport.=C2=A0 It=E2=80=99s a combination of a Web API curr=
ently under development in W3C WICG [0], a protocol framework and some prot=
ocols that fit into that framework.=C2=A0 Combined, they would allow web ap=
plications to establish WebSocket-like connections that instead of ordered =
reliable messages use multiple streams and datagrams (datagrams are unrelia=
ble and streams do not have head-of-line blocking).=C2=A0 This is highly us=
eful for real-time and other latency sensitive applications.<br></div></div=
></blockquote><div><br></div><div>Establish connections to where ? Between =
eachother P2P ? Or to any server? Or just to the originating server only?=
=C2=A0</div></div></div></blockquote><div><br></div><div>Any server or p2p.=
</div><div>=C2=A0</div></div></div></blockquote><div><br></div><div>Ok, gre=
at, but the requirements are different, and the OP seemed to be claiming si=
mplifications that to my mind only work</div><div>client - server.</div><br=
><blockquote type=3D"cite"><div class=3D"gmail_quote" style=3D"font-family:=
Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-we=
ight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-tra=
nsform:none;white-space:normal;word-spacing:0px;text-decoration:none"><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:=
1px solid rgb(204,204,204);padding-left:1ex"><div><div></div></div></blockq=
uote><br><div>=C2=A0In client/server mode, WebTransport doesn&#39;t require=
 any of that, so it is a lot more simple.=C2=A0<br></div></div></blockquote=
><div><br></div><div>Right. See above. But that becomes a null statement if=
 you do want P2P support .</div><br><blockquote type=3D"cite"><div><div cla=
ss=3D"gmail_quote" style=3D"font-family:Helvetica;font-size:12px;font-style=
:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;t=
ext-align:start;text-indent:0px;text-transform:none;white-space:normal;word=
-spacing:0px;text-decoration:none"><div><br></div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,20=
4,204);padding-left:1ex"><div><div><div></div><div>Work is in progress deco=
upling the APIs from the (dreaded) SDP and simplifying the developer experi=
ence.</div><div><br></div></div></div></blockquote><div><br></div><div>What=
 work?=C2=A0 =C2=A0</div></div></div></blockquote><div><br></div><div>Slow,=
 slow, adoption of the ORTC interfaces.=C2=A0</div></div></div></blockquote=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-w=
rap: break-word;"><div><blockquote type=3D"cite"><div><div class=3D"gmail_q=
uote" style=3D"font-family:Helvetica;font-size:12px;font-style:normal;font-=
variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:sta=
rt;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;=
text-decoration:none"><div>=C2=A0</div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padd=
ing-left:1ex"><div><div><div></div><div>That said, it is by no means un-man=
agable - I&#39;ve implemented pretty much the whole stack from scratch for =
small devices and I know of 2 other implementations which don&#39;t use lib=
usrsctp=C2=A0</div><div><br></div></div></div></blockquote><div><br></div><=
div>Maybe not for you, but we regularly get emails from people saying thing=
s like &quot;It would be great to delete the mess of the WebRTC stack if th=
ere&#39;s a simpler protocol available. ICE/DTLS/SCTP all scare me=E2=80=9D=
.</div></div></div></blockquote><div><br></div><div>Yeah, but not as much a=
s ICE/QUIC-RT scares me. (which by the way isn=E2=80=99t documented _anywhe=
re_).</div><br><blockquote type=3D"cite"><div><div class=3D"gmail_quote" st=
yle=3D"font-family:Helvetica;font-size:12px;font-style:normal;font-variant-=
caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-=
indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-dec=
oration:none"><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left=
:1ex"><div><div><div></div><br><blockquote type=3D"cite"><div><div dir=3D"l=
tr"><br>Even among the people who got past the complexity barrier, the feed=
back I heard almost universally=C2=A0is that WebRTC Data Channels are hard =
to work with.=C2=A0 From the feedback I gathered, the main problem is usual=
ly around the transport protocol itself.=C2=A0 Userspace SCTP is essentiall=
y a monoculture: virtually all implementations use libusrsctp, a 80,000-lin=
e adaptation of FreeBSD SCTP implementation.=C2=A0 This lack of tool choice=
 is fairly painful since latency-sensitive real-time applications often req=
uire quite a bit of tuning on the transport side to get the best performanc=
e (custom congestion control, etc).=C2=A0 In addition, the limitations on t=
he message size stemming from both the API itself and the lack of widesprea=
d support for message interleaving (RFC 8260) means that the developers hav=
e to roll their own framing on top of SCTP messages if they want to avoid h=
ead-of-line-blocking (this is particularly bad because the framing overhead=
 in data channels is already large as-is).<br></div></div></blockquote><div=
><br></div><div>SCTP is well defined nice protocol that suffers (IMHO) from=
 a lack of &#39;coolness&#39; due to being used by the telcos for most of i=
t&#39;s life.</div><div>The head-of-line issue is a problem, but entirely f=
ixable with small tweaks to SCTP protocol. (read it wasn&#39;t in the origi=
nal requirements - so it doesn&#39;t do it).</div></div></div></blockquote>=
<div><br></div><div>It&#39;s been fixable for a very long time.=C2=A0 I hav=
e my doubts about when it will actually get fixed.</div></div></div></block=
quote><div><br></div><div>With that being the view from a browser team you =
can see why people choose not to donate time to submit patches that fix it.=
</div><br><blockquote type=3D"cite"><div><div class=3D"gmail_quote" style=
=3D"font-family:Helvetica;font-size:12px;font-style:normal;font-variant-cap=
s:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-ind=
ent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decora=
tion:none"><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1e=
x"><div><div><br><blockquote type=3D"cite"><div><div dir=3D"ltr"><br>In sum=
mary, we have a system that technically provides what everyone wants, but t=
hat nobody is happy with, and that is not usable by all but the most well-r=
esourced users.<br></div></div></blockquote><div><br></div><div>I&#39;d poi=
nt out that 2.5bn endpoints support it right now - including _every_ smartp=
hone shipped in the last 2 years. Last I heard DataChannel traffic was 0.1%=
 of chrome&#39;s traffic=C2=A0</div><div>which is _astonishingly_ high</div=
></div></div></blockquote><blockquote class=3D"gmail_quote" style=3D"margin=
:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"=
><div><div><br><blockquote type=3D"cite"><div><div dir=3D"ltr"><br># Propos=
al<br><br>Our initial idea for fixing this was to take QUIC and do what Web=
Socket 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.=
=C2=A0 This would get us out of libusrsctp monoculture (QUIC is not yet fin=
ished, but=C2=A0 it=C2=A0already has a fairly diverse implementation ecosys=
tem, see [3]), and remove all P2P-related complexity involving SDP and ICE.=
=C2=A0 The original proposal for that was called QuicTransport; we showed i=
t 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 SCT=
P to QUIC, tying it to QUIC specificially would not be wise), and (2) it sh=
ouldn=E2=80=99t fail hard when QUIC is unavailable.<br><br></div></div></bl=
ockquote><div><br></div><div>It would be _great_ if that abstraction could =
_also_ work over SCTP data channels. That way you could cover the P2P use c=
ases too.</div></div></div></blockquote><div><br></div><div>QUIC can be p2p=
.=C2=A0 In fact, we already have it implemented in Chrome (in an original t=
rial).</div></div></div></blockquote><div><br></div><div>On top of ICE with=
 a laughable security model no media and no RFC - not a good basis (yet) fo=
r others to implement.</div><br></div></div></blockquote><div><br></div><di=
v>It&#39;s the same security model as the RTCDataChannel uses, and it can t=
ransfer media just fine (transport doesn&#39;t care if it&#39;s media or no=
t).=C2=A0 As for RFCs, there are ICE and QUIC RFCs.=C2=A0 What more do you =
need?=C2=A0 If you tried to write a &quot;ICE=C2=A0+ QUIC&quot; RFC, I&#39;=
m pretty sure it would be all boilerplate and no actual text.<br></div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left=
:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: =
break-word;"><div><blockquote type=3D"cite"><div><div class=3D"gmail_quote"=
 style=3D"font-family:Helvetica;font-size:12px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;te=
xt-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-=
decoration:none"><div><br></div><div>The WebTransport API could run on top =
of SCTP, and you could probably make a JS polyfill for it.=C2=A0 But I&#39;=
m not sure why anyone would bother.</div></div></div></blockquote><div><br>=
</div>2.5 billion shipped endpoints. near 100% penetration in smartphones t=
oday.</div><div><br></div></div></blockquote><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204=
);padding-left:1ex"><div style=3D"overflow-wrap: break-word;"><div><blockqu=
ote type=3D"cite"><div><div class=3D"gmail_quote" style=3D"font-family:Helv=
etica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight=
:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transfo=
rm:none;white-space:normal;word-spacing:0px;text-decoration:none"><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px =
solid rgb(204,204,204);padding-left:1ex"><div><div><div><br></div><div>I&#3=
9;d really like to be clear what that problem is.=C2=A0</div><br></div></di=
v></blockquote><div><br></div><div>Web developers routinely ask for an unre=
liable websocket, or something close to UDP (but it needs to be secure).=C2=
=A0 =C2=A0It turns out that ICE/DTLS/SCTP is too complex to fill that roll.=
=C2=A0=C2=A0</div></div></div></blockquote><div><br></div><div>I agree. If =
they want unreliable websockets to their servers, then we can cut out most =
of the layers and produce something based on QUIC</div><div>very simply. If=
 they want P2P and want it to carry media (especially audio) then there is =
a _lot_ of work to do to ensure it is usable, secure etc.</div><div><br></d=
iv></div></div></blockquote><div><br>When you say &quot;a lot&quot; of work=
, what do you mean?=C2=A0 You can already send audio on top of the QUIC-bas=
ed WebTransport we have in origin trial and it&#39;s usable and secure.=C2=
=A0 It would be nice to have low-level audio codec APIs in the browser (whi=
ch we are working on), but it&#39;s feasible to do audio codecs in WASM in =
conjunction with WebAudio.</div><div><br></div><div>The only thing that see=
ms like an issue is proving out whether or not one can implement a good aud=
io jitter buffer in WASM/JS (or port NetEq).=C2=A0 That is work left to be =
done, indeed.=C2=A0 Is that what you were referring to?</div><div><br></div=
><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div st=
yle=3D"overflow-wrap: break-word;"><div><div></div><div>It seems to me the =
right way forward is to define a data-only interface and PoC implementation=
 that:</div><div><br></div><div>1) works with QUIC to the originating serve=
r almost immediately on one or 2 of the QUIC stacks</div></div></div></bloc=
kquote><div><br></div><div>I&#39;m not completely sure what you mean here b=
y &quot;one or two of the QUIC stacks&quot;, but see #3 below for code that=
 can (hopefully soon) can do the server side.</div><div><br></div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px =
solid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break=
-word;"><div><div>2) has a polyfill on top of data channels so it can be us=
ed P2P immediately</div></div></div></blockquote><div>=C2=A03) a companion =
GOLang/python server side implementation on top of the existing webRTC (non=
 usrsctp) libraries.</div><div><br></div><div>I have one in progress:=C2=A0=
<a href=3D"https://github.com/pthatcherg/quic-go/blob/gquic/example/webtran=
sport/server.go">https://github.com/pthatcherg/quic-go/blob/gquic/example/w=
ebtransport/server.go</a>=C2=A0.=C2=A0 I don&#39;t quite have it working, b=
ut it&#39;s close.</div><div><br></div><div>And there is also this one:=C2=
=A0=C2=A0<a href=3D"https://github.com/pion/webrtc/blob/736e0bb5062c473a61d=
9aae0dbb3bac7f32db377/quictransport.go">https://github.com/pion/webrtc/blob=
/736e0bb5062c473a61d9aae0dbb3bac7f32db377/quictransport.go</a></div><div><b=
r></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex=
;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"ove=
rflow-wrap: break-word;"><div><div>4) serves as a target/requiements for th=
e necessary protocol work on Quic-RT</div><div><br></div><div>The realtime =
media can be added later.</div><div><br></div><div>T.</div><div><br></div><=
/div><br><div><br></div></div></blockquote></div></div>

--0000000000000d2dce058d2fe3a8--


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Subject: [dispatch] IETF105 DISPATCH Agenda
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Hi All,

The preliminary agenda for the IETF105 DISPATCH meeting ia available at =
https://datatracker.ietf.org/meeting/105/materials/agenda-105-dispatch-00 =
<https://datatracker.ietf.org/meeting/105/materials/agenda-105-dispatch-00=
>. A text version is copied below for your convenience, but please bear =
in mind that the agenda may change between now and the meeting.

Also, please notice that DISPATCH will be in the last afternoon session =
on Monday, rather than the usual Monday morning session.

Thanks!

Ben

=E2=80=94=E2=80=94=E2=80=94

# Agenda DISPATCH @IETF-105 - v2

DISPATCH Meeting
----------------

### Status and agenda bash - Chairs and ADs (10 min)

### WebTransport - Victor Vasiliev (20 min)

=
[Overview](https://tools.ietf.org/html/draft-vvv-webtransport-overview-00)=

[QUIC =
Transport](https://tools.ietf.org/html/draft-vvv-webtransport-quic-00)
[H3 =
Transport](https://tools.ietf.org/html/draft-vvv-webtransport-http3-00)
=
[Discussion](https://mailarchive.ietf.org/arch/msg/dispatch/sIih9gBhJ4_fao=
N0GrQ9Xweue2w)

See [DISPATCH wiki](https://trac.ietf.org/trac/dispatch/wiki) for =
additional links to materials.

ART AREA Meeting
----------------

### BoFs and meetings of interest - ADs (10 min)

### Real Time Internet Peering Protocol (RIPP) - Cullen Jennings (10 =
min)

=
[Draft](https://tools.ietf.org/html/draft-rosenbergjennings-dispatch-ripp-=
02)

### BCP 190 Applicability to TRANS - TBD (10 min)

=
[Discussion](https://mailarchive.ietf.org/arch/msg/trans/_P51dDMy_-Mo3lojb=
ijQPySv3q8)

## AOB

--Apple-Mail=_FCB44BC9-46B4-420E-8735-BF7A98600854
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space;" class=3D"">Hi All,<div =
class=3D""><br class=3D""></div><div class=3D"">The preliminary agenda =
for the IETF105 DISPATCH meeting ia available at&nbsp;<a =
href=3D"https://datatracker.ietf.org/meeting/105/materials/agenda-105-disp=
atch-00" =
class=3D"">https://datatracker.ietf.org/meeting/105/materials/agenda-105-d=
ispatch-00</a>. A text version is copied below for your convenience, but =
please bear in mind that the agenda may change between now and the =
meeting.</div><div class=3D""><br class=3D""></div><div class=3D"">Also, =
please notice that DISPATCH will be in the last afternoon session on =
Monday, rather than the usual Monday morning session.</div><div =
class=3D""><br class=3D""></div><div class=3D"">Thanks!</div><div =
class=3D""><br class=3D""></div><div class=3D"">Ben</div><div =
class=3D""><br class=3D""></div><div class=3D"">=E2=80=94=E2=80=94=E2=80=94=
</div><div class=3D""><br class=3D""></div><div class=3D""><div =
class=3D""># Agenda DISPATCH @IETF-105 - v2</div><div class=3D""><br =
class=3D""></div><div class=3D"">DISPATCH Meeting&nbsp;</div><div =
class=3D"">----------------</div><div class=3D""><br class=3D""></div><div=
 class=3D"">### Status and agenda bash - Chairs and ADs (10 =
min)</div><div class=3D""><br class=3D""></div><div class=3D"">### =
WebTransport - Victor Vasiliev (20 min)</div><div class=3D""><br =
class=3D""></div><div class=3D"">[Overview](<a =
href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-overview-00" =
class=3D"">https://tools.ietf.org/html/draft-vvv-webtransport-overview-00<=
/a>) &nbsp;</div><div class=3D"">[QUIC Transport](<a =
href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-quic-00" =
class=3D"">https://tools.ietf.org/html/draft-vvv-webtransport-quic-00</a>)=
 &nbsp;</div><div class=3D"">[H3 Transport](<a =
href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-http3-00" =
class=3D"">https://tools.ietf.org/html/draft-vvv-webtransport-http3-00</a>=
) &nbsp;</div><div class=3D"">[Discussion](<a =
href=3D"https://mailarchive.ietf.org/arch/msg/dispatch/sIih9gBhJ4_faoN0GrQ=
9Xweue2w" =
class=3D"">https://mailarchive.ietf.org/arch/msg/dispatch/sIih9gBhJ4_faoN0=
GrQ9Xweue2w</a>)</div><div class=3D"">&nbsp;&nbsp;</div><div =
class=3D"">See [DISPATCH wiki](<a =
href=3D"https://trac.ietf.org/trac/dispatch/wiki" =
class=3D"">https://trac.ietf.org/trac/dispatch/wiki</a>) for additional =
links to materials.</div><div class=3D""><br class=3D""></div><div =
class=3D"">ART AREA Meeting&nbsp;</div><div =
class=3D"">----------------</div><div class=3D""><br class=3D""></div><div=
 class=3D"">### BoFs and meetings of interest - ADs (10 min)</div><div =
class=3D""><br class=3D""></div><div class=3D"">### Real Time Internet =
Peering Protocol (RIPP) - Cullen Jennings (10 min)</div><div =
class=3D""><br class=3D""></div><div class=3D"">[Draft](<a =
href=3D"https://tools.ietf.org/html/draft-rosenbergjennings-dispatch-ripp-=
02" =
class=3D"">https://tools.ietf.org/html/draft-rosenbergjennings-dispatch-ri=
pp-02</a>)</div><div class=3D""><br class=3D""></div><div class=3D"">### =
BCP 190 Applicability to TRANS - TBD (10 min)</div><div class=3D""><br =
class=3D""></div><div class=3D"">[Discussion](<a =
href=3D"https://mailarchive.ietf.org/arch/msg/trans/_P51dDMy_-Mo3lojbijQPy=
Sv3q8" =
class=3D"">https://mailarchive.ietf.org/arch/msg/trans/_P51dDMy_-Mo3lojbij=
QPySv3q8</a>)</div><div class=3D""><br class=3D""></div><div class=3D"">##=
 AOB</div></div></body></html>=

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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 10 Jul 2019 12:59:45 -0700
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Subject: Re: [dispatch] Dispatching WebTransport (boiling it down)
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Inaki Baz Castillo said:

"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?"

[BA] Indeed.  The protocol proposals in draft-vvv-webtransport-quic and
draft-vvv-webtransport-http3 relate to the client/server scenario, and so
issues relating to peer-to-peer operation, ICE, WebRTC, W3C APIs, etc. seem
to be peripheral.

To my mind, draft-vvv-webtransport-http3 appears to raise the following
question:

"Do we need additional work to update "Bootstrapping WebSockets with HTTP/2"
(RFC 8441), in order to enable bootstrapping {bi-directional stream,
uni-directional stream, datagrams} on HTTP/3"?

Given that QUIC offers relief from TCP's head-of-line blocking problems,
one can make the case that WebTransport(s) over HTTP/3 will perform better
than WebSocket(s) over HTTP/2. Also, I can see uses for a uni-directional
stream (server -> client or client -> server). And with the recent interest
in streaming over unreliable transport (e.g. SRT), the datagram mode could
come in handy as well.

Since draft-ietf-quic-transport only covers bi-directional and
uni-directional streams, it might make sense to focus on that functionality
initially and then add datagrams support once we understand how and where
QUIC datagrams (as described in draft-pauly-quic-datagram ) will be
progressed.

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<div dir=3D"ltr">Inaki Baz Castillo said:=C2=A0<div><br></div><div>&quot;<s=
pan style=3D"color:rgb(33,37,41);font-family:SFMono-Regular,Menlo,Monaco,Co=
nsolas,&quot;Liberation Mono&quot;,&quot;Courier New&quot;,monospace;font-s=
ize:12.25px;white-space:pre-wrap">if this is a pure client-to-server transp=
ort, why is webrtc mentioned in the background at all? shouldn&#39;t this b=
e a replacement of websockets and NOT of the webrtc datachannels?</span>&qu=
ot;</div><div><br></div><div>[BA] Indeed.=C2=A0 The protocol proposals in d=
raft-vvv-webtransport-quic and draft-vvv-webtransport-http3 relate to the c=
lient/server scenario, and so issues relating to peer-to-peer operation, IC=
E, WebRTC, W3C APIs, etc. seem to be peripheral.=C2=A0</div><div><br></div>=
<div>To my mind, draft-vvv-webtransport-http3 appears to raise the followin=
g question:=C2=A0</div><div><br></div><div>&quot;Do we need additional work=
 to update &quot;<span style=3D"font-size:1em;font-weight:bold;color:rgb(0,=
0,0)">Bootstrapping WebSockets with HTTP/2</span>&quot; (RFC 8441), in orde=
r to enable bootstrapping {bi-directional stream, uni-directional stream, d=
atagrams} on HTTP/3&quot;?=C2=A0</div><div><br></div><div>Given that QUIC o=
ffers relief from TCP&#39;s head-of-line blocking problems, one can make th=
e case that WebTransport(s) over HTTP/3 will perform better than WebSocket(=
s) over HTTP/2. Also, I can see uses for a uni-directional stream (server -=
&gt; client or client -&gt; server). And with the recent interest in stream=
ing over unreliable transport (e.g. SRT), the datagram mode could come in h=
andy as well.=C2=A0</div><div><br></div><div>Since draft-ietf-quic-transpor=
t only covers bi-directional and uni-directional streams, it might make sen=
se to focus on that functionality initially and then add datagrams support =
once we understand how and where QUIC datagrams (as described in=C2=A0draft=
-pauly-quic-datagram ) will be progressed.</div><div><br></div><div><br></d=
iv><div><br></div><div><br></div><div><br></div><div><br></div></div>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 11 Jul 2019 10:34:55 +0200
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On Wed, 10 Jul 2019 at 21:59, Bernard Aboba <bernard.aboba@gmail.com> wrote=
:
>
> Inaki Baz Castillo said:
>
> "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 N=
OT of the webrtc datachannels?"

Actually I did not say that =F0=9F=91=86

:)

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>


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Subject: [dispatch] Thoughts on draft-rosenbergjennings-dispatch-ripp-03
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General: For the most part, I like the idea of a draft that basically =
says, =E2=80=9Cwe tried to be pure but failed. This proposal is not pure =
but it for the most part represents running code.=E2=80=9D PHB will be =
happy (ref. the NAT discussion on the IETF list).


Section 1.1: background
Will much of it be meaningful to a protocol implementer in a year, no =
less five? I.e., referencing Kubernetes, Istio, CanaryDeploys, etc., =
more especially without going into detail as to what they are? Sure, if =
you are an AWS developer you know them like the back of your hand. =
However, not many folks in IETF-land are.

1.2:
Why is it =E2=80=98unfortunate=E2=80=99 that people want to deploy cloud =
applications that interact with the PSTN? Is that not an opportunity? I =
would take out the initial =E2=80=9CUnfortunately.=E2=80=9D It=E2=80=99s =
just a statement of fact that applications like to call and be called to =
most any person on the planet on a fully interoperable basis.

There needs to be more justification than =E2=80=9CWeb people don=E2=80=99=
t understand SIP.=E2=80=9D You might want to explain how RIPP is =
different than ParlayX. I would offer that RIPP:ParlayX::LDAP:X.500, but =
you don=E2=80=99t say that anywhere. All the draft implies is Web people =
cannot grok SIP and SIP does not grok Web deployment, which does not =
reflect well on either community.

The last paragraph of 1.2 implies that SIP is the source of robocalls =
and a mechanism that will be even easier to inject boatloads of calls =
will be better. Please explain how this will be so (or remove the =
assertion). As you know, I personally wish it would be so, so PLEASE =
give me ammunition!

1.4:
Given the virtual mandate to use SIP over TCP, is saying =E2=80=9Cwe can =
finally use HTTP over UDP=E2=80=9D not kind of a retrograde assertion?

REQ 12 (supporting only audio) is an odd requirement. Moreover, most =
applications in the U.S. would require at least RTT support if not =
video. I am not a lawyer, but since we are talking interoperability with =
the PSTN, one is likely to run into Section 255 and 706 issues if RTT =
(and video) is not there from day one.

REQ 13 looks totally duplicative of REQ 10 (use secure caller ID =
technologies). Is there a distinction? If so, please make it clearer

Section 3.6:
This is a place where we need to distinguish between theory and reality. =
In theory, SIP could provide end-to-end media security. In practice, no =
one figured it out. The good news is RIPP is no worse than SIP as it is =
deployed today. Moreover, RIPP is no worse than HTTP. My concern is RIPP =
structurally will never have end-to-end media security. While that is no =
worse than today, will we be running into some privacy expectations of =
users and regulators? For example, a RIPP server can record everything. =
Problems here are (1) in the U.S. that is likely to be a violation of =
the Wiretap Act and (2) people can use those recordings for very =
nefarious purposes (see RealTalk, Lyrebird, etc.).

Section 8.2:
Just curious: why MUST NOT the identifier included the authority =
component for uniqueness? Is that a Web best practice? A reference or =
explanation would be helpful.

Section 8.3:
What are the capabilities *of*? Are they codec parameters? Are they =
decoded stream parameters? Are they codec dependent? Is there any value =
to these parameters, outside parameterized codec negotiation that I am =
missing?

For example, is the max-bitrate the decoded maximum bitrate or the =
bitrate of the codec? What is the purpose?
If I come up with a cool LPC that can do better than G.711 with 4000 =
samples per section, why cannot I use it?

BIG ONE: force-cbr
Do we require a CBR codec or do we require uniform packet sizes (in =
bytes)? If the former, why not just ask for a CBR codec in the request. =
If the latter, change the parameter to something like =E2=80=9Crequire =
privacy.=E2=80=9D There are much better ways of achieving privacy than =
using a CBR codec; see e.g., =
https://repository.library.georgetown.edu/handle/10822/1040699 =
<https://repository.library.georgetown.edu/handle/10822/1040699>

Section 8.4:
I agree with Cullen. It goes against the design principle of using =
HTTP/3 for transport and then start putting application semantics into =
the transport layer. Is there a reason to break layers, e.g., for load =
balancing or something else?

Section 8.7
Why 20 for forward and 10 for reverse? Are these rules of thumb, =
calculated optimal places to start, SWAGs, or irrelevant?
Also (and in Section 8.8), what about the speed of light and the =
appearance of blocking?

Section 8.8
Perhaps I missed it, but what happens if a byway appears to be blocked =
for a =E2=80=98while=E2=80=99? Since this is voice and this is UDP and I =
am tolerant of packet loss, do I care a packet was lost? Also, I=E2=80=99m=
 highly skeptical of 1:1 packet:ack this protocol offers. That sounds =
like a bad thing for LANs and a terrible thing for WANs. Really?

Section 8.10
I kind of agree with Cullen=E2=80=99s point, but here is an opposite =
observation that I came up with before I saw Cullen=E2=80=99s point =
(that SIP result codes don=E2=80=99t cover ISDN). Why have a separate =
=E2=80=98rejected=E2=80=99 result in light of =E2=80=98failed.=E2=80=99 =
Unless rejected is referencing the SIP 607 result code, I would think =
all the call center cares about is whether the call completed and if it =
did not, will a retry be likely to succeed or fail.


There are a ton of typos in the draft. I=E2=80=99d be happy to hand my =
mark-up to whomever wants it in DISPATCH.

--Apple-Mail=_A892FA49-03ED-4103-B980-7A6A6EE9EF20
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	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><div =
class=3D"">General: For the most part, I like the idea of a draft that =
basically says, =E2=80=9Cwe tried to be pure but failed. This proposal =
is not pure but it for the most part represents running code.=E2=80=9D =
PHB will be happy (ref. the NAT discussion on the IETF =
list).&nbsp;</div><div class=3D""><br class=3D""></div><div class=3D""><br=
 class=3D""></div><div class=3D"">Section 1.1: background<div =
class=3D"">Will much of it be meaningful to a protocol implementer in a =
year, no less five? I.e., referencing Kubernetes, Istio, CanaryDeploys, =
etc., more especially without going into detail as to what they are? =
Sure, if you are an AWS developer you know them like the back of your =
hand. However, not many folks in IETF-land are.</div><div class=3D""><br =
class=3D""></div><div class=3D"">1.2:</div><div class=3D"">Why is it =
=E2=80=98unfortunate=E2=80=99 that people want to deploy cloud =
applications that interact with the PSTN? Is that not an opportunity? I =
would take out the initial =E2=80=9CUnfortunately.=E2=80=9D It=E2=80=99s =
just a statement of fact that applications like to call and be called to =
most any person on the planet on a fully interoperable basis.</div><div =
class=3D""><br class=3D""></div><div class=3D"">There needs to be more =
justification than =E2=80=9CWeb people don=E2=80=99t understand SIP.=E2=80=
=9D You might want to explain how RIPP is different than ParlayX. I =
would offer that RIPP:ParlayX::LDAP:X.500, but you don=E2=80=99t say =
that anywhere. All the draft implies is Web people cannot grok SIP and =
SIP does not grok Web deployment, which does not reflect well on either =
community.</div><div class=3D""><br class=3D""></div><div class=3D"">The =
last paragraph of 1.2 implies that SIP is the source of robocalls and a =
mechanism that will be even easier to inject boatloads of calls will be =
better. Please explain how this will be so (or remove the assertion). As =
you know, I personally wish it would be so, so PLEASE give me =
ammunition!</div><div class=3D""><br class=3D""></div><div =
class=3D"">1.4:</div><div class=3D"">Given the virtual mandate to use =
SIP over TCP, is saying =E2=80=9Cwe can finally use HTTP over UDP=E2=80=9D=
 not kind of a retrograde assertion?</div><div class=3D""><br =
class=3D""></div><div class=3D"">REQ 12 (supporting only audio) is an =
odd requirement. Moreover, most applications in the U.S. would require =
at least RTT support if not video. I am not a lawyer, but since we are =
talking interoperability with the PSTN, one is likely to run into =
Section 255 and 706 issues if RTT (and video) is not there from day =
one.</div><div class=3D""><br class=3D""></div><div class=3D"">REQ 13 =
looks totally duplicative of REQ 10 (use secure caller ID technologies). =
Is there a distinction? If so, please make it clearer</div></div><div =
class=3D""><br class=3D""></div><div class=3D"">Section 3.6:</div><div =
class=3D"">This is a place where we need to distinguish between theory =
and reality. In theory, SIP could provide end-to-end media security. In =
practice, no one figured it out. The good news is RIPP is no worse than =
SIP as it is deployed today. Moreover, RIPP is no worse than HTTP. My =
concern is RIPP structurally will never have end-to-end media security. =
While that is no worse than today, will we be running into some privacy =
expectations of users and regulators? For example, a RIPP server can =
record everything. Problems here are (1) in the U.S. that is likely to =
be a violation of the Wiretap Act and (2) people can use those =
recordings for very nefarious purposes (see RealTalk, Lyrebird, =
etc.).</div><div class=3D""><br class=3D""></div><div class=3D"">Section =
8.2:</div><div class=3D"">Just curious: why MUST NOT the identifier =
included the authority component for uniqueness? Is that a Web best =
practice? A reference or explanation would be helpful.</div><div =
class=3D""><br class=3D""></div><div class=3D"">Section 8.3:</div><div =
class=3D"">What are the capabilities *of*? Are they codec parameters? =
Are they decoded stream parameters? Are they codec dependent? Is there =
any value to these parameters, outside parameterized codec negotiation =
that I am missing?</div><div class=3D""><br class=3D""></div><div =
class=3D"">For example, is the max-bitrate the decoded maximum bitrate =
or the bitrate of the codec? What is the purpose?</div><div class=3D"">If =
I come up with a cool LPC that can do better than G.711 with 4000 =
samples per section, why cannot I use it?</div><div class=3D""><br =
class=3D""></div><div class=3D"">BIG ONE: force-cbr</div><div =
class=3D"">Do we require a CBR codec or do we require uniform packet =
sizes (in bytes)? If the former, why not just ask for a CBR codec in the =
request. If the latter, change the parameter to something like =
=E2=80=9Crequire privacy.=E2=80=9D There are much better ways of =
achieving privacy than using a CBR codec; see e.g.,&nbsp;<a =
href=3D"https://repository.library.georgetown.edu/handle/10822/1040699" =
class=3D"">https://repository.library.georgetown.edu/handle/10822/1040699<=
/a></div><div class=3D""><br class=3D""></div><div class=3D"">Section =
8.4:</div><div class=3D"">I agree with Cullen. It goes against the =
design principle of using HTTP/3 for transport and then start putting =
application semantics into the transport layer. Is there a reason to =
break layers, e.g., for load balancing or something else?</div><div =
class=3D""><br class=3D""></div><div class=3D"">Section 8.7</div><div =
class=3D"">Why 20 for forward and 10 for reverse? Are these rules of =
thumb, calculated optimal places to start, SWAGs, or =
irrelevant?</div><div class=3D"">Also (and in Section 8.8), what about =
the speed of light and the appearance of blocking?</div><div =
class=3D""><br class=3D""></div><div class=3D"">Section 8.8</div><div =
class=3D"">Perhaps I missed it, but what happens if a byway appears to =
be blocked for a =E2=80=98while=E2=80=99? Since this is voice and this =
is UDP and I am tolerant of packet loss, do I care a packet was lost? =
Also, I=E2=80=99m highly skeptical of 1:1 packet:ack this protocol =
offers. That sounds like a bad thing for LANs and a terrible thing for =
WANs. Really?</div><div class=3D""><br class=3D""></div><div =
class=3D"">Section 8.10</div><div class=3D"">I kind of agree with =
Cullen=E2=80=99s point, but here is an opposite observation that I came =
up with before I saw Cullen=E2=80=99s point (that SIP result codes =
don=E2=80=99t cover ISDN). Why have a separate =E2=80=98rejected=E2=80=99 =
result in light of =E2=80=98failed.=E2=80=99 Unless rejected is =
referencing the SIP 607 result code, I would think all the call center =
cares about is whether the call completed and if it did not, will a =
retry be likely to succeed or fail.</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D"">There are a ton of typos in the draft. I=E2=80=99d be happy =
to hand my mark-up to whomever wants it in DISPATCH.</div></body></html>=

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From nobody Sun Jul 21 19:21:55 2019
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Date: Sun, 21 Jul 2019 22:21:45 -0400
Cc: dispatch chairs <dispatch-chairs@ietf.org>
To: DISPATCH <dispatch@ietf.org>
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Subject: [dispatch] Volunteers
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Hi Everyone,

We will have a tight schedule in Monday=E2=80=99s DISPATCH meeting. =
I=E2=80=99d like to get one of our administrative tasks done prior to =
the meeting, namely getting volunteers to take meeting notes and act as =
Jabber relay.

We need one or preferably two note-takers. This doesn=E2=80=99t have to =
be a play-by-play (unless you like doing that); we mainly need a record =
of decisions, actions items, or similar things of note.

We also need a Jabber relay. This mainly means taking comments from the =
jabber room to the mike if requested. Remote participants are encouraged =
to use MeetEcho to make comments, but we still sometimes get comments =
from jabber.

Many thanks in advance to our most honored volunteers!

Ben.

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To: Eric Burger <eburger@standardstrack.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Thread-Topic: [dispatch] Thoughts on draft-rosenbergjennings-dispatch-ripp-03
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From: Jonathan Rosenberg <jdrosen@jdrosen.net>
Date: Mon, 22 Jul 2019 08:13:56 -0400
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To: "Roni Even (A)" <roni.even@huawei.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/dispatch/ZYLDIhVkCPbCvF8El_xDhtXhow4>
Subject: Re: [dispatch] Thoughts on draft-rosenbergjennings-dispatch-ripp-03
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Thanks Roni for the comments!

Though quic overall is reliable, the way we=E2=80=99re using it achieves an
interesting effect - there is no hol blocking. This means an actual IP loss
would result in packet 1 showing up, then 3, then much later, 2. As a
result, with hitter buffer, 2 is effectively lost and thus fec would be
useful in recovering 2.

Thx,
Jonathan R.



On Mon, Jul 22, 2019 at 7:16 AM Roni Even (A) <roni.even@huawei.com> wrote:

> I agree with Eric=E2=80=99s comments.
>
> I was also wondering about the capabilities support. For example OPUS
> supports in band FEC which addresses packet loss but if you use a reliabl=
e
> transport it will lose the effectiveness of FEC.  This is why there is so=
me
> discussion about unreliable streams.
>
> It look to me that replacing RTP is not as simple as you present in the
> document.
>
> Roni Even
>
>
>
> *From:* dispatch [mailto:dispatch-bounces@ietf.org] *On Behalf Of *Eric
> Burger
> *Sent:* Monday, July 22, 2019 12:18 AM
> *To:* dispatch@ietf.org
> *Subject:* [dispatch] Thoughts on draft-rosenbergjennings-dispatch-ripp-0=
3
>
>
>
> General: For the most part, I like the idea of a draft that basically
> says, =E2=80=9Cwe tried to be pure but failed. This proposal is not pure =
but it for
> the most part represents running code.=E2=80=9D PHB will be happy (ref. t=
he NAT
> discussion on the IETF list).
>
>
>
>
>
> Section 1.1: background
>
> Will much of it be meaningful to a protocol implementer in a year, no les=
s
> five? I.e., referencing Kubernetes, Istio, CanaryDeploys, etc., more
> especially without going into detail as to what they are? Sure, if you ar=
e
> an AWS developer you know them like the back of your hand. However, not
> many folks in IETF-land are.
>
>
>
> 1.2:
>
> Why is it =E2=80=98unfortunate=E2=80=99 that people want to deploy cloud =
applications that
> interact with the PSTN? Is that not an opportunity? I would take out the
> initial =E2=80=9CUnfortunately.=E2=80=9D It=E2=80=99s just a statement of=
 fact that applications
> like to call and be called to most any person on the planet on a fully
> interoperable basis.
>
>
>
> There needs to be more justification than =E2=80=9CWeb people don=E2=80=
=99t understand
> SIP.=E2=80=9D You might want to explain how RIPP is different than Parlay=
X. I would
> offer that RIPP:ParlayX::LDAP:X.500, but you don=E2=80=99t say that anywh=
ere. All
> the draft implies is Web people cannot grok SIP and SIP does not grok Web
> deployment, which does not reflect well on either community.
>
>
>
> The last paragraph of 1.2 implies that SIP is the source of robocalls and
> a mechanism that will be even easier to inject boatloads of calls will be
> better. Please explain how this will be so (or remove the assertion). As
> you know, I personally wish it would be so, so PLEASE give me ammunition!
>
>
>
> 1.4:
>
> Given the virtual mandate to use SIP over TCP, is saying =E2=80=9Cwe can =
finally
> use HTTP over UDP=E2=80=9D not kind of a retrograde assertion?
>
>
>
> REQ 12 (supporting only audio) is an odd requirement. Moreover, most
> applications in the U.S. would require at least RTT support if not video.=
 I
> am not a lawyer, but since we are talking interoperability with the PSTN,
> one is likely to run into Section 255 and 706 issues if RTT (and video) i=
s
> not there from day one.
>
>
>
> REQ 13 looks totally duplicative of REQ 10 (use secure caller ID
> technologies). Is there a distinction? If so, please make it clearer
>
>
>
> Section 3.6:
>
> This is a place where we need to distinguish between theory and reality.
> In theory, SIP could provide end-to-end media security. In practice, no o=
ne
> figured it out. The good news is RIPP is no worse than SIP as it is
> deployed today. Moreover, RIPP is no worse than HTTP. My concern is RIPP
> structurally will never have end-to-end media security. While that is no
> worse than today, will we be running into some privacy expectations of
> users and regulators? For example, a RIPP server can record everything.
> Problems here are (1) in the U.S. that is likely to be a violation of the
> Wiretap Act and (2) people can use those recordings for very nefarious
> purposes (see RealTalk, Lyrebird, etc.).
>
>
>
> Section 8.2:
>
> Just curious: why MUST NOT the identifier included the authority componen=
t
> for uniqueness? Is that a Web best practice? A reference or explanation
> would be helpful.
>
>
>
> Section 8.3:
>
> What are the capabilities *of*? Are they codec parameters? Are they
> decoded stream parameters? Are they codec dependent? Is there any value t=
o
> these parameters, outside parameterized codec negotiation that I am missi=
ng?
>
>
>
> For example, is the max-bitrate the decoded maximum bitrate or the bitrat=
e
> of the codec? What is the purpose?
>
> If I come up with a cool LPC that can do better than G.711 with 4000
> samples per section, why cannot I use it?
>
>
>
> BIG ONE: force-cbr
>
> Do we require a CBR codec or do we require uniform packet sizes (in
> bytes)? If the former, why not just ask for a CBR codec in the request. I=
f
> the latter, change the parameter to something like =E2=80=9Crequire priva=
cy.=E2=80=9D There
> are much better ways of achieving privacy than using a CBR codec; see e.g=
.,
> https://repository.library.georgetown.edu/handle/10822/1040699
>
>
>
> Section 8.4:
>
> I agree with Cullen. It goes against the design principle of using HTTP/3
> for transport and then start putting application semantics into the
> transport layer. Is there a reason to break layers, e.g., for load
> balancing or something else?
>
>
>
> Section 8.7
>
> Why 20 for forward and 10 for reverse? Are these rules of thumb,
> calculated optimal places to start, SWAGs, or irrelevant?
>
> Also (and in Section 8.8), what about the speed of light and the
> appearance of blocking?
>
>
>
> Section 8.8
>
> Perhaps I missed it, but what happens if a byway appears to be blocked fo=
r
> a =E2=80=98while=E2=80=99? Since this is voice and this is UDP and I am t=
olerant of packet
> loss, do I care a packet was lost? Also, I=E2=80=99m highly skeptical of =
1:1
> packet:ack this protocol offers. That sounds like a bad thing for LANs an=
d
> a terrible thing for WANs. Really?
>
>
>
> Section 8.10
>
> I kind of agree with Cullen=E2=80=99s point, but here is an opposite obse=
rvation
> that I came up with before I saw Cullen=E2=80=99s point (that SIP result =
codes
> don=E2=80=99t cover ISDN). Why have a separate =E2=80=98rejected=E2=80=99=
 result in light of
> =E2=80=98failed.=E2=80=99 Unless rejected is referencing the SIP 607 resu=
lt code, I would
> think all the call center cares about is whether the call completed and i=
f
> it did not, will a retry be likely to succeed or fail.
>
>
>
>
>
> There are a ton of typos in the draft. I=E2=80=99d be happy to hand my ma=
rk-up to
> whomever wants it in DISPATCH.
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
--=20
Jonathan Rosenberg, Ph.D.
jdrosen@jdrosen.net
http://www.jdrosen.net

--00000000000046a8b7058e4403ef
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div><div dir=3D"auto">Thanks Roni for the comments!</div></div><div dir=3D=
"auto"><br></div><div dir=3D"auto">Though quic overall is reliable, the way=
 we=E2=80=99re using it achieves an interesting effect - there is no hol bl=
ocking. This means an actual IP loss would result in packet 1 showing up, t=
hen 3, then much later, 2. As a result, with hitter buffer, 2 is effectivel=
y lost and thus fec would be useful in recovering 2.=C2=A0</div><div dir=3D=
"auto"><br></div><div dir=3D"auto">Thx,</div><div dir=3D"auto">Jonathan R.<=
/div><div dir=3D"auto"><br></div><div dir=3D"auto"><br></div><div><br><div =
class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Jul 22,=
 2019 at 7:16 AM Roni Even (A) &lt;<a href=3D"mailto:roni.even@huawei.com">=
roni.even@huawei.com</a>&gt; wrote:<br></div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_635509396196840167WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">I agree with Eric=E2=80=
=99s comments.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">I was also wondering abou=
t the capabilities support. For example OPUS supports in band FEC which add=
resses packet loss but if you use a reliable transport it
 will lose the effectiveness of FEC.=C2=A0 This is why there is some discus=
sion about unreliable streams.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">It look to me that replac=
ing RTP is not as simple as you present in the document.</span></p></div></=
div><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"m_6355=
09396196840167WordSection1"><p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Roni Even<u></u><u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> dispatch=
 [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org" target=3D"_blank">dis=
patch-bounces@ietf.org</a>]
<b>On Behalf Of </b>Eric Burger<br>
<b>Sent:</b> Monday, July 22, 2019 12:18 AM<br>
<b>To:</b> <a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@=
ietf.org</a><br>
<b>Subject:</b> [dispatch] Thoughts on draft-rosenbergjennings-dispatch-rip=
p-03<u></u><u></u></span></p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">General: For the most part, I like the idea of a dra=
ft that basically says, =E2=80=9Cwe tried to be pure but failed. This propo=
sal is not pure but it for the most part represents running code.=E2=80=9D =
PHB will be happy (ref. the NAT discussion on the
 IETF list).=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 1.1: background<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">Will much of it be meaningful to a protocol implemen=
ter in a year, no less five? I.e., referencing Kubernetes, Istio, CanaryDep=
loys, etc., more especially without going into detail as to what they are? =
Sure, if you are an AWS developer
 you know them like the back of your hand. However, not many folks in IETF-=
land are.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">1.2:<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Why is it =E2=80=98unfortunate=E2=80=99 that people =
want to deploy cloud applications that interact with the PSTN? Is that not =
an opportunity? I would take out the initial =E2=80=9CUnfortunately.=E2=80=
=9D It=E2=80=99s just a statement of fact that applications like to call an=
d
 be called to most any person on the planet on a fully interoperable basis.=
<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">There needs to be more justification than =E2=80=9CW=
eb people don=E2=80=99t understand SIP.=E2=80=9D You might want to explain =
how RIPP is different than ParlayX. I would offer that RIPP:ParlayX::LDAP:X=
.500, but you don=E2=80=99t say that anywhere. All the draft implies
 is Web people cannot grok SIP and SIP does not grok Web deployment, which =
does not reflect well on either community.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The last paragraph of 1.2 implies that SIP is the so=
urce of robocalls and a mechanism that will be even easier to inject boatlo=
ads of calls will be better. Please explain how this will be so (or remove =
the assertion). As you know, I personally
 wish it would be so, so PLEASE give me ammunition!<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">1.4:<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Given the virtual mandate to use SIP over TCP, is sa=
ying =E2=80=9Cwe can finally use HTTP over UDP=E2=80=9D not kind of a retro=
grade assertion?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">REQ 12 (supporting only audio) is an odd requirement=
. Moreover, most applications in the U.S. would require at least RTT suppor=
t if not video. I am not a lawyer, but since we are talking interoperabilit=
y with the PSTN, one is likely to
 run into Section 255 and 706 issues if RTT (and video) is not there from d=
ay one.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">REQ 13 looks totally duplicative of REQ 10 (use secu=
re caller ID technologies). Is there a distinction? If so, please make it c=
learer<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 3.6:<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">This is a place where we need to distinguish between=
 theory and reality. In theory, SIP could provide end-to-end media security=
. In practice, no one figured it out. The good news is RIPP is no worse tha=
n SIP as it is deployed today. Moreover,
 RIPP is no worse than HTTP. My concern is RIPP structurally will never hav=
e end-to-end media security. While that is no worse than today, will we be =
running into some privacy expectations of users and regulators? For example=
, a RIPP server can record everything.
 Problems here are (1) in the U.S. that is likely to be a violation of the =
Wiretap Act and (2) people can use those recordings for very nefarious purp=
oses (see RealTalk, Lyrebird, etc.).<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 8.2:<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Just curious: why MUST NOT the identifier included t=
he authority component for uniqueness? Is that a Web best practice? A refer=
ence or explanation would be helpful.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 8.3:<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">What are the capabilities *of*? Are they codec param=
eters? Are they decoded stream parameters? Are they codec dependent? Is the=
re any value to these parameters, outside parameterized codec negotiation t=
hat I am missing?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">For example, is the max-bitrate the decoded maximum =
bitrate or the bitrate of the codec? What is the purpose?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">If I come up with a cool LPC that can do better than=
 G.711 with 4000 samples per section, why cannot I use it?<u></u><u></u></p=
>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">BIG ONE: force-cbr<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Do we require a CBR codec or do we require uniform p=
acket sizes (in bytes)? If the former, why not just ask for a CBR codec in =
the request. If the latter, change the parameter to something like =E2=80=
=9Crequire privacy.=E2=80=9D There are much better ways
 of achieving privacy than using a CBR codec; see e.g.,=C2=A0<a href=3D"htt=
ps://repository.library.georgetown.edu/handle/10822/1040699" target=3D"_bla=
nk">https://repository.library.georgetown.edu/handle/10822/1040699</a><u></=
u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 8.4:<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I agree with Cullen. It goes against the design prin=
ciple of using HTTP/3 for transport and then start putting application sema=
ntics into the transport layer. Is there a reason to break layers, e.g., fo=
r load balancing or something else?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 8.7<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Why 20 for forward and 10 for reverse? Are these rul=
es of thumb, calculated optimal places to start, SWAGs, or irrelevant?<u></=
u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Also (and in Section 8.8), what about the speed of l=
ight and the appearance of blocking?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 8.8<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Perhaps I missed it, but what happens if a byway app=
ears to be blocked for a =E2=80=98while=E2=80=99? Since this is voice and t=
his is UDP and I am tolerant of packet loss, do I care a packet was lost? A=
lso, I=E2=80=99m highly skeptical of 1:1 packet:ack this protocol
 offers. That sounds like a bad thing for LANs and a terrible thing for WAN=
s. Really?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Section 8.10<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I kind of agree with Cullen=E2=80=99s point, but her=
e is an opposite observation that I came up with before I saw Cullen=E2=80=
=99s point (that SIP result codes don=E2=80=99t cover ISDN). Why have a sep=
arate =E2=80=98rejected=E2=80=99 result in light of =E2=80=98failed.=E2=80=
=99 Unless rejected
 is referencing the SIP 607 result code, I would think all the call center =
cares about is whether the call completed and if it did not, will a retry b=
e likely to succeed or fail.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">There are a ton of typos in the draft. I=E2=80=99d b=
e happy to hand my mark-up to whomever wants it in DISPATCH.<u></u><u></u><=
/p>
</div>
</div>
</div>

_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"noreferre=
r" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div></div>-- <br><div dir=3D"ltr" class=3D"gmail_signature" =
data-smartmail=3D"gmail_signature"><div dir=3D"ltr">Jonathan Rosenberg, Ph.=
D.<br><a href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdro=
sen.net</a><br><a href=3D"http://www.jdrosen.net" target=3D"_blank">http://=
www.jdrosen.net</a></div></div>

--00000000000046a8b7058e4403ef--


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S=E2=80=99il vous pla=C3=AEt ?

> On Jul 21, 2019, at 10:21 PM, Ben Campbell <ben@nostrum.com> wrote:
>=20
> Signed PGP part
> Hi Everyone,
>=20
> We will have a tight schedule in Monday=E2=80=99s DISPATCH meeting. =
I=E2=80=99d like to get one of our administrative tasks done prior to =
the meeting, namely getting volunteers to take meeting notes and act as =
Jabber relay.
>=20
> We need one or preferably two note-takers. This doesn=E2=80=99t have =
to be a play-by-play (unless you like doing that); we mainly need a =
record of decisions, actions items, or similar things of note.
>=20
> We also need a Jabber relay. This mainly means taking comments from =
the jabber room to the mike if requested. Remote participants are =
encouraged to use MeetEcho to make comments, but we still sometimes get =
comments from jabber.
>=20
> Many thanks in advance to our most honored volunteers!
>=20
> Ben.
>=20
>=20


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Update: We have one note taker and a jabber scribe. We still have an =
opening for one more note taker. These opportunities are going fast, so =
don=E2=80=99t wait to volunteer!

Thanks!

Ben.

> On Jul 22, 2019, at 10:14 AM, Ben Campbell <ben@nostrum.com> wrote:
>=20
> Signed PGP part
> S=E2=80=99il vous pla=C3=AEt ?
>=20
>> On Jul 21, 2019, at 10:21 PM, Ben Campbell <ben@nostrum.com> wrote:
>>=20
>> Signed PGP part
>> Hi Everyone,
>>=20
>> We will have a tight schedule in Monday=E2=80=99s DISPATCH meeting. =
I=E2=80=99d like to get one of our administrative tasks done prior to =
the meeting, namely getting volunteers to take meeting notes and act as =
Jabber relay.
>>=20
>> We need one or preferably two note-takers. This doesn=E2=80=99t have =
to be a play-by-play (unless you like doing that); we mainly need a =
record of decisions, actions items, or similar things of note.
>>=20
>> We also need a Jabber relay. This mainly means taking comments from =
the jabber room to the mike if requested. Remote participants are =
encouraged to use MeetEcho to make comments, but we still sometimes get =
comments from jabber.
>>=20
>> Many thanks in advance to our most honored volunteers!
>>=20
>> Ben.
>>=20
>>=20
>=20
>=20


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Subject: Re: [dispatch] Thoughts on draft-rosenbergjennings-dispatch-ripp-03
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Hi Eric - thanks for taking the time to read and for the thoughtful
comments.

Responses inline below - but as a high level response - right now we're
less interested in the details of the spec (for which the authors
themselves dont all agree...), but rather to see if there is interest in
solving this problem. The co-authors will be working on prototyping from
here, so if anyone is interested, let us know... goal is working code
before next ietf.


On Sun, Jul 21, 2019 at 6:24 PM Eric Burger <eburger@standardstrack.com>
wrote:

> General: For the most part, I like the idea of a draft that basically
> says, =E2=80=9Cwe tried to be pure but failed. This proposal is not pure =
but it for
> the most part represents running code.=E2=80=9D PHB will be happy (ref. t=
he NAT
> discussion on the IETF list).
>

Yes, this is very much a pragmatic spec, aimed at solving some very
real-world problems (ie., i've personally been beating my head against the
wall trying to usefully deploy freeswitch within Istio on GKE. Let me know
if you figured it out - its all hard due to how different http is from sip.=
)



>
>
> Section 1.1: background
> Will much of it be meaningful to a protocol implementer in a year, no les=
s
> five? I.e., referencing Kubernetes, Istio, CanaryDeploys, etc., more
> especially without going into detail as to what they are? Sure, if you ar=
e
> an AWS developer you know them like the back of your hand. However, not
> many folks in IETF-land are.
>

Happy to remove later on. Right now its just looknig to motivate the
problem.



>
> 1.2:
> Why is it =E2=80=98unfortunate=E2=80=99 that people want to deploy cloud =
applications that
> interact with the PSTN? Is that not an opportunity? I would take out the
> initial =E2=80=9CUnfortunately.=E2=80=9D It=E2=80=99s just a statement of=
 fact that applications
> like to call and be called to most any person on the planet on a fully
> interoperable basis.
>

Stray word from a few rounds of edits, will remove thx.


>
> There needs to be more justification than =E2=80=9CWeb people don=E2=80=
=99t understand
> SIP.=E2=80=9D You might want to explain how RIPP is different than Parlay=
X. I would
> offer that RIPP:ParlayX::LDAP:X.500, but you don=E2=80=99t say that anywh=
ere. All
> the draft implies is Web people cannot grok SIP and SIP does not grok Web
> deployment, which does not reflect well on either community.
>

No - its not about how well people understand anything. Its about how
applicable modern cloud paas is to SIP and real-time services (which it
really isnt), and how easy it is to build real-time apps which follow
established design patterns for scalable and reliable systems (its super
hard if not impossible).


>
> The last paragraph of 1.2 implies that SIP is the source of robocalls and
> a mechanism that will be even easier to inject boatloads of calls will be
> better. Please explain how this will be so (or remove the assertion). As
> you know, I personally wish it would be so, so PLEASE give me ammunition!
>

It mandates passport as the one and only way to even know the caller ID.
COnsequently, the protocol drags stir along for the ride. There is no other
special magic or anything.


>
> 1.4:
> Given the virtual mandate to use SIP over TCP, is saying =E2=80=9Cwe can =
finally
> use HTTP over UDP=E2=80=9D not kind of a retrograde assertion?
>

I wasn't being specific enough here. "SIP" refers to the whole community of
specs needed for it, and this specific sentence is referring to RTP. I'll
change to RTP.



>
> REQ 12 (supporting only audio) is an odd requirement. Moreover, most
> applications in the U.S. would require at least RTT support if not video.=
 I
> am not a lawyer, but since we are talking interoperability with the PSTN,
> one is likely to run into Section 255 and 706 issues if RTT (and video) i=
s
> not there from day one.
>

I dont object - was just trying to start simple and solve one specific
problem well. Do current SIP trunking providers support RTT and video?



>
> REQ 13 looks totally duplicative of REQ 10 (use secure caller ID
> technologies). Is there a distinction? If so, please make it clearer
>

Bug - thx.


>
> Section 3.6:
> This is a place where we need to distinguish between theory and reality.
> In theory, SIP could provide end-to-end media security. In practice, no o=
ne
> figured it out. The good news is RIPP is no worse than SIP as it is
> deployed today. Moreover, RIPP is no worse than HTTP. My concern is RIPP
> structurally will never have end-to-end media security. While that is no
> worse than today, will we be running into some privacy expectations of
> users and regulators? For example, a RIPP server can record everything.
> Problems here are (1) in the U.S. that is likely to be a violation of the
> Wiretap Act and (2) people can use those recordings for very nefarious
> purposes (see RealTalk, Lyrebird, etc.).
>

I would assert that RIPP would be better than SIP today. There is a lot of
SIP trunking out there (note RIPP *only* addresses inter-domain trunking
cases, so thats the point of comparison). Most SIP trunking interconnection
is over private IP with no TLS or SRTP. There is also some over public
Internet, I suspect much of it is without SRTP and probably runs over VPN
often. So, a peering protocol which requires hop encryption of signaling
and media, and actually authenticates both sides, is a big improvement over
the current situation.

E2E encryption isnt really relevant for the applicability of RIPP, since it
is targeted at inter-domain connectivity towards PSTN. PSTN as you well
know has no e2e encryption, and there must be decryption at the pstn
gateway.

I dont follow how having a server at the edge of a provider network might
be in violation of the wiretap act, or somehow make recordings worse. THis
is no different than the situation today with SBCs, where the vast majority
have no SRTP at all, and even those that do, are decrypting so they have
access to the media.




>
> Section 8.2:
> Just curious: why MUST NOT the identifier included the authority componen=
t
> for uniqueness? Is that a Web best practice? A reference or explanation
> would be helpful.
>

THis is the text:
"The path component MUST

   be a globally unique identifier for this trunk, and not depend on the
   authority component as part of the namespace for purposes of
   uniqueness."


The main motivation is to eliminate the need for wildcard certs to authoriz=
e:

https://12345abcdef.ripp-provider.com requires one, whereas:

https://ripp-provider.com/12345abcdef does not


I'll include a mention on that, thanks.





>
> Section 8.3:
> What are the capabilities *of*? Are they codec parameters? Are they
> decoded stream parameters? Are they codec dependent? Is there any value t=
o
> these parameters, outside parameterized codec negotiation that I am missi=
ng?
>

THese are the capabilities of the ripp trunk. These are things normally
configured as capabilities/configuration of sip trunks today. Codecs,
bitrates, etc. They avoid the need for per-call capabilities exchange.
Practically speaking, such exchanges are un-needed since the capabilities
are static properties of the configured sip trunks. Again, we're not
replacing sip overall - just providing a solution for inter-domain peering
for voip to the pstn.



>
> For example, is the max-bitrate the decoded maximum bitrate or the bitrat=
e
> of the codec? What is the purpose?
> If I come up with a cool LPC that can do better than G.711 with 4000
> samples per section, why cannot I use it?
>

No this is the max codec bitrate, same as current RTP definition. Meant
more for like opus.
The protocol of course is extensible to new codecs - each side advertises
the codecs they support and their associated max rates. If someone however
says they dont want to receive 4000 samples per second, they'd advertise a
lower upper bound. This part is really no different than offer/answer.


>
> BIG ONE: force-cbr
> Do we require a CBR codec or do we require uniform packet sizes (in
> bytes)? If the former, why not just ask for a CBR codec in the request. I=
f
> the latter, change the parameter to something like =E2=80=9Crequire priva=
cy.=E2=80=9D There
> are much better ways of achieving privacy than using a CBR codec; see e.g=
.,
> https://repository.library.georgetown.edu/handle/10822/1040699
>

Cullen was the one who wanted this one - so he's better positioned to speak
to it. There was a study that found you could actually figure out what was
being said just by looking at patterns of lengths of variable bitrate audio
codecs.


>
> Section 8.4:
> I agree with Cullen. It goes against the design principle of using HTTP/3
> for transport and then start putting application semantics into the
> transport layer. Is there a reason to break layers, e.g., for load
> balancing or something else?
>

Adding http headers is very common for web apps, so this isnt really
different from normal web practice. That said, i dont feel that strongly on
this one. Per above, right now its not so much about the specicic details
of this proposal, as it is a starting point strawman for working on the
problem.



>
> Section 8.7
> Why 20 for forward and 10 for reverse? Are these rules of thumb,
> calculated optimal places to start, SWAGs, or irrelevant?
> Also (and in Section 8.8), what about the speed of light and the
> appearance of blocking?
>

TOtal SWAGs right now - we need some implementation experience to make this
a more usable spec.

Not sure I follow what you mean by speed of light?


>
> Section 8.8
> Perhaps I missed it, but what happens if a byway appears to be blocked fo=
r
> a =E2=80=98while=E2=80=99? Since this is voice and this is UDP and I am t=
olerant of packet
> loss, do I care a packet was lost? Also, I=E2=80=99m highly skeptical of =
1:1
> packet:ack this protocol offers. That sounds like a bad thing for LANs an=
d
> a terrible thing for WANs. Really?
>

We didnt spec out yet what happens if its blocked for a while. This would
happen under significant packet loss, so things are really bad. Probably
drop the call.

Do you care if a packet is lost? in normal voip cases, no. There are some
use cases where the retransmission would be handy (same reason we have an
RTP extension for it..)

On the 1-1 packet:ack - thats a good comment. Definitely a drawback. Would
be better if its piggybacked on reverse media, but we also currently have
reverse media in separate byways. SO thats a tradeoff to be explored.



>
> Section 8.10
> I kind of agree with Cullen=E2=80=99s point, but here is an opposite obse=
rvation
> that I came up with before I saw Cullen=E2=80=99s point (that SIP result =
codes
> don=E2=80=99t cover ISDN). Why have a separate =E2=80=98rejected=E2=80=99=
 result in light of
> =E2=80=98failed.=E2=80=99 Unless rejected is referencing the SIP 607 resu=
lt code, I would
> think all the call center cares about is whether the call completed and i=
f
> it did not, will a retry be likely to succeed or fail.
>


Rejected is meant to handle the case, like mobile, where the end user
actually hits the decline button.



>
>
> There are a ton of typos in the draft. I=E2=80=99d be happy to hand my ma=
rk-up to
> whomever wants it in DISPATCH.
>

Thanks! Probably not worth fixing till we've gotten more experience under
our belt and revised it to reflect that.




> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


--=20
Jonathan Rosenberg, Ph.D.
jdrosen@jdrosen.net
http://www.jdrosen.net

--00000000000085bc37058e47e3a4
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hi Eric - thanks for taking the time to read and for =
the thoughtful comments.</div><div><br></div><div>Responses inline below - =
but as a high level response - right now we&#39;re less interested in the d=
etails of the spec (for which the authors themselves dont all agree...), bu=
t rather to see if there is interest in solving this problem. The co-author=
s will be working on prototyping from here, so if anyone is interested, let=
 us know... goal is working code before next ietf.</div><div><br></div><br>=
<div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Sun, Ju=
l 21, 2019 at 6:24 PM Eric Burger &lt;<a href=3D"mailto:eburger@standardstr=
ack.com">eburger@standardstrack.com</a>&gt; wrote:<br></div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid =
rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;=
"><div>General: For the most part, I like the idea of a draft that basicall=
y says, =E2=80=9Cwe tried to be pure but failed. This proposal is not pure =
but it for the most part represents running code.=E2=80=9D PHB will be happ=
y (ref. the NAT discussion on the IETF list).=C2=A0</div></div></blockquote=
><div><br></div><div>Yes, this is very much a pragmatic spec, aimed at solv=
ing some very real-world problems (ie., i&#39;ve personally been beating my=
 head against the wall trying to usefully deploy freeswitch within Istio on=
 GKE. Let me know if you figured it out - its all hard due to how different=
 http is from sip.)</div><div><br></div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;">=
<div><br></div><div><br></div><div>Section 1.1: background<div>Will much of=
 it be meaningful to a protocol implementer in a year, no less five? I.e., =
referencing Kubernetes, Istio, CanaryDeploys, etc., more especially without=
 going into detail as to what they are? Sure, if you are an AWS developer y=
ou know them like the back of your hand. However, not many folks in IETF-la=
nd are.</div></div></div></blockquote><div><br></div><div>Happy to remove l=
ater on. Right now its just looknig to motivate the problem.=C2=A0</div><di=
v><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1=
ex"><div style=3D"overflow-wrap: break-word;"><div><div><br></div><div>1.2:=
</div><div>Why is it =E2=80=98unfortunate=E2=80=99 that people want to depl=
oy cloud applications that interact with the PSTN? Is that not an opportuni=
ty? I would take out the initial =E2=80=9CUnfortunately.=E2=80=9D It=E2=80=
=99s just a statement of fact that applications like to call and be called =
to most any person on the planet on a fully interoperable basis.</div></div=
></div></blockquote><div><br></div><div>Stray word from a few rounds of edi=
ts, will remove thx.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);=
padding-left:1ex"><div style=3D"overflow-wrap: break-word;"><div><div><br><=
/div><div>There needs to be more justification than =E2=80=9CWeb people don=
=E2=80=99t understand SIP.=E2=80=9D You might want to explain how RIPP is d=
ifferent than ParlayX. I would offer that RIPP:ParlayX::LDAP:X.500, but you=
 don=E2=80=99t say that anywhere. All the draft implies is Web people canno=
t grok SIP and SIP does not grok Web deployment, which does not reflect wel=
l on either community.</div></div></div></blockquote><div><br></div><div>No=
 - its not about how well people understand anything. Its about how applica=
ble modern cloud paas is to SIP and real-time services (which it really isn=
t), and how easy it is to build real-time apps which follow established des=
ign patterns for scalable and reliable systems (its super hard if not impos=
sible).=C2=A0</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex"><div style=3D"overflow-wrap: break-word;"><div><div><br></div><d=
iv>The last paragraph of 1.2 implies that SIP is the source of robocalls an=
d a mechanism that will be even easier to inject boatloads of calls will be=
 better. Please explain how this will be so (or remove the assertion). As y=
ou know, I personally wish it would be so, so PLEASE give me ammunition!</d=
iv></div></div></blockquote><div><br></div><div>It mandates passport as the=
 one and only way to even know the caller ID. COnsequently, the protocol dr=
ags stir along for the ride. There is no other special magic or anything.=
=C2=A0</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1=
ex"><div style=3D"overflow-wrap: break-word;"><div><div><br></div><div>1.4:=
</div><div>Given the virtual mandate to use SIP over TCP, is saying =E2=80=
=9Cwe can finally use HTTP over UDP=E2=80=9D not kind of a retrograde asser=
tion?</div></div></div></blockquote><div><br></div><div>I wasn&#39;t being =
specific enough here. &quot;SIP&quot; refers to the whole community of spec=
s needed for it, and this specific sentence is referring to RTP. I&#39;ll c=
hange to RTP.</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;"><div><d=
iv><br></div><div>REQ 12 (supporting only audio) is an odd requirement. Mor=
eover, most applications in the U.S. would require at least RTT support if =
not video. I am not a lawyer, but since we are talking interoperability wit=
h the PSTN, one is likely to run into Section 255 and 706 issues if RTT (an=
d video) is not there from day one.</div></div></div></blockquote><div><br>=
</div><div>I dont object - was just trying to start simple and solve one sp=
ecific problem well. Do current SIP trunking providers support RTT and vide=
o?=C2=A0</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,20=
4);padding-left:1ex"><div style=3D"overflow-wrap: break-word;"><div><div><b=
r></div><div>REQ 13 looks totally duplicative of REQ 10 (use secure caller =
ID technologies). Is there a distinction? If so, please make it clearer</di=
v></div></div></blockquote><div><br></div><div>Bug - thx.</div><div>=C2=A0<=
/div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"overfl=
ow-wrap: break-word;"><div><br></div><div>Section 3.6:</div><div>This is a =
place where we need to distinguish between theory and reality. In theory, S=
IP could provide end-to-end media security. In practice, no one figured it =
out. The good news is RIPP is no worse than SIP as it is deployed today. Mo=
reover, RIPP is no worse than HTTP. My concern is RIPP structurally will ne=
ver have end-to-end media security. While that is no worse than today, will=
 we be running into some privacy expectations of users and regulators? For =
example, a RIPP server can record everything. Problems here are (1) in the =
U.S. that is likely to be a violation of the Wiretap Act and (2) people can=
 use those recordings for very nefarious purposes (see RealTalk, Lyrebird, =
etc.).</div></div></blockquote><div><br></div><div>I would assert that RIPP=
 would be better than SIP today. There is a lot of SIP trunking out there (=
note RIPP *only* addresses inter-domain trunking cases, so thats the point =
of comparison). Most SIP trunking interconnection is over private IP with n=
o TLS or SRTP. There is also some over public Internet, I suspect much of i=
t is without SRTP and probably runs over VPN often. So, a peering protocol =
which requires hop encryption of signaling and media, and actually authenti=
cates both sides, is a big improvement over the current situation.</div><di=
v><br></div><div>E2E encryption isnt really relevant for the applicability =
of RIPP, since it is targeted at inter-domain connectivity towards PSTN. PS=
TN as you well know has no e2e encryption, and there must be decryption at =
the pstn gateway.=C2=A0</div><div><br></div><div>I dont follow how having a=
 server at the edge of a provider network might be in violation of the wire=
tap act, or somehow make recordings worse. THis is no different than the si=
tuation today with SBCs, where the vast majority have no SRTP at all, and e=
ven those that do, are decrypting so they have access to the media.=C2=A0</=
div><div><br></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;"><div><b=
r></div><div>Section 8.2:</div><div>Just curious: why MUST NOT the identifi=
er included the authority component for uniqueness? Is that a Web best prac=
tice? A reference or explanation would be helpful.</div></div></blockquote>=
<div><br></div><div>THis is the text:</div><div>&quot;<span style=3D"color:=
rgb(0,0,0);font-size:13.3333px">The path component MUST</span></div><pre cl=
ass=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bo=
ttom:0px;break-before:page;color:rgb(0,0,0)">   be a globally unique identi=
fier for this trunk, and not depend on the
   authority component as part of the namespace for purposes of
   uniqueness.&quot;</pre><pre class=3D"gmail-newpage" style=3D"font-size:1=
3.3333px;margin-top:0px;margin-bottom:0px;break-before:page;color:rgb(0,0,0=
)"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;marg=
in-top:0px;margin-bottom:0px;break-before:page;color:rgb(0,0,0)">The main m=
otivation is to eliminate the need for wildcard certs to authorize:</pre><p=
re class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;marg=
in-bottom:0px;break-before:page;color:rgb(0,0,0)"><a href=3D"https://12345a=
bcdef.ripp-provider.com">https://12345abcdef.ripp-provider.com</a> requires=
 one, whereas:</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333=
px;margin-top:0px;margin-bottom:0px;break-before:page;color:rgb(0,0,0)"><a =
href=3D"https://ripp-provider.com/12345abcdef">https://ripp-provider.com/12=
345abcdef</a> does not</pre><pre class=3D"gmail-newpage" style=3D"font-size=
:13.3333px;margin-top:0px;margin-bottom:0px;break-before:page;color:rgb(0,0=
,0)"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;ma=
rgin-top:0px;margin-bottom:0px;break-before:page;color:rgb(0,0,0)">I&#39;ll=
 include a mention on that, thanks.</pre><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;break-before:page;=
color:rgb(0,0,0)"><br></pre><div><br></div><div>=C2=A0</div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid =
rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;=
"><div><br></div><div>Section 8.3:</div><div>What are the capabilities *of*=
? Are they codec parameters? Are they decoded stream parameters? Are they c=
odec dependent? Is there any value to these parameters, outside parameteriz=
ed codec negotiation that I am missing?</div></div></blockquote><div><br></=
div><div>THese are the capabilities of the ripp trunk. These are things nor=
mally configured as capabilities/configuration of sip trunks today. Codecs,=
 bitrates, etc. They avoid the need for per-call capabilities exchange. Pra=
ctically speaking, such exchanges are un-needed since the capabilities are =
static properties of the configured sip trunks. Again, we&#39;re not replac=
ing sip overall - just providing a solution for inter-domain peering for vo=
ip to the pstn.</div><div><br></div><div>=C2=A0</div><blockquote class=3D"g=
mail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204=
,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;"><div>=
<br></div><div>For example, is the max-bitrate the decoded maximum bitrate =
or the bitrate of the codec? What is the purpose?</div><div>If I come up wi=
th a cool LPC that can do better than G.711 with 4000 samples per section, =
why cannot I use it?</div></div></blockquote><div><br></div><div>No this is=
 the max codec bitrate, same as current RTP definition. Meant more for like=
 opus.</div><div>The protocol of course is extensible to new codecs - each =
side advertises the codecs they support and their associated max rates. If =
someone however says they dont want to receive 4000 samples per second, the=
y&#39;d advertise a lower upper bound. This part is really no different tha=
n offer/answer.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);paddi=
ng-left:1ex"><div style=3D"overflow-wrap: break-word;"><div><br></div><div>=
BIG ONE: force-cbr</div><div>Do we require a CBR codec or do we require uni=
form packet sizes (in bytes)? If the former, why not just ask for a CBR cod=
ec in the request. If the latter, change the parameter to something like =
=E2=80=9Crequire privacy.=E2=80=9D There are much better ways of achieving =
privacy than using a CBR codec; see e.g.,=C2=A0<a href=3D"https://repositor=
y.library.georgetown.edu/handle/10822/1040699" target=3D"_blank">https://re=
pository.library.georgetown.edu/handle/10822/1040699</a></div></div></block=
quote><div><br></div><div>Cullen was the one who wanted this one - so he&#3=
9;s better positioned to speak to it. There was a study that found you coul=
d actually figure out what was being said just by looking at patterns of le=
ngths of variable bitrate audio codecs.</div><div>=C2=A0</div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px soli=
d rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-wor=
d;"><div><br></div><div>Section 8.4:</div><div>I agree with Cullen. It goes=
 against the design principle of using HTTP/3 for transport and then start =
putting application semantics into the transport layer. Is there a reason t=
o break layers, e.g., for load balancing or something else?</div></div></bl=
ockquote><div><br></div><div>Adding http headers is very common for web app=
s, so this isnt really different from normal web practice. That said, i don=
t feel that strongly on this one. Per above, right now its not so much abou=
t the specicic details of this proposal, as it is a starting point strawman=
 for working on the problem.</div><div><br></div><div>=C2=A0</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px =
solid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break=
-word;"><div><br></div><div>Section 8.7</div><div>Why 20 for forward and 10=
 for reverse? Are these rules of thumb, calculated optimal places to start,=
 SWAGs, or irrelevant?</div><div>Also (and in Section 8.8), what about the =
speed of light and the appearance of blocking?</div></div></blockquote><div=
><br></div><div>TOtal SWAGs right now - we need some implementation experie=
nce to make this a more usable spec.</div><div><br></div><div>Not sure I fo=
llow what you mean by speed of light?</div><div>=C2=A0</div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid =
rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;=
"><div><br></div><div>Section 8.8</div><div>Perhaps I missed it, but what h=
appens if a byway appears to be blocked for a =E2=80=98while=E2=80=99? Sinc=
e this is voice and this is UDP and I am tolerant of packet loss, do I care=
 a packet was lost? Also, I=E2=80=99m highly skeptical of 1:1 packet:ack th=
is protocol offers. That sounds like a bad thing for LANs and a terrible th=
ing for WANs. Really?</div></div></blockquote><div><br></div><div>We didnt =
spec out yet what happens if its blocked for a while. This would happen und=
er significant packet loss, so things are really bad. Probably drop the cal=
l.</div><div><br></div><div>Do you care if a packet is lost? in normal voip=
 cases, no. There are some use cases where the retransmission would be hand=
y (same reason we have an RTP extension for it..)</div><div><br></div><div>=
On the 1-1 packet:ack - thats a good comment. Definitely a drawback. Would =
be better if its piggybacked on reverse media, but we also currently have r=
everse media in separate byways. SO thats a tradeoff to be explored.</div><=
div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left=
:1ex"><div style=3D"overflow-wrap: break-word;"><div><br></div><div>Section=
 8.10</div><div>I kind of agree with Cullen=E2=80=99s point, but here is an=
 opposite observation that I came up with before I saw Cullen=E2=80=99s poi=
nt (that SIP result codes don=E2=80=99t cover ISDN). Why have a separate =
=E2=80=98rejected=E2=80=99 result in light of =E2=80=98failed.=E2=80=99 Unl=
ess rejected is referencing the SIP 607 result code, I would think all the =
call center cares about is whether the call completed and if it did not, wi=
ll a retry be likely to succeed or fail.</div></div></blockquote><div><br><=
/div><div><br></div><div>Rejected is meant to handle the case, like mobile,=
 where the end user actually hits the decline button.</div><div><br></div><=
div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0=
px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div styl=
e=3D"overflow-wrap: break-word;"><div><br></div><div><br></div><div>There a=
re a ton of typos in the draft. I=E2=80=99d be happy to hand my mark-up to =
whomever wants it in DISPATCH.</div></div></blockquote><div><br></div><div>=
Thanks! Probably not worth fixing till we&#39;ve gotten more experience und=
er our belt and revised it to reflect that.</div><div><br></div><div><br></=
div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">____=
___________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"noreferre=
r" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div><br clear=3D"all"><div><br></div>-- <br><div dir=3D"ltr"=
 class=3D"gmail_signature"><div dir=3D"ltr">Jonathan Rosenberg, Ph.D.<br><a=
 href=3D"mailto:jdrosen@jdrosen.net" target=3D"_blank">jdrosen@jdrosen.net<=
/a><br><a href=3D"http://www.jdrosen.net" target=3D"_blank">http://www.jdro=
sen.net</a></div></div></div>

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From: Victor Vasiliev <vasilvv@google.com>
Date: Mon, 22 Jul 2019 16:36:19 -0400
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Cc: IETF QUIC WG <quic@ietf.org>, HTTP Working Group <ietf-http-wg@w3.org>, hybi@ietf.org
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Subject: [dispatch] WebTransport Side Meeting (Tuesday, 15:20)
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Hello everyone,

Today, at the dispatch working group meeting (18:10), I am going to present
WebTransport. WebTransport is a protocol framework that allows multiplexed
and datagram-oriented transport protocols to be used by the web
applications (think =E2=80=9CWebSocket for UDP=E2=80=9D).

Since it=E2=80=99s quite likely we will run out of time during dispatch, I
scheduled a side-meeting to discuss this in more depth. Below are the side
meeting details.

*Time:* Tuesday, 15:20 ~ 16:50
*Place:* Room C2 (21st floor)
*Organizers:* Victor Vasiliev, David Schinazi
*Agenda:*

   1. A more in-depth overview of WebTransport.
   2. Discussion of the overall design and the use cases.
   3. As time permits, some of the major outstanding design issues in Web
   transport, e.g.:
      - What fallback protocol to use?
      - How to multiplex streams and datagrams within a single connection?
      - Can we let web applications do their own congestion control?
      - Should WebTransport sessions have URLs associated with them?

As usual, here are some helpful links:

   - WebTransport overview:
   https://tools.ietf.org/html/draft-vvv-webtransport-overview-00
   - QuicTransport:
   https://tools.ietf.org/html/draft-vvv-webtransport-quic-00
   - Http3Transport:
   https://tools.ietf.org/html/draft-vvv-webtransport-http3-00
   - Web API Spec draft: https://wicg.github.io/web-transport/
   - Discussion on use cases:
   https://discourse.wicg.io/t/webtransport-proposal/3508

Cheers,
  Victor.

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<div dir=3D"ltr">Hello everyone,<br><br>Today, at the dispatch working grou=
p meeting (18:10), I am going to present WebTransport. WebTransport is a pr=
otocol framework that allows multiplexed and datagram-oriented transport pr=
otocols to be used by the web applications (think =E2=80=9CWebSocket for UD=
P=E2=80=9D).<br><br>Since it=E2=80=99s quite likely we will run out of time=
 during dispatch, I scheduled a side-meeting to discuss this in more depth.=
 Below are the side meeting details.<br><br><b>Time:</b> Tuesday, 15:20 ~ 1=
6:50<br><b>Place:</b> Room C2 (21st floor)<br><b>Organizers:</b> Victor Vas=
iliev, David Schinazi<br><b>Agenda:</b><br><ol><li>A more in-depth overview=
 of WebTransport.</li><li>Discussion of the overall design and the use case=
s.</li><li>As time permits, some of the major outstanding design issues in =
Web transport, e.g.:</li><ul><li>What fallback protocol to use?</li><li>How=
 to multiplex streams and datagrams within a single connection?</li><li>Can=
 we let web applications do their own congestion control?</li><li>Should We=
bTransport sessions have URLs associated with them?</li></ul></ol>As usual,=
 here are some helpful links:<br><ul><li>WebTransport overview: <a href=3D"=
https://tools.ietf.org/html/draft-vvv-webtransport-overview-00">https://too=
ls.ietf.org/html/draft-vvv-webtransport-overview-00</a></li><li>QuicTranspo=
rt: <a href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-quic-00">=
https://tools.ietf.org/html/draft-vvv-webtransport-quic-00</a></li><li>Http=
3Transport: <a href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-h=
ttp3-00">https://tools.ietf.org/html/draft-vvv-webtransport-http3-00</a></l=
i><li>Web API Spec draft: <a href=3D"https://wicg.github.io/web-transport/"=
>https://wicg.github.io/web-transport/</a></li><li>Discussion on use cases:=
 <a href=3D"https://discourse.wicg.io/t/webtransport-proposal/3508">https:/=
/discourse.wicg.io/t/webtransport-proposal/3508</a></li></ul>Cheers,<br>=C2=
=A0 Victor.<br></div>

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Subject: Re: [dispatch] [hybi] WebTransport Side Meeting (Tuesday, 15:20)
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On 7/22/19 1:36 PM, Victor Vasiliev wrote:
> Hello everyone,
> 
> Today, at the dispatch working group meeting (18:10), I am going to 
> present WebTransport. WebTransport is a protocol framework that allows 
> multiplexed and datagram-oriented transport protocols to be used by the 
> web applications (think “WebSocket for UDP”).

"Historically, web applications that needed bidirectional data stream
    between a client and a server could rely on WebSockets [RFC6455], a
    message-based protocol compatible with Web security model.  However,
    since the abstraction it provides is a single ordered stream of
    messages, it suffers from head-of-line blocking (HOLB), meaning that
    all messages must be sent and received in order even if they are
    independent and some of them are no longer needed.  This makes it a
    poor fit for latency sensitive applications which rely on partial
    reliability and stream independence for performance."

The HOLB isn't really entirely the case... RFC6455 ws allows arbitrary 
fragmentation of messages allowing interleaving with control frames.

ws-over-h2 allows you to can the h2 stream when you want as well.

" Each new stream would require a WebSocket handshake to agree on
       application protocol used, meaning that it would take at least one
       RTT for each new stream before the client can write to it."

Yes it was knowingly done as a hack to try to encourage uptake from 
browser vendors... it's not really integrated into the encapsulating 
protocol.

>   * WebTransport overview:
>     https://tools.ietf.org/html/draft-vvv-webtransport-overview-00
>   * QuicTransport:
>     https://tools.ietf.org/html/draft-vvv-webtransport-quic-00
>   * Http3Transport:
>     https://tools.ietf.org/html/draft-vvv-webtransport-http3-00

There's no h2 transport implementation?

Not everything that might want to use this will get h3 capability in a 
reasonable timeframe.  If there's more momentum behind it than RFC8441 
there's probably room for a generic long-lived bidirectional extension 
to h2 either reusing DATA or a new frame type.

It's a good idea to have it ride on other protocols.  Not doing this 
really hurt RFC6455 ws since deploying it usually needed extra, 
different servers with the attendant difficulties interoperating with 
other protocols.

I really suggest thinking through the effects of not having an RFC6455 
type subprotocol (unless I failed to spot it).  It really makes an 
implicit assumption about what the stream will carry that doesn't scale 
beyond one server carrying one thing.  That's not how things tend to pan 
out if the protocol is useful.  The url path could be hacked to imply 
the subprotocol but if that's not standardized it's still a mess.  And 
the subprotocol binding may be orthogonal to the url layout complicating 
things needlessly.

-Andy

>   * Web API Spec draft: https://wicg.github.io/web-transport/
>   * Discussion on use cases:
>     https://discourse.wicg.io/t/webtransport-proposal/3508
> 
> Cheers,
>    Victor.
> 
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
> 


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Date: Mon, 22 Jul 2019 17:29:09 -0400
Cc: Victor Vasiliev <vasilvv=40google.com@dmarc.ietf.org>, dispatch@ietf.org,  David Schinazi <dschinazi@google.com>, hybi@ietf.org, IETF QUIC WG <quic@ietf.org>, HTTP Working Group <ietf-http-wg@w3.org>
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Subject: Re: [dispatch] [hybi] WebTransport Side Meeting (Tuesday, 15:20)
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> On Jul 22, 2019, at 4:59 PM, Andy Green <andy@warmcat.com> wrote:
>=20
>=20
>=20
> On 7/22/19 1:36 PM, Victor Vasiliev wrote:
>> Hello everyone,
>> Today, at the dispatch working group meeting (18:10), I am going to =
present WebTransport. WebTransport is a protocol framework that allows =
multiplexed and datagram-oriented transport protocols to be used by the =
web applications (think =E2=80=9CWebSocket for UDP=E2=80=9D).
>=20
> "Historically, web applications that needed bidirectional data stream
>   between a client and a server could rely on WebSockets [RFC6455], a
>   message-based protocol compatible with Web security model.  However,
>   since the abstraction it provides is a single ordered stream of
>   messages, it suffers from head-of-line blocking (HOLB), meaning that
>   all messages must be sent and received in order even if they are
>   independent and some of them are no longer needed.  This makes it a
>   poor fit for latency sensitive applications which rely on partial
>   reliability and stream independence for performance."
>=20
> The HOLB isn't really entirely the case... RFC6455 ws allows arbitrary =
fragmentation of messages allowing interleaving with control frames.
>=20
> ws-over-h2 allows you to can the h2 stream when you want as well.
>=20
> " Each new stream would require a WebSocket handshake to agree on
>      application protocol used, meaning that it would take at least =
one
>      RTT for each new stream before the client can write to it."
>=20
> Yes it was knowingly done as a hack to try to encourage uptake from =
browser vendors... it's not really integrated into the encapsulating =
protocol.
>=20
>>  * WebTransport overview:
>>    https://tools.ietf.org/html/draft-vvv-webtransport-overview-00
>>  * QuicTransport:
>>    https://tools.ietf.org/html/draft-vvv-webtransport-quic-00
>>  * Http3Transport:
>>    https://tools.ietf.org/html/draft-vvv-webtransport-http3-00
>=20
> There's no h2 transport implementation?
>=20
> Not everything that might want to use this will get h3 capability in a =
reasonable timeframe.  If there's more momentum behind it than RFC8441 =
there's probably room for a generic long-lived bidirectional extension =
to h2 either reusing DATA or a new frame type.

Definitely agree! I know that we=E2=80=99ve been chatting a bit with =
Victor about =
https://datatracker.ietf.org/doc/draft-kinnear-httpbis-http2-transport/ =
which aims to provide this, and I think it would be worth making sure =
that this works nicely with WebTransport.=20
-00 for that document covers effectively what you=E2=80=99d get with a =
new frame type, and -01 extends 8441 to cover more than just WebSockets =
with the extended CONNECT handshake.
I don=E2=80=99t have a particularly strong preference for the mechanism =
used, but rather care more about the outcome =E2=80=94 very much agree =
that this is a useful component.

Thanks,
Eric

>=20
> It's a good idea to have it ride on other protocols.  Not doing this =
really hurt RFC6455 ws since deploying it usually needed extra, =
different servers with the attendant difficulties interoperating with =
other protocols.
>=20
> I really suggest thinking through the effects of not having an RFC6455 =
type subprotocol (unless I failed to spot it).  It really makes an =
implicit assumption about what the stream will carry that doesn't scale =
beyond one server carrying one thing.  That's not how things tend to pan =
out if the protocol is useful.  The url path could be hacked to imply =
the subprotocol but if that's not standardized it's still a mess.  And =
the subprotocol binding may be orthogonal to the url layout complicating =
things needlessly.
>=20
> -Andy
>=20
>>  * Web API Spec draft: https://wicg.github.io/web-transport/
>>  * Discussion on use cases:
>>    https://discourse.wicg.io/t/webtransport-proposal/3508
>> Cheers,
>>   Victor.
>> _______________________________________________
>> hybi mailing list
>> hybi@ietf.org
>> https://www.ietf.org/mailman/listinfo/hybi
>=20


From nobody Mon Jul 22 16:03:06 2019
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From: Bernie Hoeneisen <bernie@ietf.hoeneisen.ch>
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Subject: [dispatch] Side-Meeting MEDUP (incl. Private Key Synchronization)
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As just instructed by the chair, I am sending the side meeting information 
for MEDUP (Missing Elemments for Decentralized and Usable Privacy) to the 
list. Please find more information below.

The most insteresting topic appears to be the protocol to synchronize 
Private Keys among different devices belonging to the same user. For a few 
weeks we have runing code and there is an early Internet Draft to 
specify the KeySync protocol:

   https://tools.ietf.org/html/draft-hoeneisen-pep-keysync-00

Discussions will be on the MEDUP Mailing list:

  https://www.ietf.org/mailman/listinfo/MEDUP

Looking forward to see you on Wed, 14:50-16:50 in room C2 (21st floor)!

All the best,
  Bernie


> On Mon, 22 Jul 2019, Hernâni Marques (p≡p foundation) wrote:
>
>> Dear folks
>> 
>> 
>> Following our meeting @ IETF-104 in Prague [1][2], we will have our second 
>> non-WG meeting @ IETF-105 in Montreal, taking place Wed, 24 July, from 
>> 14:50-16:50 EDT in room C2 (21st floor).
>> 
>> To know how many people to expect, it would be nice you could sign up
>> here (with some name):
>>
>>    https://du7f.koalatux.ch/SrtXRHemfNY-xM3k-QWGKA
>> 
>> Please note that this time we will also provide a Jitsi/WebRTC remote
>> participation channel (by best effort):
>>
>>    https://meet.digitale-gesellschaft.ch/medup
>> 
>> The current agenda is here:
>> 
>> 
>> https://pep.foundation/dev/repos/internet-drafts/file/tip/medup/ietf-105/agenda.txt
>> 
>> In that respect, if you want to present an approach / a missing piece in
>> scope of MEDUP or suggest other changes to the agenda, please reach out!
>> 
>> 
>> Cu soon & thanks!
>> 
>> Greets
>> 
>> Hernani
>> 
>> 
>> [1] Minutes from IETF-104:
>>     https://pep.foundation/dev/repos/internet-drafts/file/tip/medup/ietf-104/notes-ietf-104-medup-00.txt
>> 
>> [2] Meeting materials and video podcast:
>>     https://pep.foundation/docs/ietf/104/medup/
>> 
>> -- 
>> p≡p foundation: https://pep.foundation/
>
--37663318-251774225-1563836578=:12363--


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To: Eric Kinnear <ekinnear=40apple.com@dmarc.ietf.org>
Cc: Andy Green <andy@warmcat.com>, Victor Vasiliev <vasilvv=40google.com@dmarc.ietf.org>, hybi@ietf.org,  David Schinazi <dschinazi@google.com>, dispatch@ietf.org,  HTTP Working Group <ietf-http-wg@w3.org>, IETF QUIC WG <quic@ietf.org>
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Subject: Re: [dispatch] [hybi] WebTransport Side Meeting (Tuesday, 15:20)
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At a quick glance,
https://datatracker.ietf.org/doc/draft-kinnear-httpbis-http2-transport/?inc=
lude_text=3D1
seems
like a decent fit for the WebTransport  API.  It seems better than trying
to fit the WebSocket API.  But do we expect people to implement it on
servers before they implement QUIC?  I suppose even if it takes longer, it
may have the advantage of working on more networks than QUIC and HTTP/3
potentially (for networks that still block UDP, for example).



On Mon, Jul 22, 2019 at 5:29 PM Eric Kinnear <ekinnear=3D
40apple.com@dmarc.ietf.org> wrote:

>
>
> > On Jul 22, 2019, at 4:59 PM, Andy Green <andy@warmcat.com> wrote:
> >
> >
> >
> > On 7/22/19 1:36 PM, Victor Vasiliev wrote:
> >> Hello everyone,
> >> Today, at the dispatch working group meeting (18:10), I am going to
> present WebTransport. WebTransport is a protocol framework that allows
> multiplexed and datagram-oriented transport protocols to be used by the w=
eb
> applications (think =E2=80=9CWebSocket for UDP=E2=80=9D).
> >
> > "Historically, web applications that needed bidirectional data stream
> >   between a client and a server could rely on WebSockets [RFC6455], a
> >   message-based protocol compatible with Web security model.  However,
> >   since the abstraction it provides is a single ordered stream of
> >   messages, it suffers from head-of-line blocking (HOLB), meaning that
> >   all messages must be sent and received in order even if they are
> >   independent and some of them are no longer needed.  This makes it a
> >   poor fit for latency sensitive applications which rely on partial
> >   reliability and stream independence for performance."
> >
> > The HOLB isn't really entirely the case... RFC6455 ws allows arbitrary
> fragmentation of messages allowing interleaving with control frames.
> >
> > ws-over-h2 allows you to can the h2 stream when you want as well.
> >
> > " Each new stream would require a WebSocket handshake to agree on
> >      application protocol used, meaning that it would take at least one
> >      RTT for each new stream before the client can write to it."
> >
> > Yes it was knowingly done as a hack to try to encourage uptake from
> browser vendors... it's not really integrated into the encapsulating
> protocol.
> >
> >>  * WebTransport overview:
> >>    https://tools.ietf.org/html/draft-vvv-webtransport-overview-00
> >>  * QuicTransport:
> >>    https://tools.ietf.org/html/draft-vvv-webtransport-quic-00
> >>  * Http3Transport:
> >>    https://tools.ietf.org/html/draft-vvv-webtransport-http3-00
> >
> > There's no h2 transport implementation?
> >
> > Not everything that might want to use this will get h3 capability in a
> reasonable timeframe.  If there's more momentum behind it than RFC8441
> there's probably room for a generic long-lived bidirectional extension to
> h2 either reusing DATA or a new frame type.
>
> Definitely agree! I know that we=E2=80=99ve been chatting a bit with Vict=
or about
> https://datatracker.ietf.org/doc/draft-kinnear-httpbis-http2-transport/
> which aims to provide this, and I think it would be worth making sure tha=
t
> this works nicely with WebTransport.
> -00 for that document covers effectively what you=E2=80=99d get with a ne=
w frame
> type, and -01 extends 8441 to cover more than just WebSockets with the
> extended CONNECT handshake.
> I don=E2=80=99t have a particularly strong preference for the mechanism u=
sed, but
> rather care more about the outcome =E2=80=94 very much agree that this is=
 a useful
> component.
>
> Thanks,
> Eric
>
> >
> > It's a good idea to have it ride on other protocols.  Not doing this
> really hurt RFC6455 ws since deploying it usually needed extra, different
> servers with the attendant difficulties interoperating with other protoco=
ls.
> >
> > I really suggest thinking through the effects of not having an RFC6455
> type subprotocol (unless I failed to spot it).  It really makes an implic=
it
> assumption about what the stream will carry that doesn't scale beyond one
> server carrying one thing.  That's not how things tend to pan out if the
> protocol is useful.  The url path could be hacked to imply the subprotoco=
l
> but if that's not standardized it's still a mess.  And the subprotocol
> binding may be orthogonal to the url layout complicating things needlessl=
y.
> >
> > -Andy
> >
> >>  * Web API Spec draft: https://wicg.github.io/web-transport/
> >>  * Discussion on use cases:
> >>    https://discourse.wicg.io/t/webtransport-proposal/3508
> >> Cheers,
> >>   Victor.
> >> _______________________________________________
> >> hybi mailing list
> >> hybi@ietf.org
> >> https://www.ietf.org/mailman/listinfo/hybi
> >
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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<div dir=3D"ltr"><div><br></div>At a quick glance,=C2=A0<a href=3D"https://=
datatracker.ietf.org/doc/draft-kinnear-httpbis-http2-transport/?include_tex=
t=3D1">https://datatracker.ietf.org/doc/draft-kinnear-httpbis-http2-transpo=
rt/?include_text=3D1</a>=C2=A0seems like a decent fit for the WebTransport=
=C2=A0=C2=A0API.=C2=A0 It seems better than trying to fit the WebSocket API=
.=C2=A0 But do we expect people to implement it on servers before they impl=
ement QUIC?=C2=A0 I suppose even if it takes longer, it may have the advant=
age of working on more networks than QUIC and HTTP/3 potentially (for netwo=
rks that still block UDP, for example).=C2=A0<div><br></div><div><br></div>=
</div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">=
On Mon, Jul 22, 2019 at 5:29 PM Eric Kinnear &lt;ekinnear=3D<a href=3D"mail=
to:40apple.com@dmarc.ietf.org">40apple.com@dmarc.ietf.org</a>&gt; wrote:<br=
></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left:1px solid rgb(204,204,204);padding-left:1ex"><br>
<br>
&gt; On Jul 22, 2019, at 4:59 PM, Andy Green &lt;<a href=3D"mailto:andy@war=
mcat.com" target=3D"_blank">andy@warmcat.com</a>&gt; wrote:<br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; On 7/22/19 1:36 PM, Victor Vasiliev wrote:<br>
&gt;&gt; Hello everyone,<br>
&gt;&gt; Today, at the dispatch working group meeting (18:10), I am going t=
o present WebTransport. WebTransport is a protocol framework that allows mu=
ltiplexed and datagram-oriented transport protocols to be used by the web a=
pplications (think =E2=80=9CWebSocket for UDP=E2=80=9D).<br>
&gt; <br>
&gt; &quot;Historically, web applications that needed bidirectional data st=
ream<br>
&gt;=C2=A0 =C2=A0between a client and a server could rely on WebSockets [RF=
C6455], a<br>
&gt;=C2=A0 =C2=A0message-based protocol compatible with Web security model.=
=C2=A0 However,<br>
&gt;=C2=A0 =C2=A0since the abstraction it provides is a single ordered stre=
am of<br>
&gt;=C2=A0 =C2=A0messages, it suffers from head-of-line blocking (HOLB), me=
aning that<br>
&gt;=C2=A0 =C2=A0all messages must be sent and received in order even if th=
ey are<br>
&gt;=C2=A0 =C2=A0independent and some of them are no longer needed.=C2=A0 T=
his makes it a<br>
&gt;=C2=A0 =C2=A0poor fit for latency sensitive applications which rely on =
partial<br>
&gt;=C2=A0 =C2=A0reliability and stream independence for performance.&quot;=
<br>
&gt; <br>
&gt; The HOLB isn&#39;t really entirely the case... RFC6455 ws allows arbit=
rary fragmentation of messages allowing interleaving with control frames.<b=
r>
&gt; <br>
&gt; ws-over-h2 allows you to can the h2 stream when you want as well.<br>
&gt; <br>
&gt; &quot; Each new stream would require a WebSocket handshake to agree on=
<br>
&gt;=C2=A0 =C2=A0 =C2=A0 application protocol used, meaning that it would t=
ake at least one<br>
&gt;=C2=A0 =C2=A0 =C2=A0 RTT for each new stream before the client can writ=
e to it.&quot;<br>
&gt; <br>
&gt; Yes it was knowingly done as a hack to try to encourage uptake from br=
owser vendors... it&#39;s not really integrated into the encapsulating prot=
ocol.<br>
&gt; <br>
&gt;&gt;=C2=A0 * WebTransport overview:<br>
&gt;&gt;=C2=A0 =C2=A0 <a href=3D"https://tools.ietf.org/html/draft-vvv-webt=
ransport-overview-00" rel=3D"noreferrer" target=3D"_blank">https://tools.ie=
tf.org/html/draft-vvv-webtransport-overview-00</a><br>
&gt;&gt;=C2=A0 * QuicTransport:<br>
&gt;&gt;=C2=A0 =C2=A0 <a href=3D"https://tools.ietf.org/html/draft-vvv-webt=
ransport-quic-00" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.o=
rg/html/draft-vvv-webtransport-quic-00</a><br>
&gt;&gt;=C2=A0 * Http3Transport:<br>
&gt;&gt;=C2=A0 =C2=A0 <a href=3D"https://tools.ietf.org/html/draft-vvv-webt=
ransport-http3-00" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.=
org/html/draft-vvv-webtransport-http3-00</a><br>
&gt; <br>
&gt; There&#39;s no h2 transport implementation?<br>
&gt; <br>
&gt; Not everything that might want to use this will get h3 capability in a=
 reasonable timeframe.=C2=A0 If there&#39;s more momentum behind it than RF=
C8441 there&#39;s probably room for a generic long-lived bidirectional exte=
nsion to h2 either reusing DATA or a new frame type.<br>
<br>
Definitely agree! I know that we=E2=80=99ve been chatting a bit with Victor=
 about <a href=3D"https://datatracker.ietf.org/doc/draft-kinnear-httpbis-ht=
tp2-transport/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.ie=
tf.org/doc/draft-kinnear-httpbis-http2-transport/</a> which aims to provide=
 this, and I think it would be worth making sure that this works nicely wit=
h WebTransport. <br>
-00 for that document covers effectively what you=E2=80=99d get with a new =
frame type, and -01 extends 8441 to cover more than just WebSockets with th=
e extended CONNECT handshake.<br>
I don=E2=80=99t have a particularly strong preference for the mechanism use=
d, but rather care more about the outcome =E2=80=94 very much agree that th=
is is a useful component.<br>
<br>
Thanks,<br>
Eric<br>
<br>
&gt; <br>
&gt; It&#39;s a good idea to have it ride on other protocols.=C2=A0 Not doi=
ng this really hurt RFC6455 ws since deploying it usually needed extra, dif=
ferent servers with the attendant difficulties interoperating with other pr=
otocols.<br>
&gt; <br>
&gt; I really suggest thinking through the effects of not having an RFC6455=
 type subprotocol (unless I failed to spot it).=C2=A0 It really makes an im=
plicit assumption about what the stream will carry that doesn&#39;t scale b=
eyond one server carrying one thing.=C2=A0 That&#39;s not how things tend t=
o pan out if the protocol is useful.=C2=A0 The url path could be hacked to =
imply the subprotocol but if that&#39;s not standardized it&#39;s still a m=
ess.=C2=A0 And the subprotocol binding may be orthogonal to the url layout =
complicating things needlessly.<br>
&gt; <br>
&gt; -Andy<br>
&gt; <br>
&gt;&gt;=C2=A0 * Web API Spec draft: <a href=3D"https://wicg.github.io/web-=
transport/" rel=3D"noreferrer" target=3D"_blank">https://wicg.github.io/web=
-transport/</a><br>
&gt;&gt;=C2=A0 * Discussion on use cases:<br>
&gt;&gt;=C2=A0 =C2=A0 <a href=3D"https://discourse.wicg.io/t/webtransport-p=
roposal/3508" rel=3D"noreferrer" target=3D"_blank">https://discourse.wicg.i=
o/t/webtransport-proposal/3508</a><br>
&gt;&gt; Cheers,<br>
&gt;&gt;=C2=A0 =C2=A0Victor.<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; hybi mailing list<br>
&gt;&gt; <a href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</=
a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/hybi" rel=3D"nore=
ferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><br=
>
&gt; <br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"noreferre=
r" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div>

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Subject: Re: [dispatch] WebTransport Side Meeting (Tuesday, 15:20)
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Thanks to everyone who came to the side-meeting today. We had 27 people in
the room and strong interest in the topic.

In terms of next steps: this work will proceed in parallel at the W3C, and
we will seek to open an IETF mailing list to discuss WebTransport (we'll
report the list here when it exists).

Thanks in particular to Steve Anton for recording the following minutes:

Roberto Peon: best to share connection even for data with different
congestion controllers
Seth Blank: many games open many TCP connections for reliable data, in
addition to UDP
Seth: others will layer a reliable layer on top of UDP
Jana Iyengar: having all types of data under the same cc is very useful,
but needs a way to prioritize traffic. Another issue with existing
approaches: some connections (if multiple TCP/UDP/etc) may fail which is
awkward (e.g., TCP may work but UDP not).
Is connection setup latency important?
Yes, RTC use cases find call setup very important
QUIC should help a lot here
Use case: for legacy apps, map UDP sockets to QUIC datagrams, TCP sockets
to QUIC streams
Brian Trammell: but these legacy apps want to interop with things that
speak TCP and UDP, not QUIC
Solution: host a shim proxy
Roberto: a L7 proxy wants very low level APIs to forward data avoid
compounding delays.
Partial reliability models
Jana: agree that datagrams are too low level for from-scratch applications.
SCTP made this mistake. What is the API for sending application-sized
messages? (streams?)
Roberto: QUIC is missing the concept of =E2=80=9Csession=E2=80=9D (bundling=
 streams
together to the same endpoint). Use case: L7 proxies that route at the
session level.
Roberto: protocol needs to know about framing and ordering (for flow
control)
Martin Thomson: don=E2=80=99t invent partial reliability here, or else it c=
ould be
the hill to die on.
Victor Vasiliev: want to just expose APIs for QUIC primitives that are
already defined (imagine streams are a useful abstraction, datagrams are
the most minimal abstraction)
Target application: real-time client-server streaming
Not going to solve this in WebTransport, instead offer primitives to build
on.
TAPS
Brian: Underlying transport depends on application and deployment. Will
likely need a transport selection algorithm.
Victor: want to do transport selection as a higher level layer, not in
WebTransport.
Victor: some of the data API will use Web primitives (like WHATWG Streams).
Don=E2=80=99t want to over abstract.
Brian: take a close look at the TAPS architecture document.
QuicTransport vs. Http3Transport
Roberto: things we implement in HTTP: redirection, shut down state,
caching, multiplexing. These are very valuable, let=E2=80=99s not forget.
Important to ship an API which performs and is what people want to use.
Eric Kinnear: HTTP has a few downsides, e.g. bidirectionality. There=E2=80=
=99s a
use case for just a QUIC transport.
Peter Thatcher: HTTP doesn=E2=80=99t make sense for P2P.
Martin: How does this spec handle fallback if e.g. QUIC can=E2=80=99t conne=
ct?
There should be a transparent fallback.
Victor: WebSocket fallback (can even implement as a polyfill).
Jana: may need to break down spec into pieces before bringing it into the
IETF.
Victor: wire format changes are clearly IETF. API design is clearly W3C.
Framework definition could go either way, not sure which venue right now.
Martin: kind of weird but seems to work to have the same people wear IETF
and W3C hats
David Schinazi: seems to be interest: in terms of process, should a new
IETF mailing list be formed?
Martin: get an AD to form a mailing list. IETF should do work for W3C, not
the other way around.
Roberto: implementations of WebTransport could be useful without it
necessarily being a Web API.
Mark Nottingham/Martin/Peter: W3C work is in WICG right now. Needs to be a
working group before IETF would form a working group.
Can still make a mailing list, have a BOF before the working group was
formed.
Jana: Other way of doing it: W3C driving requirements for IETF protocol,
creates API.
Roberto: some use cases for L7 proxies even if this doesn=E2=80=99t end up =
in the
browser.
Jana: Is the RTCDataChannel a first application for WebTransport?
Harald Alvestrand: desire in WebRTC to redesign the RTCDataChannel API.
Roberto: a major application that wants this is low latency broadcast video=
.
Bandwidth prediction APIs
Roberto: value for this, very complicated though. Today the server which
has lower level control can send the browser cc info.
Harald: started with receiver cc, went back to sender cc later
Brian: should take this out of scope for a BOF
Roberto: defer this or else it will fail (e.g., HTTP2 started without QUIC
even though they knew QUIC was needed).
Jana: don=E2=80=99t bring bandwidth prediction up or else it will be front =
and
center and kill the effort

On Mon, Jul 22, 2019 at 4:36 PM Victor Vasiliev <vasilvv=3D
40google.com@dmarc.ietf.org> wrote:

> Hello everyone,
>
> Today, at the dispatch working group meeting (18:10), I am going to
> present WebTransport. WebTransport is a protocol framework that allows
> multiplexed and datagram-oriented transport protocols to be used by the w=
eb
> applications (think =E2=80=9CWebSocket for UDP=E2=80=9D).
>
> Since it=E2=80=99s quite likely we will run out of time during dispatch, =
I
> scheduled a side-meeting to discuss this in more depth. Below are the sid=
e
> meeting details.
>
> *Time:* Tuesday, 15:20 ~ 16:50
> *Place:* Room C2 (21st floor)
> *Organizers:* Victor Vasiliev, David Schinazi
> *Agenda:*
>
>    1. A more in-depth overview of WebTransport.
>    2. Discussion of the overall design and the use cases.
>    3. As time permits, some of the major outstanding design issues in Web
>    transport, e.g.:
>       - What fallback protocol to use?
>       - How to multiplex streams and datagrams within a single connection=
?
>       - Can we let web applications do their own congestion control?
>       - Should WebTransport sessions have URLs associated with them?
>
> As usual, here are some helpful links:
>
>    - WebTransport overview:
>    https://tools.ietf.org/html/draft-vvv-webtransport-overview-00
>    - QuicTransport:
>    https://tools.ietf.org/html/draft-vvv-webtransport-quic-00
>    - Http3Transport:
>    https://tools.ietf.org/html/draft-vvv-webtransport-http3-00
>    - Web API Spec draft: https://wicg.github.io/web-transport/
>    - Discussion on use cases:
>    https://discourse.wicg.io/t/webtransport-proposal/3508
>
> Cheers,
>   Victor.
>

--00000000000003b8ff058e5ff839
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks to everyone who came to the side-meeting today. We =
had 27 people in the room and strong interest in the topic.<div><br></div><=
div>In terms of next steps: this work will proceed in parallel at the W3C, =
and we will seek to open an IETF mailing list to discuss WebTransport (we&#=
39;ll report the list here when it exists).<br><div><br></div><div>Thanks i=
n particular to Steve Anton for recording the following minutes:</div><div>=
<br>Roberto Peon: best to share connection even for data with different con=
gestion controllers<br>Seth Blank: many games open many TCP connections for=
 reliable data, in addition to UDP<br>Seth: others will layer a reliable la=
yer on top of UDP<br>Jana Iyengar: having all types of data under the same =
cc is very useful, but needs a way to prioritize traffic. Another issue wit=
h existing approaches: some connections (if multiple TCP/UDP/etc) may fail =
which is awkward (e.g., TCP may work but UDP not).<br>Is connection setup l=
atency important?<br>Yes, RTC use cases find call setup very important<br>Q=
UIC should help a lot here<br>Use case: for legacy apps, map UDP sockets to=
 QUIC datagrams, TCP sockets to QUIC streams<br>Brian Trammell: but these l=
egacy apps want to interop with things that speak TCP and UDP, not QUIC<br>=
Solution: host a shim proxy<br>Roberto: a L7 proxy wants very low level API=
s to forward data avoid compounding delays.<br>Partial reliability models<b=
r>Jana: agree that datagrams are too low level for from-scratch application=
s. SCTP made this mistake. What is the API for sending application-sized me=
ssages? (streams?)<br>Roberto: QUIC is missing the concept of =E2=80=9Csess=
ion=E2=80=9D (bundling streams together to the same endpoint). Use case: L7=
 proxies that route at the session level.<br>Roberto: protocol needs to kno=
w about framing and ordering (for flow control)<br>Martin Thomson: don=E2=
=80=99t invent partial reliability here, or else it could be the hill to di=
e on.<br>Victor Vasiliev: want to just expose APIs for QUIC primitives that=
 are already defined (imagine streams are a useful abstraction, datagrams a=
re the most minimal abstraction)<br>Target application: real-time client-se=
rver streaming<br>Not going to solve this in WebTransport, instead offer pr=
imitives to build on.<br>TAPS<br>Brian: Underlying transport depends on app=
lication and deployment. Will likely need a transport selection algorithm.<=
br>Victor: want to do transport selection as a higher level layer, not in W=
ebTransport.<br>Victor: some of the data API will use Web primitives (like =
WHATWG Streams). Don=E2=80=99t want to over abstract.<br>Brian: take a clos=
e look at the TAPS architecture document.<br>QuicTransport vs. Http3Transpo=
rt<br>Roberto: things we implement in HTTP: redirection, shut down state, c=
aching, multiplexing. These are very valuable, let=E2=80=99s not forget.<br=
>Important to ship an API which performs and is what people want to use.<br=
>Eric Kinnear: HTTP has a few downsides, e.g. bidirectionality. There=E2=80=
=99s a use case for just a QUIC transport.<br>Peter Thatcher: HTTP doesn=E2=
=80=99t make sense for P2P.<br>Martin: How does this spec handle fallback i=
f e.g. QUIC can=E2=80=99t connect? There should be a transparent fallback.<=
br>Victor: WebSocket fallback (can even implement as a polyfill).<br>Jana: =
may need to break down spec into pieces before bringing it into the IETF.<b=
r>Victor: wire format changes are clearly IETF. API design is clearly W3C. =
Framework definition could go either way, not sure which venue right now.<b=
r>Martin: kind of weird but seems to work to have the same people wear IETF=
 and W3C hats<br>David Schinazi: seems to be interest: in terms of process,=
 should a new IETF mailing list be formed?<br>Martin: get an AD to form a m=
ailing list. IETF should do work for W3C, not the other way around.<br>Robe=
rto: implementations of WebTransport could be useful without it necessarily=
 being a Web API.<br>Mark Nottingham/Martin/Peter: W3C work is in WICG righ=
t now. Needs to be a working group before IETF would form a working group.<=
br>Can still make a mailing list, have a BOF before the working group was f=
ormed.<br>Jana: Other way of doing it: W3C driving requirements for IETF pr=
otocol, creates API.<br>Roberto: some use cases for L7 proxies even if this=
 doesn=E2=80=99t end up in the browser.<br>Jana: Is the RTCDataChannel a fi=
rst application for WebTransport?<br>Harald Alvestrand: desire in WebRTC to=
 redesign the RTCDataChannel API.<br>Roberto: a major application that want=
s this is low latency broadcast video.<br>Bandwidth prediction APIs<br>Robe=
rto: value for this, very complicated though. Today the server which has lo=
wer level control can send the browser cc info.<br>Harald: started with rec=
eiver cc, went back to sender cc later<br>Brian: should take this out of sc=
ope for a BOF<br>Roberto: defer this or else it will fail (e.g., HTTP2 star=
ted without QUIC even though they knew QUIC was needed).<br>Jana: don=E2=80=
=99t bring bandwidth prediction up or else it will be front and center and =
kill the effort<br></div></div></div><br><div class=3D"gmail_quote"><div di=
r=3D"ltr" class=3D"gmail_attr">On Mon, Jul 22, 2019 at 4:36 PM Victor Vasil=
iev &lt;vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org">40google.c=
om@dmarc.ietf.org</a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);p=
adding-left:1ex"><div dir=3D"ltr">Hello everyone,<br><br>Today, at the disp=
atch working group meeting (18:10), I am going to present WebTransport. Web=
Transport is a protocol framework that allows multiplexed and datagram-orie=
nted transport protocols to be used by the web applications (think =E2=80=
=9CWebSocket for UDP=E2=80=9D).<br><br>Since it=E2=80=99s quite likely we w=
ill run out of time during dispatch, I scheduled a side-meeting to discuss =
this in more depth. Below are the side meeting details.<br><br><b>Time:</b>=
 Tuesday, 15:20 ~ 16:50<br><b>Place:</b> Room C2 (21st floor)<br><b>Organiz=
ers:</b> Victor Vasiliev, David Schinazi<br><b>Agenda:</b><br><ol><li>A mor=
e in-depth overview of WebTransport.</li><li>Discussion of the overall desi=
gn and the use cases.</li><li>As time permits, some of the major outstandin=
g design issues in Web transport, e.g.:</li><ul><li>What fallback protocol =
to use?</li><li>How to multiplex streams and datagrams within a single conn=
ection?</li><li>Can we let web applications do their own congestion control=
?</li><li>Should WebTransport sessions have URLs associated with them?</li>=
</ul></ol>As usual, here are some helpful links:<br><ul><li>WebTransport ov=
erview: <a href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-overv=
iew-00" target=3D"_blank">https://tools.ietf.org/html/draft-vvv-webtranspor=
t-overview-00</a></li><li>QuicTransport: <a href=3D"https://tools.ietf.org/=
html/draft-vvv-webtransport-quic-00" target=3D"_blank">https://tools.ietf.o=
rg/html/draft-vvv-webtransport-quic-00</a></li><li>Http3Transport: <a href=
=3D"https://tools.ietf.org/html/draft-vvv-webtransport-http3-00" target=3D"=
_blank">https://tools.ietf.org/html/draft-vvv-webtransport-http3-00</a></li=
><li>Web API Spec draft: <a href=3D"https://wicg.github.io/web-transport/" =
target=3D"_blank">https://wicg.github.io/web-transport/</a></li><li>Discuss=
ion on use cases: <a href=3D"https://discourse.wicg.io/t/webtransport-propo=
sal/3508" target=3D"_blank">https://discourse.wicg.io/t/webtransport-propos=
al/3508</a></li></ul>Cheers,<br>=C2=A0 Victor.<br></div>
</blockquote></div>

--00000000000003b8ff058e5ff839--


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Hi,

The draft minutes from this week=E2=80=99s session are up on the =
materials page[1]. Please send any comments or corrections to the =
DISPATCH list as soon as possible.

[1] =
https://datatracker.ietf.org/meeting/105/materials/minutes-105-dispatch-01=


Thanks!

Ben.

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A complete nit, but it was hard to read:

    Why replace SIP is it is 60% of interconnected VoIP?

I think should be

    Why replace SIP as it is 60% of interconnected VoIP?
                    ^^
(maybe "if") but I'm not sure.

Dale


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From: Vijay Gurbani <vijay.gurbani@gmail.com>
Date: Mon, 29 Jul 2019 09:26:01 -0500
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Dale: Hi.  Indeed, s/SIP is is is/SIP if it is/

My (typo) bad.

Cheers,

- vijay

On Sun, Jul 28, 2019 at 4:04 PM Dale R. Worley <worley@ariadne.com> wrote:

> A complete nit, but it was hard to read:
>
>     Why replace SIP is it is 60% of interconnected VoIP?
>
> I think should be
>
>     Why replace SIP as it is 60% of interconnected VoIP?
>                     ^^
> (maybe "if") but I'm not sure.
>
> Dale
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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<div dir=3D"ltr"><div>Dale: Hi.=C2=A0 Indeed, s/SIP is is is/SIP if it is/<=
/div><div><br></div><div>My (typo) bad.</div><div><br></div><div>Cheers,</d=
iv><div><br></div><div>- vijay<br></div></div><br><div class=3D"gmail_quote=
"><div dir=3D"ltr" class=3D"gmail_attr">On Sun, Jul 28, 2019 at 4:04 PM Dal=
e R. Worley &lt;<a href=3D"mailto:worley@ariadne.com">worley@ariadne.com</a=
>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px=
 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">A c=
omplete nit, but it was hard to read:<br>
<br>
=C2=A0 =C2=A0 Why replace SIP is it is 60% of interconnected VoIP?<br>
<br>
I think should be<br>
<br>
=C2=A0 =C2=A0 Why replace SIP as it is 60% of interconnected VoIP?<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 ^^<br=
>
(maybe &quot;if&quot;) but I&#39;m not sure.<br>
<br>
Dale<br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"noreferre=
r" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div>

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From: Victor Vasiliev <vasilvv@google.com>
Date: Mon, 29 Jul 2019 16:58:23 -0400
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Hello everyone,

Judging from the in-person discussions in Montreal, there appears to be a
decent amount of interest in the WebTransport work.  Everyone interested is
invited to join our mailing list:

  https://www.ietf.org/mailman/listinfo/webtransport

Cheers,
  Victor.

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<div dir=3D"ltr"><div>Hello everyone,</div><div><br></div><div>Judging from=
 the in-person discussions in Montreal, there appears to be a decent amount=
 of interest in the WebTransport work.=C2=A0 Everyone interested is invited=
 to join our mailing list:<br></div><div><br></div><div>=C2=A0=C2=A0<a href=
=3D"https://www.ietf.org/mailman/listinfo/webtransport">https://www.ietf.or=
g/mailman/listinfo/webtransport</a></div><div><br></div><div>Cheers,</div><=
div>=C2=A0 Victor.</div></div>

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Hi Victor,

Sorry if this has been covered, but for those of us out of the room, =
could you provide two or three sentences as to what the mailing list is =
supposed to cover?  Is this more BCP 190 stuff or something more generic =
or just something else?

Thanks,

Eliot

> On 29 Jul 2019, at 22:58, Victor Vasiliev =
<vasilvv=3D40google.com@dmarc.ietf.org> wrote:
>=20
> Hello everyone,
>=20
> Judging from the in-person discussions in Montreal, there appears to =
be a decent amount of interest in the WebTransport work.  Everyone =
interested is invited to join our mailing list:
>=20
>   https://www.ietf.org/mailman/listinfo/webtransport =
<https://www.ietf.org/mailman/listinfo/webtransport>
>=20
> Cheers,
>   Victor.


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D"">Hi =
Victor,<div class=3D""><br class=3D""></div><div class=3D"">Sorry if =
this has been covered, but for those of us out of the room, could you =
provide two or three sentences as to what the mailing list is supposed =
to cover? &nbsp;Is this more BCP 190 stuff or something more generic or =
just something else?</div><div class=3D""><br class=3D""></div><div =
class=3D"">Thanks,</div><div class=3D""><br class=3D""></div><div =
class=3D"">Eliot</div><div class=3D""><div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D"">On 29 Jul 2019, at 22:58, =
Victor Vasiliev &lt;<a href=3D"mailto:vasilvv=3D40google.com@dmarc.ietf.or=
g" class=3D"">vasilvv=3D40google.com@dmarc.ietf.org</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><meta =
http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dutf-8" =
class=3D""><div dir=3D"ltr" class=3D""><div class=3D"">Hello =
everyone,</div><div class=3D""><br class=3D""></div><div =
class=3D"">Judging from the in-person discussions in Montreal, there =
appears to be a decent amount of interest in the WebTransport =
work.&nbsp; Everyone interested is invited to join our mailing list:<br =
class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D"">&nbsp;&nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/webtransport" =
class=3D"">https://www.ietf.org/mailman/listinfo/webtransport</a></div><di=
v class=3D""><br class=3D""></div><div class=3D"">Cheers,</div><div =
class=3D"">&nbsp; Victor.</div></div>
</div></blockquote></div><br class=3D""></div></body></html>=

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In-Reply-To: <85E74088-6D84-47CB-97CC-C637167959B0@cisco.com>
From: Victor Vasiliev <vasilvv@google.com>
Date: Mon, 29 Jul 2019 17:32:30 -0400
Message-ID: <CAAZdMaff4y7s=SfajaPxjfqGqnNQ3trcqxO+Xofq2-M35s7-FA@mail.gmail.com>
To: Eliot Lear <lear@cisco.com>
Cc: dispatch@ietf.org, IETF QUIC WG <quic@ietf.org>,  HTTP Working Group <ietf-http-wg@w3.org>, hybi@ietf.org, webtransport@ietf.org
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Archived-At: <https://mailarchive.ietf.org/arch/msg/dispatch/2j3H4-BVJvK3MqDJKvPN0iE47TU>
Subject: Re: [dispatch] WebTransport Mailing List
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Hi Eliot,

WebTransport is roughly "WebSockets, but for unreliable transport",
currently described in
https://tools.ietf.org/html/draft-vvv-webtransport-overview-00 and
https://wicg.github.io/web-transport/.  It's not related to BCP 190
discussion that happened at the other portion of ARTAREA meeting.

Cheers,
  Victor.

On Mon, Jul 29, 2019 at 5:29 PM Eliot Lear <lear@cisco.com> wrote:

> Hi Victor,
>
> Sorry if this has been covered, but for those of us out of the room, could
> you provide two or three sentences as to what the mailing list is supposed
> to cover?  Is this more BCP 190 stuff or something more generic or just
> something else?
>
> Thanks,
>
> Eliot
>
> On 29 Jul 2019, at 22:58, Victor Vasiliev <
> vasilvv=40google.com@dmarc.ietf.org> wrote:
>
> Hello everyone,
>
> Judging from the in-person discussions in Montreal, there appears to be a
> decent amount of interest in the WebTransport work.  Everyone interested is
> invited to join our mailing list:
>
>   https://www.ietf.org/mailman/listinfo/webtransport
>
> Cheers,
>   Victor.
>
>
>

--000000000000c49c62058ed8a126
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<div dir=3D"ltr">Hi Eliot,<div><br></div><div>WebTransport is roughly &quot=
;WebSockets, but for unreliable transport&quot;, currently described in=C2=
=A0<a href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-overview-0=
0">https://tools.ietf.org/html/draft-vvv-webtransport-overview-00</a>=C2=A0=
and=C2=A0<a href=3D"https://wicg.github.io/web-transport/">https://wicg.git=
hub.io/web-transport/</a>.=C2=A0 It&#39;s not related to BCP 190 discussion=
 that happened at the other portion of ARTAREA meeting.</div><div><br></div=
><div>Cheers,</div><div>=C2=A0 Victor.</div></div><br><div class=3D"gmail_q=
uote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Jul 29, 2019 at 5:29 PM=
 Eliot Lear &lt;<a href=3D"mailto:lear@cisco.com">lear@cisco.com</a>&gt; wr=
ote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=
=3D"overflow-wrap: break-word;">Hi Victor,<div><br></div><div>Sorry if this=
 has been covered, but for those of us out of the room, could you provide t=
wo or three sentences as to what the mailing list is supposed to cover?=C2=
=A0 Is this more BCP 190 stuff or something more generic or just something =
else?</div><div><br></div><div>Thanks,</div><div><br></div><div>Eliot</div>=
<div><div><br><blockquote type=3D"cite"><div>On 29 Jul 2019, at 22:58, Vict=
or Vasiliev &lt;<a href=3D"mailto:vasilvv=3D40google.com@dmarc.ietf.org" ta=
rget=3D"_blank">vasilvv=3D40google.com@dmarc.ietf.org</a>&gt; wrote:</div><=
br class=3D"gmail-m_-8947632560352852245Apple-interchange-newline"><div><di=
v dir=3D"ltr"><div>Hello everyone,</div><div><br></div><div>Judging from th=
e in-person discussions in Montreal, there appears to be a decent amount of=
 interest in the WebTransport work.=C2=A0 Everyone interested is invited to=
 join our mailing list:<br></div><div><br></div><div>=C2=A0=C2=A0<a href=3D=
"https://www.ietf.org/mailman/listinfo/webtransport" target=3D"_blank">http=
s://www.ietf.org/mailman/listinfo/webtransport</a></div><div><br></div><div=
>Cheers,</div><div>=C2=A0 Victor.</div></div>
</div></blockquote></div><br></div></div></blockquote></div>

--000000000000c49c62058ed8a126--


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From: Mark Nottingham <mnot@mnot.net>
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Cc: Victor Vasiliev <vasilvv=40google.com@dmarc.ietf.org>, Eliot Lear <lear@cisco.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "webtransport@ietf.org" <webtransport@ietf.org>, "hybi@ietf.org" <hybi@ietf.org>, HTTP Working Group <ietf-http-wg@w3.org>, IETF QUIC WG <quic@ietf.org>
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+1.=20

Naming things is hard, but when they're generic and confusable with =
other things, it's worth the pain.


> On 29 Jul 2019, at 11:53 pm, Martin J. D=C3=BCrst =
<duerst@it.aoyama.ac.jp> wrote:
>=20
> I think Eliot's question shows that "WebTransport" may greatly benefit=20=

> from a better name (or acronym).
>=20
> Regards,   Martin.
>=20
> P.S.: I already subscribed to the list.
>=20
> On 2019/07/30 06:32, Victor Vasiliev wrote:
>> Hi Eliot,
>>=20
>> WebTransport is roughly "WebSockets, but for unreliable transport",
>> currently described in
>> https://tools.ietf.org/html/draft-vvv-webtransport-overview-00 and
>> https://wicg.github.io/web-transport/.  It's not related to BCP 190
>> discussion that happened at the other portion of ARTAREA meeting.
>>=20
>> Cheers,
>>   Victor.
>>=20
>> On Mon, Jul 29, 2019 at 5:29 PM Eliot Lear <lear@cisco.com> wrote:
>>=20
>>> Hi Victor,
>>>=20
>>> Sorry if this has been covered, but for those of us out of the room, =
could
>>> you provide two or three sentences as to what the mailing list is =
supposed
>>> to cover?  Is this more BCP 190 stuff or something more generic or =
just
>>> something else?
>>>=20
>>> Thanks,
>>>=20
>>> Eliot
>>>=20
>>> On 29 Jul 2019, at 22:58, Victor Vasiliev <
>>> vasilvv=3D40google.com@dmarc.ietf.org> wrote:
>>>=20
>>> Hello everyone,
>>>=20
>>> Judging from the in-person discussions in Montreal, there appears to =
be a
>>> decent amount of interest in the WebTransport work.  Everyone =
interested is
>>> invited to join our mailing list:
>>>=20
>>>   https://www.ietf.org/mailman/listinfo/webtransport
>>>=20
>>> Cheers,
>>>   Victor.
>>>=20
>>>=20
>>>=20
>>=20
>>=20
>> _______________________________________________
>> hybi mailing list
>> hybi@ietf.org
>> https://www.ietf.org/mailman/listinfo/hybi
>>=20
>=20
> --=20
> Prof. Dr.sc. Martin J. D=C3=BCrst
> Department of Intelligent Information Technology
> College of Science and Engineering
> Aoyama Gakuin University
> Fuchinobe 5-1-10, Chuo-ku, Sagamihara
> 252-5258 Japan
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

--
Mark Nottingham   https://www.mnot.net/


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--0000000000005284d9058f001741
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What's wrong with "WebTransport"?  It seems like a good name to me.  With
what is it confusable?

On Wed, Jul 31, 2019 at 6:50 AM Mark Nottingham <mnot@mnot.net> wrote:

> +1.
>
> Naming things is hard, but when they're generic and confusable with other
> things, it's worth the pain.
>
>
> > On 29 Jul 2019, at 11:53 pm, Martin J. D=C3=BCrst <duerst@it.aoyama.ac.=
jp>
> wrote:
> >
> > I think Eliot's question shows that "WebTransport" may greatly benefit
> > from a better name (or acronym).
> >
> > Regards,   Martin.
> >
> > P.S.: I already subscribed to the list.
> >
> > On 2019/07/30 06:32, Victor Vasiliev wrote:
> >> Hi Eliot,
> >>
> >> WebTransport is roughly "WebSockets, but for unreliable transport",
> >> currently described in
> >> https://tools.ietf.org/html/draft-vvv-webtransport-overview-00 and
> >> https://wicg.github.io/web-transport/.  It's not related to BCP 190
> >> discussion that happened at the other portion of ARTAREA meeting.
> >>
> >> Cheers,
> >>   Victor.
> >>
> >> On Mon, Jul 29, 2019 at 5:29 PM Eliot Lear <lear@cisco.com> wrote:
> >>
> >>> Hi Victor,
> >>>
> >>> Sorry if this has been covered, but for those of us out of the room,
> could
> >>> you provide two or three sentences as to what the mailing list is
> supposed
> >>> to cover?  Is this more BCP 190 stuff or something more generic or ju=
st
> >>> something else?
> >>>
> >>> Thanks,
> >>>
> >>> Eliot
> >>>
> >>> On 29 Jul 2019, at 22:58, Victor Vasiliev <
> >>> vasilvv=3D40google.com@dmarc.ietf.org> wrote:
> >>>
> >>> Hello everyone,
> >>>
> >>> Judging from the in-person discussions in Montreal, there appears to
> be a
> >>> decent amount of interest in the WebTransport work.  Everyone
> interested is
> >>> invited to join our mailing list:
> >>>
> >>>   https://www.ietf.org/mailman/listinfo/webtransport
> >>>
> >>> Cheers,
> >>>   Victor.
> >>>
> >>>
> >>>
> >>
> >>
> >> _______________________________________________
> >> hybi mailing list
> >> hybi@ietf.org
> >> https://www.ietf.org/mailman/listinfo/hybi
> >>
> >
> > --
> > Prof. Dr.sc. Martin J. D=C3=BCrst
> > Department of Intelligent Information Technology
> > College of Science and Engineering
> > Aoyama Gakuin University
> > Fuchinobe 5-1-10, Chuo-ku, Sagamihara
> > 252-5258 Japan
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>
> --
> Mark Nottingham   https://www.mnot.net/
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

--0000000000005284d9058f001741
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>What&#39;s wrong with &quot;WebTransport&quot;?=C2=A0=
 It seems like a good name to me.=C2=A0 With what is it confusable?</div></=
div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On=
 Wed, Jul 31, 2019 at 6:50 AM Mark Nottingham &lt;<a href=3D"mailto:mnot@mn=
ot.net">mnot@mnot.net</a>&gt; wrote:<br></div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,20=
4);padding-left:1ex">+1. <br>
<br>
Naming things is hard, but when they&#39;re generic and confusable with oth=
er things, it&#39;s worth the pain.<br>
<br>
<br>
&gt; On 29 Jul 2019, at 11:53 pm, Martin J. D=C3=BCrst &lt;<a href=3D"mailt=
o:duerst@it.aoyama.ac.jp" target=3D"_blank">duerst@it.aoyama.ac.jp</a>&gt; =
wrote:<br>
&gt; <br>
&gt; I think Eliot&#39;s question shows that &quot;WebTransport&quot; may g=
reatly benefit <br>
&gt; from a better name (or acronym).<br>
&gt; <br>
&gt; Regards,=C2=A0 =C2=A0Martin.<br>
&gt; <br>
&gt; P.S.: I already subscribed to the list.<br>
&gt; <br>
&gt; On 2019/07/30 06:32, Victor Vasiliev wrote:<br>
&gt;&gt; Hi Eliot,<br>
&gt;&gt; <br>
&gt;&gt; WebTransport is roughly &quot;WebSockets, but for unreliable trans=
port&quot;,<br>
&gt;&gt; currently described in<br>
&gt;&gt; <a href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-over=
view-00" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/d=
raft-vvv-webtransport-overview-00</a> and<br>
&gt;&gt; <a href=3D"https://wicg.github.io/web-transport/" rel=3D"noreferre=
r" target=3D"_blank">https://wicg.github.io/web-transport/</a>.=C2=A0 It&#3=
9;s not related to BCP 190<br>
&gt;&gt; discussion that happened at the other portion of ARTAREA meeting.<=
br>
&gt;&gt; <br>
&gt;&gt; Cheers,<br>
&gt;&gt;=C2=A0 =C2=A0Victor.<br>
&gt;&gt; <br>
&gt;&gt; On Mon, Jul 29, 2019 at 5:29 PM Eliot Lear &lt;<a href=3D"mailto:l=
ear@cisco.com" target=3D"_blank">lear@cisco.com</a>&gt; wrote:<br>
&gt;&gt; <br>
&gt;&gt;&gt; Hi Victor,<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Sorry if this has been covered, but for those of us out of the=
 room, could<br>
&gt;&gt;&gt; you provide two or three sentences as to what the mailing list=
 is supposed<br>
&gt;&gt;&gt; to cover?=C2=A0 Is this more BCP 190 stuff or something more g=
eneric or just<br>
&gt;&gt;&gt; something else?<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Thanks,<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Eliot<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; On 29 Jul 2019, at 22:58, Victor Vasiliev &lt;<br>
&gt;&gt;&gt; vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" targe=
t=3D"_blank">40google.com@dmarc.ietf.org</a>&gt; wrote:<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Hello everyone,<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Judging from the in-person discussions in Montreal, there appe=
ars to be a<br>
&gt;&gt;&gt; decent amount of interest in the WebTransport work.=C2=A0 Ever=
yone interested is<br>
&gt;&gt;&gt; invited to join our mailing list:<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt;=C2=A0 =C2=A0<a href=3D"https://www.ietf.org/mailman/listinfo/w=
ebtransport" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mail=
man/listinfo/webtransport</a><br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Cheers,<br>
&gt;&gt;&gt;=C2=A0 =C2=A0Victor.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; hybi mailing list<br>
&gt;&gt; <a href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</=
a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/hybi" rel=3D"nore=
ferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><br=
>
&gt;&gt; <br>
&gt; <br>
&gt; -- <br>
&gt; Prof. Dr.sc. Martin J. D=C3=BCrst<br>
&gt; Department of Intelligent Information Technology<br>
&gt; College of Science and Engineering<br>
&gt; Aoyama Gakuin University<br>
&gt; Fuchinobe 5-1-10, Chuo-ku, Sagamihara<br>
&gt; 252-5258 Japan<br>
&gt; _______________________________________________<br>
&gt; dispatch mailing list<br>
&gt; <a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.o=
rg</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"nore=
ferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a=
><br>
<br>
--<br>
Mark Nottingham=C2=A0 =C2=A0<a href=3D"https://www.mnot.net/" rel=3D"norefe=
rrer" target=3D"_blank">https://www.mnot.net/</a><br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"noreferre=
r" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div>

--0000000000005284d9058f001741--


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On July 29, 2019 9:58:23 PM GMT+01:00, Victor Vasiliev <vasilvv=3D40google=
=2Ecom@dmarc=2Eietf=2Eorg> wrote:
>Hello everyone,
>
>Judging from the in-person discussions in Montreal, there appears to be
>a

Could you find a moment in your busy schedule to respond to what I wrote o=
n this list before making your own new and exciting list, esp wrt this shou=
ld find a way to target h2?

-Andy

>decent amount of interest in the WebTransport work=2E  Everyone
>interested is
>invited to join our mailing list:
>
>  https://www=2Eietf=2Eorg/mailman/listinfo/webtransport
>
>Cheers,
>  Victor=2E


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To: Peter Thatcher <pthatcher=40google.com@dmarc.ietf.org>, Mark Nottingham <mnot@mnot.net>
Cc: "webtransport@ietf.org" <webtransport@ietf.org>, IETF QUIC WG <quic@ietf.org>,  Victor Vasiliev <vasilvv=40google.com@dmarc.ietf.org>,  HTTP Working Group <ietf-http-wg@w3.org>, =?UTF-8?Q?Martin_J=2E_D=C3=BCrst?= <duerst@it.aoyama.ac.jp>,  "dispatch@ietf.org" <dispatch@ietf.org>, "hybi@ietf.org" <hybi@ietf.org>, Eliot Lear <lear@cisco.com>
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Same as for WebAssembly and WebSocket when not used with a browser with
HTTP.

For example MQTT over WebSocket to penetrate firewalls has very little to
do with Web.

On 31 July 2019 at 22.37.39, Peter Thatcher (
pthatcher=3D40google.com@dmarc.ietf.org) wrote:

What's wrong with "WebTransport"?  It seems like a good name to me.  With
what is it confusable?

On Wed, Jul 31, 2019 at 6:50 AM Mark Nottingham <mnot@mnot.net> wrote:

> +1.
>
> Naming things is hard, but when they're generic and confusable with other
> things, it's worth the pain.
>
>
> > On 29 Jul 2019, at 11:53 pm, Martin J. D=C3=BCrst <duerst@it.aoyama.ac.=
jp>
> wrote:
> >
> > I think Eliot's question shows that "WebTransport" may greatly benefit
> > from a better name (or acronym).
> >
> > Regards,   Martin.
> >
> > P.S.: I already subscribed to the list.
> >
> > On 2019/07/30 06:32, Victor Vasiliev wrote:
> >> Hi Eliot,
> >>
> >> WebTransport is roughly "WebSockets, but for unreliable transport",
> >> currently described in
> >> https://tools.ietf.org/html/draft-vvv-webtransport-overview-00 and
> >> https://wicg.github.io/web-transport/.  It's not related to BCP 190
> >> discussion that happened at the other portion of ARTAREA meeting.
> >>
> >> Cheers,
> >>   Victor.
> >>
> >> On Mon, Jul 29, 2019 at 5:29 PM Eliot Lear <lear@cisco.com> wrote:
> >>
> >>> Hi Victor,
> >>>
> >>> Sorry if this has been covered, but for those of us out of the room,
> could
> >>> you provide two or three sentences as to what the mailing list is
> supposed
> >>> to cover?  Is this more BCP 190 stuff or something more generic or ju=
st
> >>> something else?
> >>>
> >>> Thanks,
> >>>
> >>> Eliot
> >>>
> >>> On 29 Jul 2019, at 22:58, Victor Vasiliev <
> >>> vasilvv=3D40google.com@dmarc.ietf.org> wrote:
> >>>
> >>> Hello everyone,
> >>>
> >>> Judging from the in-person discussions in Montreal, there appears to
> be a
> >>> decent amount of interest in the WebTransport work.  Everyone
> interested is
> >>> invited to join our mailing list:
> >>>
> >>>   https://www.ietf.org/mailman/listinfo/webtransport
> >>>
> >>> Cheers,
> >>>   Victor.
> >>>
> >>>
> >>>
> >>
> >>
> >> _______________________________________________
> >> hybi mailing list
> >> hybi@ietf.org
> >> https://www.ietf.org/mailman/listinfo/hybi
> >>
> >
> > --
> > Prof. Dr.sc. Martin J. D=C3=BCrst
> > Department of Intelligent Information Technology
> > College of Science and Engineering
> > Aoyama Gakuin University
> > Fuchinobe 5-1-10, Chuo-ku, Sagamihara
> > 252-5258 Japan
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>
> --
> Mark Nottingham   https://www.mnot.net/
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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<html><head><style>body{font-family:Helvetica,Arial;font-size:13px}</style>=
</head><body style=3D"word-wrap:break-word;line-break:after-white-space"><d=
iv id=3D"bloop_customfont" style=3D"font-family:Helvetica,Arial;font-size:1=
3px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto">Same as for WebAssem=
bly and WebSocket when not used with a browser with HTTP.</div><div id=3D"b=
loop_customfont" style=3D"font-family:Helvetica,Arial;font-size:13px;color:=
rgba(0,0,0,1.0);margin:0px;line-height:auto"><br></div><div id=3D"bloop_cus=
tomfont" style=3D"font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0=
,0,1.0);margin:0px;line-height:auto">For example MQTT over WebSocket to pen=
etrate firewalls has very little to do with Web.</div> <br><p class=3D"airm=
ail_on">On 31 July 2019 at 22.37.39, Peter Thatcher (<a href=3D"mailto:ptha=
tcher=3D40google.com@dmarc.ietf.org">pthatcher=3D40google.com@dmarc.ietf.or=
g</a>) wrote:</p> <blockquote type=3D"cite" class=3D"clean_bq"><span><div><=
div></div><div>


<title></title>


<div dir=3D"ltr">
<div>What&#39;s wrong with &quot;WebTransport&quot;?=C2=A0 It seems like a =
good
name to me.=C2=A0 With what is it confusable?</div>
</div>
<br>
<div class=3D"gmail_quote">
<div dir=3D"ltr" class=3D"gmail_attr">On Wed, Jul 31, 2019 at 6:50 AM
Mark Nottingham &lt;<a href=3D"mailto:mnot@mnot.net">mnot@mnot.net</a>&gt; =
wrote:<br></div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
+1.<br>
<br>
Naming things is hard, but when they&#39;re generic and confusable with
other things, it&#39;s worth the pain.<br>
<br>
<br>
&gt; On 29 Jul 2019, at 11:53 pm, Martin J. D=C3=BCrst &lt;<a href=3D"mailt=
o:duerst@it.aoyama.ac.jp" target=3D"_blank">duerst@it.aoyama.ac.jp</a>&gt; =
wrote:<br>
&gt;<br>
&gt; I think Eliot&#39;s question shows that &quot;WebTransport&quot; may g=
reatly
benefit<br>
&gt; from a better name (or acronym).<br>
&gt;<br>
&gt; Regards,=C2=A0 =C2=A0Martin.<br>
&gt;<br>
&gt; P.S.: I already subscribed to the list.<br>
&gt;<br>
&gt; On 2019/07/30 06:32, Victor Vasiliev wrote:<br>
&gt;&gt; Hi Eliot,<br>
&gt;&gt;<br>
&gt;&gt; WebTransport is roughly &quot;WebSockets, but for unreliable
transport&quot;,<br>
&gt;&gt; currently described in<br>
&gt;&gt; <a href=3D"https://tools.ietf.org/html/draft-vvv-webtransport-over=
view-00" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/d=
raft-vvv-webtransport-overview-00</a>
and<br>
&gt;&gt; <a href=3D"https://wicg.github.io/web-transport/" rel=3D"noreferre=
r" target=3D"_blank">https://wicg.github.io/web-transport/</a>.=C2=A0 It&#3=
9;s not
related to BCP 190<br>
&gt;&gt; discussion that happened at the other portion of ARTAREA
meeting.<br>
&gt;&gt;<br>
&gt;&gt; Cheers,<br>
&gt;&gt;=C2=A0 =C2=A0Victor.<br>
&gt;&gt;<br>
&gt;&gt; On Mon, Jul 29, 2019 at 5:29 PM Eliot Lear &lt;<a href=3D"mailto:l=
ear@cisco.com" target=3D"_blank">lear@cisco.com</a>&gt;
wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; Hi Victor,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Sorry if this has been covered, but for those of us
out of the room, could<br>
&gt;&gt;&gt; you provide two or three sentences as to what the
mailing list is supposed<br>
&gt;&gt;&gt; to cover?=C2=A0 Is this more BCP 190 stuff or
something more generic or just<br>
&gt;&gt;&gt; something else?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Thanks,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Eliot<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On 29 Jul 2019, at 22:58, Victor Vasiliev &lt;<br>
&gt;&gt;&gt; vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" targe=
t=3D"_blank">40google.com@dmarc.ietf.org</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Hello everyone,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Judging from the in-person discussions in Montreal,
there appears to be a<br>
&gt;&gt;&gt; decent amount of interest in the WebTransport
work.=C2=A0 Everyone interested is<br>
&gt;&gt;&gt; invited to join our mailing list:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;=C2=A0 =C2=A0<a href=3D"https://www.ietf.org/mailman/listinfo/w=
ebtransport" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mail=
man/listinfo/webtransport</a><br>

&gt;&gt;&gt;<br>
&gt;&gt;&gt; Cheers,<br>
&gt;&gt;&gt;=C2=A0 =C2=A0Victor.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; hybi mailing list<br>
&gt;&gt; <a href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</=
a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/hybi" rel=3D"nore=
ferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><br=
>
&gt;&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Prof. Dr.sc. Martin J. D=C3=BCrst<br>
&gt; Department of Intelligent Information Technology<br>
&gt; College of Science and Engineering<br>
&gt; Aoyama Gakuin University<br>
&gt; Fuchinobe 5-1-10, Chuo-ku, Sagamihara<br>
&gt; 252-5258 Japan<br>
&gt; _______________________________________________<br>
&gt; dispatch mailing list<br>
&gt; <a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.o=
rg</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" rel=3D"nore=
ferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a=
><br>
<br>
--<br>
Mark Nottingham=C2=A0 =C2=A0<a href=3D"https://www.mnot.net/" rel=3D"norefe=
rrer" target=3D"_blank">https://www.mnot.net/</a><br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
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r" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>=
</blockquote>
</div>


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