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From: Jonathan Rosenberg <jdrosen@five9.com>
To: "Roni Even (A)" <roni.even@huawei.com>, "v3@ietf.org" <v3@ietf.org>
CC: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: RIPT BoF approved for IETF 107 - Draft charter below
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Date: Mon, 2 Mar 2020 16:23:16 +0000
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Archived-At: <https://mailarchive.ietf.org/arch/msg/v3/WDcYkW_vNeac39FVORApBLDAIgQ>
Subject: Re: [V3] RIPT BoF approved for IETF 107 - Draft charter below
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Thanks Roni. I think its clear we want datagrams for media in general. The =
trick here is how to get them while utilizing http3 as the transport.

From: Roni Even (A) <roni.even@huawei.com>
Sent: Saturday, February 29, 2020 2:51 AM
To: Jonathan Rosenberg <jdrosen@five9.com>; v3@ietf.org
Cc: Cullen Jennings (fluffy) <fluffy@cisco.com>
Subject: RE: RIPT BoF approved for IETF 107 - Draft charter below

Hi,
I read the drafts. I like the idea of having the endpoint capabilities know=
n in advance making call establishment simpler. We have the must support co=
decs in RTCweb and this is a step forwards. Maybe it is also about time to =
deprecate the "MUST support H.261/G.711" in older documents (H.323,...).
As for media transport, I think that Jorge and Colin document provide good =
starting point and motivation why to use datagrams for the media and it is =
my preference for conversional services to have the application layer decid=
e if to retransmit.  My view is that for audio it is not needed (use redund=
ancy, the best retransmission is for the listener to say "I could not hear =
it, can you please repeat".
As for video my view is that the requirement is not retransmission but re-s=
ynchronize. For "talking heads" video conference just resynch and continue.

Roni


From: V3 [mailto:v3-bounces@ietf.org] On Behalf Of Jonathan Rosenberg
Sent: Saturday, February 15, 2020 12:00 AM
To: v3@ietf.org<mailto:v3@ietf.org>
Cc: Cullen Jennings (fluffy)
Subject: [V3] RIPT BoF approved for IETF 107 - Draft charter below

Good news folks - RIPT bof was approved for IETF107 and we'll be meeting. G=
oal is to try and get a WG formed. As such, charter discussion is in order.=
 Cullen, myself and Justin worked a draft charter. Comments please:


This working group will standardize a protocol, capable of operating
atop HTTP/3, which supports real-time voice and video communications,
including signaling, media negotiation, and transmission of media
packets.

The primary rationale for this new protocol is to enable deployment of
real-time communications services on modern "cloud" systems. These
systems are built around HTTP, and for HTTP-based applications, they
enable load balancing, HA, auto-scaling, and so on. However, real-time
communications protocols are based on SIP and RTP, and they cannot take
advantage of these HTTP-based capabilities.

It is a non-goal of this working group to replicate all of SIP and its
many extensions into HTTP. The group will limit itself to supporting the
functionality in widespread actual usage today.

The protocol uses HTTP techniques for authentication and authorization
(notably OAuth), and requires hop by hop encryption (i.e., https). The
protocol will also allow for e2e media encryption, although keying is
out of scope, and is expected to be handled by other protocols such as
MLS. This extension will also utilize STIR for callerID.

This protocol should be implementable in browsers, thick desktop
clients, mobile apps and servers.

The group will standardize an extension to the baseline protocol that
automates the configuration necessary to achieve calling between on
organization which is a customer of the other (for example, enterprise
to service provider).

[OPEN ISSUE: Is P2P media in or out? If in, ICE or something else ?]

The group will do its work in conjunction with active software
development efforts, so that implementation experience feeds directly
into protocol development.

The group will coordinate with the Webtranport, QUIC, HTTPbis and STIR
working groups.

Milestones:

Dec 2020: Submit baseline protocol to IESG
Mar 2021: Submit autoconfiguration protocol to IESG

Thx,
Jonathan R.

--
Jonathan Rosenberg
CTO and Head of AI, Five9


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<p class=3D"MsoNormal">Thanks Roni. I think its clear we want datagrams for=
 media in general. The trick here is how to get them while utilizing http3 =
as the transport.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b>From:</b> Roni Even (A) &lt;roni.even@huawei.com&=
gt; <br>
<b>Sent:</b> Saturday, February 29, 2020 2:51 AM<br>
<b>To:</b> Jonathan Rosenberg &lt;jdrosen@five9.com&gt;; v3@ietf.org<br>
<b>Cc:</b> Cullen Jennings (fluffy) &lt;fluffy@cisco.com&gt;<br>
<b>Subject:</b> RE: RIPT BoF approved for IETF 107 - Draft charter below<o:=
p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi,<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I read the drafts. I l=
ike the idea of having the endpoint capabilities known in advance making ca=
ll establishment simpler. We have the must support codecs in RTCweb and thi=
s is a step forwards. Maybe it is also
 about time to deprecate the &#8220;MUST support H.261/G.711&#8221; in olde=
r documents (H.323,&#8230;).
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">As for media transport=
, I think that Jorge and Colin document provide good starting point and mot=
ivation why to use datagrams for the media and it is my preference for conv=
ersional services to have the application
 layer decide if to retransmit.&nbsp; My view is that for audio it is not n=
eeded (use redundancy, the best retransmission is for the listener to say &=
#8220;I could not hear it, can you please repeat&#8221;.<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">As for video my view i=
s that the requirement is not retransmission but re-synchronize. For &#8220=
;talking heads&#8221; video conference just resynch and continue.<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Roni &nbsp;<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
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<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
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<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,sans-serif">From:</span></b><span style=3D"font-size:10.0pt;f=
ont-family:&quot;Tahoma&quot;,sans-serif"> V3 [<a href=3D"mailto:v3-bounces=
@ietf.org">mailto:v3-bounces@ietf.org</a>]
<b>On Behalf Of </b>Jonathan Rosenberg<br>
<b>Sent:</b> Saturday, February 15, 2020 12:00 AM<br>
<b>To:</b> <a href=3D"mailto:v3@ietf.org">v3@ietf.org</a><br>
<b>Cc:</b> Cullen Jennings (fluffy)<br>
<b>Subject:</b> [V3] RIPT BoF approved for IETF 107 - Draft charter below<o=
:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Good news folks &#8211; RIPT bof was approved for IE=
TF107 and we&#8217;ll be meeting. Goal is to try and get a WG formed. As su=
ch, charter discussion is in order. Cullen, myself and Justin worked a draf=
t charter. Comments please:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This working group will standardize a protocol, capa=
ble of operating<o:p></o:p></p>
<p class=3D"MsoNormal">atop HTTP/3, which supports real-time voice and vide=
o communications,<o:p></o:p></p>
<p class=3D"MsoNormal">including signaling, media negotiation, and transmis=
sion of media<o:p></o:p></p>
<p class=3D"MsoNormal">packets.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The primary rationale for this new protocol is to en=
able deployment of<o:p></o:p></p>
<p class=3D"MsoNormal">real-time communications services on modern &quot;cl=
oud&quot; systems. These<o:p></o:p></p>
<p class=3D"MsoNormal">systems are built around HTTP, and for HTTP-based ap=
plications, they<o:p></o:p></p>
<p class=3D"MsoNormal">enable load balancing, HA, auto-scaling, and so on. =
However, real-time<o:p></o:p></p>
<p class=3D"MsoNormal">communications protocols are based on SIP and RTP, a=
nd they cannot take<o:p></o:p></p>
<p class=3D"MsoNormal">advantage of these HTTP-based capabilities.<o:p></o:=
p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">It is a non-goal of this working group to replicate =
all of SIP and its<o:p></o:p></p>
<p class=3D"MsoNormal">many extensions into HTTP. The group will limit itse=
lf to supporting the<o:p></o:p></p>
<p class=3D"MsoNormal">functionality in widespread actual usage today.<o:p>=
</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The protocol uses HTTP techniques for authentication=
 and authorization<o:p></o:p></p>
<p class=3D"MsoNormal">(notably OAuth), and requires hop by hop encryption =
(i.e., https). The<o:p></o:p></p>
<p class=3D"MsoNormal">protocol will also allow for e2e media encryption, a=
lthough keying is<o:p></o:p></p>
<p class=3D"MsoNormal">out of scope, and is expected to be handled by other=
 protocols such as<o:p></o:p></p>
<p class=3D"MsoNormal">MLS. This extension will also utilize STIR for calle=
rID.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This protocol should be implementable in browsers, t=
hick desktop<o:p></o:p></p>
<p class=3D"MsoNormal">clients, mobile apps and servers.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The group will standardize an extension to the basel=
ine protocol that<o:p></o:p></p>
<p class=3D"MsoNormal">automates the configuration necessary to achieve cal=
ling between on<o:p></o:p></p>
<p class=3D"MsoNormal">organization which is a customer of the other (for e=
xample, enterprise<o:p></o:p></p>
<p class=3D"MsoNormal">to service provider).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">[OPEN ISSUE: Is P2P media in or out? If in, ICE or s=
omething else ?]<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The group will do its work in conjunction with activ=
e software<o:p></o:p></p>
<p class=3D"MsoNormal">development efforts, so that implementation experien=
ce feeds directly<o:p></o:p></p>
<p class=3D"MsoNormal">into protocol development.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The group will coordinate with the Webtranport, QUIC=
, HTTPbis and STIR<o:p></o:p></p>
<p class=3D"MsoNormal">working groups.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Milestones:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Dec 2020: Submit baseline protocol to IESG<o:p></o:p=
></p>
<p class=3D"MsoNormal">Mar 2021: Submit autoconfiguration protocol to IESG<=
o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thx,<o:p></o:p></p>
<p class=3D"MsoNormal">Jonathan R.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">--<o:p></o:p></p>
<p class=3D"MsoNormal">Jonathan Rosenberg<o:p></o:p></p>
<p class=3D"MsoNormal">CTO and Head of AI, Five9<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,serif"><o:p>&nbsp;</o:p></span></p>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"font-size:12.0pt;font-family:&quot;Times New Roman&quot;,serif">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Ari=
al&quot;,sans-serif;color:gray"><br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
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y acting, or refraining from acting,
 on any information contained in this e-mail is hereby excluded. If you are=
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From: "Roni Even (A)" <roni.even@huawei.com>
To: Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
CC: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: RIPT BoF approved for IETF 107 - Draft charter below
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Subject: Re: [V3] RIPT BoF approved for IETF 107 - Draft charter below
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Hi Jonathan,
I think we also need to look how to convey the RTP extensions that may be n=
eeded (RFC5104) and the RTCP feedback, I found Jorge and Colin suggestion t=
o use the QUIC reliable streams as a good direction, also unlike UDP when u=
sing QUIC you create a bi-directional connection.
I am not that good with HTTP3, I probably need to learn more
Regards
Roni

From: V3 [mailto:v3-bounces@ietf.org] On Behalf Of Jonathan Rosenberg
Sent: Monday, March 02, 2020 6:23 PM
To: Roni Even (A); v3@ietf.org
Cc: Cullen Jennings (fluffy)
Subject: Re: [V3] RIPT BoF approved for IETF 107 - Draft charter below

Thanks Roni. I think its clear we want datagrams for media in general. The =
trick here is how to get them while utilizing http3 as the transport.

From: Roni Even (A) <roni.even@huawei.com<mailto:roni.even@huawei.com>>
Sent: Saturday, February 29, 2020 2:51 AM
To: Jonathan Rosenberg <jdrosen@five9.com<mailto:jdrosen@five9.com>>; v3@ie=
tf.org<mailto:v3@ietf.org>
Cc: Cullen Jennings (fluffy) <fluffy@cisco.com<mailto:fluffy@cisco.com>>
Subject: RE: RIPT BoF approved for IETF 107 - Draft charter below

Hi,
I read the drafts. I like the idea of having the endpoint capabilities know=
n in advance making call establishment simpler. We have the must support co=
decs in RTCweb and this is a step forwards. Maybe it is also about time to =
deprecate the "MUST support H.261/G.711" in older documents (H.323,...).
As for media transport, I think that Jorge and Colin document provide good =
starting point and motivation why to use datagrams for the media and it is =
my preference for conversional services to have the application layer decid=
e if to retransmit.  My view is that for audio it is not needed (use redund=
ancy, the best retransmission is for the listener to say "I could not hear =
it, can you please repeat".
As for video my view is that the requirement is not retransmission but re-s=
ynchronize. For "talking heads" video conference just resynch and continue.

Roni


From: V3 [mailto:v3-bounces@ietf.org] On Behalf Of Jonathan Rosenberg
Sent: Saturday, February 15, 2020 12:00 AM
To: v3@ietf.org<mailto:v3@ietf.org>
Cc: Cullen Jennings (fluffy)
Subject: [V3] RIPT BoF approved for IETF 107 - Draft charter below

Good news folks - RIPT bof was approved for IETF107 and we'll be meeting. G=
oal is to try and get a WG formed. As such, charter discussion is in order.=
 Cullen, myself and Justin worked a draft charter. Comments please:


This working group will standardize a protocol, capable of operating
atop HTTP/3, which supports real-time voice and video communications,
including signaling, media negotiation, and transmission of media
packets.

The primary rationale for this new protocol is to enable deployment of
real-time communications services on modern "cloud" systems. These
systems are built around HTTP, and for HTTP-based applications, they
enable load balancing, HA, auto-scaling, and so on. However, real-time
communications protocols are based on SIP and RTP, and they cannot take
advantage of these HTTP-based capabilities.

It is a non-goal of this working group to replicate all of SIP and its
many extensions into HTTP. The group will limit itself to supporting the
functionality in widespread actual usage today.

The protocol uses HTTP techniques for authentication and authorization
(notably OAuth), and requires hop by hop encryption (i.e., https). The
protocol will also allow for e2e media encryption, although keying is
out of scope, and is expected to be handled by other protocols such as
MLS. This extension will also utilize STIR for callerID.

This protocol should be implementable in browsers, thick desktop
clients, mobile apps and servers.

The group will standardize an extension to the baseline protocol that
automates the configuration necessary to achieve calling between on
organization which is a customer of the other (for example, enterprise
to service provider).

[OPEN ISSUE: Is P2P media in or out? If in, ICE or something else ?]

The group will do its work in conjunction with active software
development efforts, so that implementation experience feeds directly
into protocol development.

The group will coordinate with the Webtranport, QUIC, HTTPbis and STIR
working groups.

Milestones:

Dec 2020: Submit baseline protocol to IESG
Mar 2021: Submit autoconfiguration protocol to IESG

Thx,
Jonathan R.

--
Jonathan Rosenberg
CTO and Head of AI, Five9


________________________________

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y acting, or refraining from acting, on any information contained in this e=
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hments and do not disclose the contents to any other person, use it for any=
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 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
.

________________________________

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hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
.

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Jonathan,<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I think we also need t=
o look how to convey the RTP extensions that may be needed (RFC5104) and th=
e RTCP feedback, I found Jorge and Colin suggestion to use the QUIC reliabl=
e streams as a good direction, also
 unlike UDP when using QUIC you create a bi-directional connection.<o:p></o=
:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I am not that good wit=
h HTTP3, I probably need to learn more<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Regards<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Roni<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<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;"> V3 [mail=
to:v3-bounces@ietf.org]
<b>On Behalf Of </b>Jonathan Rosenberg<br>
<b>Sent:</b> Monday, March 02, 2020 6:23 PM<br>
<b>To:</b> Roni Even (A); v3@ietf.org<br>
<b>Cc:</b> Cullen Jennings (fluffy)<br>
<b>Subject:</b> Re: [V3] RIPT BoF approved for IETF 107 - Draft charter bel=
ow<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks Roni. I think its clear we want datagrams for=
 media in general. The trick here is how to get them while utilizing http3 =
as the transport.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b>From:</b> Roni Even (A) &lt;<a href=3D"mailto:ron=
i.even@huawei.com">roni.even@huawei.com</a>&gt;
<br>
<b>Sent:</b> Saturday, February 29, 2020 2:51 AM<br>
<b>To:</b> Jonathan Rosenberg &lt;<a href=3D"mailto:jdrosen@five9.com">jdro=
sen@five9.com</a>&gt;;
<a href=3D"mailto:v3@ietf.org">v3@ietf.org</a><br>
<b>Cc:</b> Cullen Jennings (fluffy) &lt;<a href=3D"mailto:fluffy@cisco.com"=
>fluffy@cisco.com</a>&gt;<br>
<b>Subject:</b> RE: RIPT BoF approved for IETF 107 - Draft charter below<o:=
p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi,<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I read the drafts. I l=
ike the idea of having the endpoint capabilities known in advance making ca=
ll establishment simpler. We have the must support codecs in RTCweb and thi=
s is a step forwards. Maybe it is also
 about time to deprecate the &#8220;MUST support H.261/G.711&#8221; in olde=
r documents (H.323,&#8230;).
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">As for media transport=
, I think that Jorge and Colin document provide good starting point and mot=
ivation why to use datagrams for the media and it is my preference for conv=
ersional services to have the application
 layer decide if to retransmit.&nbsp; My view is that for audio it is not n=
eeded (use redundancy, the best retransmission is for the listener to say &=
#8220;I could not hear it, can you please repeat&#8221;.<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">As for video my view i=
s that the requirement is not retransmission but re-synchronize. For &#8220=
;talking heads&#8221; video conference just resynch and continue.<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Roni &nbsp;<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<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;"> V3 [<a h=
ref=3D"mailto:v3-bounces@ietf.org">mailto:v3-bounces@ietf.org</a>]
<b>On Behalf Of </b>Jonathan Rosenberg<br>
<b>Sent:</b> Saturday, February 15, 2020 12:00 AM<br>
<b>To:</b> <a href=3D"mailto:v3@ietf.org">v3@ietf.org</a><br>
<b>Cc:</b> Cullen Jennings (fluffy)<br>
<b>Subject:</b> [V3] RIPT BoF approved for IETF 107 - Draft charter below<o=
:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Good news folks &#8211; RIPT bof was approved for IE=
TF107 and we&#8217;ll be meeting. Goal is to try and get a WG formed. As su=
ch, charter discussion is in order. Cullen, myself and Justin worked a draf=
t charter. Comments please:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This working group will standardize a protocol, capa=
ble of operating<o:p></o:p></p>
<p class=3D"MsoNormal">atop HTTP/3, which supports real-time voice and vide=
o communications,<o:p></o:p></p>
<p class=3D"MsoNormal">including signaling, media negotiation, and transmis=
sion of media<o:p></o:p></p>
<p class=3D"MsoNormal">packets.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The primary rationale for this new protocol is to en=
able deployment of<o:p></o:p></p>
<p class=3D"MsoNormal">real-time communications services on modern &quot;cl=
oud&quot; systems. These<o:p></o:p></p>
<p class=3D"MsoNormal">systems are built around HTTP, and for HTTP-based ap=
plications, they<o:p></o:p></p>
<p class=3D"MsoNormal">enable load balancing, HA, auto-scaling, and so on. =
However, real-time<o:p></o:p></p>
<p class=3D"MsoNormal">communications protocols are based on SIP and RTP, a=
nd they cannot take<o:p></o:p></p>
<p class=3D"MsoNormal">advantage of these HTTP-based capabilities.<o:p></o:=
p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">It is a non-goal of this working group to replicate =
all of SIP and its<o:p></o:p></p>
<p class=3D"MsoNormal">many extensions into HTTP. The group will limit itse=
lf to supporting the<o:p></o:p></p>
<p class=3D"MsoNormal">functionality in widespread actual usage today.<o:p>=
</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The protocol uses HTTP techniques for authentication=
 and authorization<o:p></o:p></p>
<p class=3D"MsoNormal">(notably OAuth), and requires hop by hop encryption =
(i.e., https). The<o:p></o:p></p>
<p class=3D"MsoNormal">protocol will also allow for e2e media encryption, a=
lthough keying is<o:p></o:p></p>
<p class=3D"MsoNormal">out of scope, and is expected to be handled by other=
 protocols such as<o:p></o:p></p>
<p class=3D"MsoNormal">MLS. This extension will also utilize STIR for calle=
rID.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This protocol should be implementable in browsers, t=
hick desktop<o:p></o:p></p>
<p class=3D"MsoNormal">clients, mobile apps and servers.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The group will standardize an extension to the basel=
ine protocol that<o:p></o:p></p>
<p class=3D"MsoNormal">automates the configuration necessary to achieve cal=
ling between on<o:p></o:p></p>
<p class=3D"MsoNormal">organization which is a customer of the other (for e=
xample, enterprise<o:p></o:p></p>
<p class=3D"MsoNormal">to service provider).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">[OPEN ISSUE: Is P2P media in or out? If in, ICE or s=
omething else ?]<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The group will do its work in conjunction with activ=
e software<o:p></o:p></p>
<p class=3D"MsoNormal">development efforts, so that implementation experien=
ce feeds directly<o:p></o:p></p>
<p class=3D"MsoNormal">into protocol development.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The group will coordinate with the Webtranport, QUIC=
, HTTPbis and STIR<o:p></o:p></p>
<p class=3D"MsoNormal">working groups.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Milestones:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Dec 2020: Submit baseline protocol to IESG<o:p></o:p=
></p>
<p class=3D"MsoNormal">Mar 2021: Submit autoconfiguration protocol to IESG<=
o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thx,<o:p></o:p></p>
<p class=3D"MsoNormal">Jonathan R.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">--<o:p></o:p></p>
<p class=3D"MsoNormal">Jonathan Rosenberg<o:p></o:p></p>
<p class=3D"MsoNormal">CTO and Head of AI, Five9<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></p>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"font-size:12.0pt;font-family:&quot;Times New Roman&quot;,&quot;se=
rif&quot;">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Ari=
al&quot;,&quot;sans-serif&quot;;color:gray"><br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
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</div>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></p>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
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--_000_6E58094ECC8D8344914996DAD28F1CCD27DAE805dggemm526mbxchi_--


From nobody Fri Mar 13 13:59:55 2020
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From: "\"IETF Secretariat\"" <agenda@ietf.org>
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Subject: [V3] ript - Requested session has been scheduled for IETF 107
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Dear Amy Vezza,

The session(s) that you have requested have been scheduled.
Below is the scheduled session information followed by
the original request. 


    ript Session 1 (2:00 requested)
    Thursday, 26 March 2020, Session II 2210-0010
    Room Name: Virtual Track 1 size: None
    ---------------------------------------------


iCalendar: https://datatracker.ietf.org/meeting/107/sessions/ript.ics

Request Information:


---------------------------------------------------------
Working Group Name: Realtime Internet Peering for Telephony
Area Name: Applications and Real-Time Area
Session Requester: Amy Vezza

Number of Sessions: 1
Length of Session(s):  2 Hours
Number of Attendees: 100
Conflicts to Avoid: 

 Technology Overlap: httpbis quic sipcore tls wpack webtrans



People who must be present:
  Adam Roach

Resources Requested:

Special Requests:
  
---------------------------------------------------------



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Hey everyone,

I've added an agenda to the datatracker.

https://datatracker.ietf.org/meeting/107/materials/agenda-107-ript-00

I couldn't find webex details, but please prepare for the meeting in advance, using the linked guide.

If you are listed as presenting, please provide slides as soon as possible.

--

Realtime Internet Peering for Telephony (RIPT) BoF

IETF 107
2020-03-26 22:10-00:10 UTC (ends 2020-03-27)
Virtual Track 1: (link TBA)
Guide to participation: https://www.ietf.org/how/meetings/107/session-participant-guide/

Agenda

Administrivia                         [Chairs]       10m
Problem Statement                     [Jonathan R.]  15m
High Level Protocol summary           [Jonathan R.]  15m
Implementation Experiences at Google  [Justin U.]    15m
Freeswitch use cases and needs        [Anthony M.]   10m
Charter review and questions          [Cullen J.]    30m

Drafts

https://tools.ietf.org/html/draft-rosenbergjennings-dispatch-ript-00
https://tools.ietf.org/html/draft-rosenbergjennings-dispatch-ript-00
https://tools.ietf.org/html/draft-rosenberg-dispatch-ript-sipdiffs-00
https://tools.ietf.org/html/draft-rosenberg-dispatch-ript-webrtc-00


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Subject: [V3] Dispatch-WG Draft On SRT (Secure Reliable Transport) FYI
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--0000000000003a405805a12822b8
Content-Type: text/plain; charset="UTF-8"

Hello my ript friends. I'm looking forward to the BoF!

But today, as co-chair of the ART Dispatch WG I am writing to let you know
about a draft that will be presented in that group on a 'as time allowed
basis' during the virtual meeting next week. We believe you may have some
interest and expertise in its content.

The draft is https://datatracker.ietf.org/doc/draft-sharabayko-mops-srt/

Abstract

   This document specifies Secure Reliable Transport (SRT) protocol.
   SRT is a user-level protocol over User Datagram Protocol and provides
   reliability and security optimized for low latency live video
   streaming, as well as generic bulk data transfer.  For this, SRT
   introduces control packet extension, improved flow control, enhanced
   congestion control and a mechanism for data encryption.

After consultation with the transport and art AD's we believe this may be
of interest to both communities. The goal of the dispatch WG is to help
identify the next step of a draft that is not currently adopted by a WG
(possibilities include but are not limited to a BoF, an existing WG, a new
WG, or more refinement on mailing lists). The group makes recommendations
but cannot actually require any of those actions. Come be part of our group!

Information on the Dispatch WG mailing list is here:
https://www.ietf.org/mailman/listinfo/dispatch

The virtual  meeting is Monday at 22:10 UTC aka
https://www.timeanddate.com/worldclock/fixedtime.html?msg=dispatch+meeting&iso=20200323T2210&p1=1440&ah=2

This work at least intersects with the interests of several areas and
groups, so please tolerate multiple copies of this invitation.

-Patrick

--0000000000003a405805a12822b8
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hello my ript friends. I&#39;m looking forward to the=
 BoF!</div><div><br></div><div>But today, as co-chair of the ART Dispatch W=
G I am writing to let you know about a draft that will be presented in that=
 group on a &#39;as time allowed basis&#39; during the virtual meeting next=
 week. We believe you may have some interest and expertise in its content.<=
br><br>The draft is=C2=A0<a href=3D"https://datatracker.ietf.org/doc/draft-=
sharabayko-mops-srt/">https://datatracker.ietf.org/doc/draft-sharabayko-mop=
s-srt/</a><br><br><pre style=3D"box-sizing:border-box;overflow:auto;font-fa=
mily:&quot;PT Mono&quot;,Monaco,monospace;font-size:14px;padding:10px;margi=
n-top:0px;margin-bottom:10.5px;line-height:1.214;color:rgb(0,0,0);word-brea=
k:break-all;background-color:rgb(255,253,245);border:1px solid rgb(204,204,=
204);border-radius:4px"><span class=3D"gmail-m_h" style=3D"box-sizing:borde=
r-box">Abstract</span>

   This document specifies Secure Reliable Transport (SRT) protocol.
   SRT is a user-level protocol over User Datagram Protocol and provides
   reliability and security optimized for low latency live video
   streaming, as well as generic bulk data transfer.  For this, SRT
   introduces control packet extension, improved flow control, enhanced
   congestion control and a mechanism for data encryption.
</pre>After consultation with the transport and art AD&#39;s we believe thi=
s may be of interest to both communities. The goal of the dispatch WG is to=
 help identify the next step of a draft that is not currently adopted by a =
WG (possibilities include but are not limited to a BoF, an existing WG, a n=
ew WG, or more refinement on mailing lists). The group makes recommendation=
s but cannot actually require any of those actions. Come be part of our gro=
up!</div><div><br></div><div>Information on the Dispatch WG mailing list is=
 here:=C2=A0<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">http=
s://www.ietf.org/mailman/listinfo/dispatch</a></div><div><br></div><div>The=
 virtual=C2=A0 meeting is Monday at 22:10 UTC aka=C2=A0<a href=3D"https://w=
ww.timeanddate.com/worldclock/fixedtime.html?msg=3Ddispatch+meeting&amp;iso=
=3D20200323T2210&amp;p1=3D1440&amp;ah=3D2">https://www.timeanddate.com/worl=
dclock/fixedtime.html?msg=3Ddispatch+meeting&amp;iso=3D20200323T2210&amp;p1=
=3D1440&amp;ah=3D2</a></div><div><br></div><div>This work at least intersec=
ts with the interests of several areas and groups, so please tolerate multi=
ple copies of this invitation.</div><div><br></div><div>-Patrick</div></div=
>

--0000000000003a405805a12822b8--


From nobody Fri Mar 20 10:30:10 2020
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Subject: [V3] Virtual IETF 107 Meeting WebEx Invitation: RIPT
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RIPT Session at IETF 107

Thursday, March 26, 2020
22:10-00:10 UTC

Join meeting: <https://ietf.webex.com/ietf/j.php?MTID=m77168c5c7592f30d7f369b49b47cfcf3>

Meeting number (access code): 646 916 587
Meeting password: avEzuWXp427

IETF 107 Agenda: <https://datatracker.ietf.org/meeting/107/agenda>


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With our BoF coming up this Thursday, I thought the time is right to restar=
t discussions with a focus in particular on SCOPE of the effort.

For me its defined entirely by the GOAL. The goal is:

It is possible to implement a ript server in public cloud, such that it is =
nothing more than a web application as far as public cloud is concerned, an=
d therefore works with web-centric features such as Kubernetes, global http=
 load balancers, and service meshes - to name a few. The server implements =
everything needed to make and receive voice and video calls.

I do think the scope needs to be inclusive of server to server peering, not=
 just client to server. The client-to-server stuff defines the core, but fo=
r me I'm really interested in building an alternative to sip trunking and t=
hat requires solving for additional things (inbound load balancing, inbound=
 authentication, number routing). I think those can easily be in a separate=
 draft but I think they should be in scope.

Thx,
Jonathan R.

--
Jonathan Rosenberg
CTO and Head of AI, Five9


________________________________

CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
idential information of Five9 and/or its affiliated entities. Access by the=
 intended recipient only is authorized. Any liability arising from any part=
y acting, or refraining from acting, on any information contained in this e=
-mail is hereby excluded. If you are not the intended recipient, please not=
ify the sender immediately, destroy the original transmission and its attac=
hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 23 Mar 2020 07:49:19 -0700
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To: Jonathan Rosenberg <jdrosen@five9.com>
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Subject: Re: [V3] Thoughts on scope for ript
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Hi Jonathan,

As I said in my review [0] I don't think that the current draft achieves
this, and I think it's lead to a number of fairly misguided design
decisions (with byways being perhaps the most obvious).

More to the point, the future is bigger than the past. H3 is not widely
deployed yet, and so we should think about how we want things to look, not
tie ourselves in knots trying to fit the precise current environment copied
onto H3.

-Ekr



[0] https://mailarchive.ietf.org/arch/msg/v3/V2BuGEZJpEMcAKcYLNHDMtV3i70/

On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg <jdrosen@five9.com>
wrote:

> With our BoF coming up this Thursday, I thought the time is right to
> restart discussions with a focus in particular on SCOPE of the effort.
>
> For me its defined entirely by the GOAL. The goal is:
>
> It is possible to implement a ript server in public cloud, such that it is
> nothing more than a web application as far as public cloud is concerned,
> and therefore works with web-centric features such as Kubernetes, global
> http load balancers, and service meshes - to name a few. The server
> implements everything needed to make and receive voice and video calls.
>
> I do think the scope needs to be inclusive of server to server peering,
> not just client to server. The client-to-server stuff defines the core, but
> for me I'm really interested in building an alternative to sip trunking and
> that requires solving for additional things (inbound load balancing,
> inbound authentication, number routing). I think those can easily be in a
> separate draft but I think they should be in scope.
>
> Thx,
> Jonathan R.
>
> --
> Jonathan Rosenberg
> CTO and Head of AI, Five9
>
>
> ________________________________
>
> CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain
> confidential information of Five9 and/or its affiliated entities. Access by
> the intended recipient only is authorized. Any liability arising from any
> party acting, or refraining from acting, on any information contained in
> this e-mail is hereby excluded. If you are not the intended recipient,
> please notify the sender immediately, destroy the original transmission and
> its attachments and do not disclose the contents to any other person, use
> it for any purpose, or store or copy the information in any medium.
> Copyright in this e-mail and any attachments belongs to Five9 and/or its
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<div dir=3D"ltr"><div>Hi Jonathan,</div><div><br></div><div>As I said in my=
 review [0] I don&#39;t think that the current draft achieves this, and I t=
hink it&#39;s lead to a number of fairly misguided design decisions (with b=
yways being perhaps the most obvious).</div><div><br></div><div>More to the=
 point, the future is bigger than the past. H3 is not widely deployed yet, =
and so we should think about how we want things to look, not tie ourselves =
in knots trying to fit the precise current environment copied onto H3.<br><=
/div><div><br></div><div>-Ekr</div><div><br></div><div><br></div><div><br><=
/div><div>[0] <a href=3D"https://mailarchive.ietf.org/arch/msg/v3/V2BuGEZJp=
EMcAKcYLNHDMtV3i70/" target=3D"_blank">https://mailarchive.ietf.org/arch/ms=
g/v3/V2BuGEZJpEMcAKcYLNHDMtV3i70/</a></div></div><br><div class=3D"gmail_qu=
ote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Mar 23, 2020 at 7:21 AM =
Jonathan Rosenberg &lt;<a href=3D"mailto:jdrosen@five9.com" target=3D"_blan=
k">jdrosen@five9.com</a>&gt; wrote:<br></div><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">With our BoF coming up this Thursday, I thought the tim=
e is right to restart discussions with a focus in particular on SCOPE of th=
e effort.<br>
<br>
For me its defined entirely by the GOAL. The goal is:<br>
<br>
It is possible to implement a ript server in public cloud, such that it is =
nothing more than a web application as far as public cloud is concerned, an=
d therefore works with web-centric features such as Kubernetes, global http=
 load balancers, and service meshes - to name a few. The server implements =
everything needed to make and receive voice and video calls.<br>
<br>
I do think the scope needs to be inclusive of server to server peering, not=
 just client to server. The client-to-server stuff defines the core, but fo=
r me I&#39;m really interested in building an alternative to sip trunking a=
nd that requires solving for additional things (inbound load balancing, inb=
ound authentication, number routing). I think those can easily be in a sepa=
rate draft but I think they should be in scope.<br>
<br>
Thx,<br>
Jonathan R.<br>
<br>
--<br>
Jonathan Rosenberg<br>
CTO and Head of AI, Five9<br>
<br>
<br>
________________________________<br>
<br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
idential information of Five9 and/or its affiliated entities. Access by the=
 intended recipient only is authorized. Any liability arising from any part=
y acting, or refraining from acting, on any information contained in this e=
-mail is hereby excluded. If you are not the intended recipient, please not=
ify the sender immediately, destroy the original transmission and its attac=
hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
.<br>
<br>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 23 Mar 2020 10:09:22 -0700
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Subject: Re: [V3] Thoughts on scope for ript
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In addition to Eric's comments, I think the biggest question regarding
goals and scope is whether we focus on both client-to-server and
server-to-server.

If we are taking a fresh look at how all of IP calling works, we could find
ourselves looking at:
a) a replacement for SIP "RPCs" (i.e, how are commands reliably exchanged)
b) a replacement for SIP operations and routing (i.e., the specific
commands and routing mechanisms)
c) a replacement for offer-answer (i.e., how do both sides agree on media
formats)
d) a replacement for SDP (i.e., the details of how media formats are
described)
e) a replacement for SRTP (i.e., how media is transmitted from point A to B)
f) a replacement for RTP packetization (i.e., how media is encapsulated
when transmitted)

I think that server-to-server, essentially a NNI, requires us to consider
all of these items, and that may ultimately make sense. However, if we
focus initially on client-to-server, we can get by by only focusing on e)
and f), as a-d) are under application control

This could still provide a lot of value, since moving to a world where even
private RTC services can be built atop HTTP would still be quite an
accomplishment. (Compared to the current state of the world, where people
deploying cloud RTC services need to source and deploy their own
ICE/DTLS/SCTP/SRTP stacks and all the load
balancing/authentication/failover tech that Jonathan mentions). It would
also be complementary to WebRTC, which works pretty well for p2p
applications, but struggles in c2s environments for the aforementioned
reasons.


On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg <jdrosen@five9.com>
wrote:

> With our BoF coming up this Thursday, I thought the time is right to
> restart discussions with a focus in particular on SCOPE of the effort.
>
> For me its defined entirely by the GOAL. The goal is:
>
> It is possible to implement a ript server in public cloud, such that it is
> nothing more than a web application as far as public cloud is concerned,
> and therefore works with web-centric features such as Kubernetes, global
> http load balancers, and service meshes - to name a few. The server
> implements everything needed to make and receive voice and video calls.
>
> I do think the scope needs to be inclusive of server to server peering,
> not just client to server. The client-to-server stuff defines the core, but
> for me I'm really interested in building an alternative to sip trunking and
> that requires solving for additional things (inbound load balancing,
> inbound authentication, number routing). I think those can easily be in a
> separate draft but I think they should be in scope.
>
> Thx,
> Jonathan R.
>
> --
> Jonathan Rosenberg
> CTO and Head of AI, Five9
>
>
> ________________________________
>
> CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain
> confidential information of Five9 and/or its affiliated entities. Access by
> the intended recipient only is authorized. Any liability arising from any
> party acting, or refraining from acting, on any information contained in
> this e-mail is hereby excluded. If you are not the intended recipient,
> please notify the sender immediately, destroy the original transmission and
> its attachments and do not disclose the contents to any other person, use
> it for any purpose, or store or copy the information in any medium.
> Copyright in this e-mail and any attachments belongs to Five9 and/or its
> affiliated entities.
>
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>

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<div dir=3D"ltr">In addition to Eric&#39;s comments, I think the biggest qu=
estion regarding goals and scope is whether we focus on both client-to-serv=
er and server-to-server.=C2=A0<div><br></div><div>If we are taking a fresh =
look at how all of IP calling works, we could find ourselves looking at:</d=
iv><div><div></div><div></div><div>a) a replacement for=C2=A0<span class=3D=
"gmail-il">SIP</span>=C2=A0&quot;RPCs&quot; (i.e, how are commands reliably=
 exchanged)</div><div>b) a replacement for SIP operations and routing (i.e.=
, the specific commands and routing mechanisms)</div><div>c) a replacement =
for offer-answer (i.e., how do both sides agree on media formats)<br></div>=
<div>d) a replacement for=C2=A0<span class=3D"gmail-il">SDP (i.e., the deta=
ils of how media formats are described)</span><br></div><div>e) a replaceme=
nt for SRTP (i.e., how media is transmitted from point A to B)</div><div>f)=
 a replacement for RTP packetization (i.e., how media is encapsulated when =
transmitted)</div><div><br></div><div>I think that server-to-server, essent=
ially a NNI, requires us to consider all of these items, and that may ultim=
ately make sense. However, if we focus initially on client-to-server, we ca=
n get by by only focusing on e) and f), as a-d) are under application contr=
ol</div><div><br></div><div>This could still provide a lot of value, since =
moving to a world where even private RTC services can be built atop HTTP wo=
uld still be quite an accomplishment. (Compared to the current state of the=
 world, where people deploying cloud RTC services need to source and deploy=
 their own ICE/DTLS/SCTP/SRTP stacks and all the load balancing/authenticat=
ion/failover tech that Jonathan mentions). It would also be complementary t=
o WebRTC, which works pretty well for p2p applications, but struggles in c2=
s environments for the aforementioned reasons.</div></div><div><br></div></=
div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On=
 Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg &lt;<a href=3D"mailto:jdro=
sen@five9.com">jdrosen@five9.com</a>&gt; wrote:<br></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">With our BoF coming up this Thursday, I th=
ought the time is right to restart discussions with a focus in particular o=
n SCOPE of the effort.<br>
<br>
For me its defined entirely by the GOAL. The goal is:<br>
<br>
It is possible to implement a ript server in public cloud, such that it is =
nothing more than a web application as far as public cloud is concerned, an=
d therefore works with web-centric features such as Kubernetes, global http=
 load balancers, and service meshes - to name a few. The server implements =
everything needed to make and receive voice and video calls.<br>
<br>
I do think the scope needs to be inclusive of server to server peering, not=
 just client to server. The client-to-server stuff defines the core, but fo=
r me I&#39;m really interested in building an alternative to sip trunking a=
nd that requires solving for additional things (inbound load balancing, inb=
ound authentication, number routing). I think those can easily be in a sepa=
rate draft but I think they should be in scope.<br>
<br>
Thx,<br>
Jonathan R.<br>
<br>
--<br>
Jonathan Rosenberg<br>
CTO and Head of AI, Five9<br>
<br>
<br>
________________________________<br>
<br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
idential information of Five9 and/or its affiliated entities. Access by the=
 intended recipient only is authorized. Any liability arising from any part=
y acting, or refraining from acting, on any information contained in this e=
-mail is hereby excluded. If you are not the intended recipient, please not=
ify the sender immediately, destroy the original transmission and its attac=
hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
.<br>
<br>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>

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From: Anthony Minessale <anthm@signalwire.com>
Date: Mon, 23 Mar 2020 17:22:31 -0500
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Cc: Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] Thoughts on scope for ript
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Also more consideration for live translation to sip/rtp/webrtc and
back-to-back scenarios.

If calls are being loaded balanced to many servers you still probably will
want a way to keep track of what calls are on what server in a way that
goes out of scope of HTTP so the application doing the clustering will bear
a pretty big burden of solving that problem from behind the cluster.

I read the memo about replacing SIP and was slightly disturbed that RFC2833
(4733/4734) is the one thing salvaged ;)
That would be the first to go IMHO, its the leading cause of interop issues
in all of telephony and very rarely implemented to its full potential not
to mention there are some timestamp translation issues I pointed out on the
github discussion so that is it's own can of worms.

Also, the description seems to be evolving more towards considerations
about making calls behave like a web app which is fine because that's the
GOAL but there is little consideration of actual trunking so far.  SIP
trunking is just a made-up marketing term to mirror original TDM trunking
which was actual trunking (bundling many calls into one stream to reduce
overhead) In reality SIP trunking is often just (a lot of SIP calls) We
have our SIP setup so you can send many calls to many different phone
numbers over a single registration and that causes introp issues because
most people don't comprehend this lol.

So it seems like there could be some improvements that could clean up
issues with WebRTC and how web-app type environments can bust through nat
etc.
WebRTC is still compatible with SIP in the sense that you can carry a
WebRTC SDP over sip and set up calls but most of the time its SIP over
WebSockets (though FS and Kamailio can translate and negotiate WebRTC
sessions to other SIP transports)

>From a server perspective with density being a major concern, removing all
the need for dialogs and route headers is a good thing. What I also think
is critical to preserve is the server's ability to limit the media formats
etc sent by the sender.  Killing Offer/Answer is ok but the replacement
should at least preserve the notions that a client can offer what it can
do, the server can narrow that scope with its own requirements and the
client must work within those shared formats.  The server is the one who
has to create thousands of connections etc and must not be treated as a
simple peer-to-peer agent.

Removing SRTP is another concern because it's pretty widely used now, it's
required in WebRTC and decryption in the middle was something I was hoping
we would go the other direction on and find ways to route encrypted packets
without needing to decode them.

I think if you want to replace SIP trunking more considerations have to be
made for the mass translation of TDM/SIP/SS7 and how that works with
routing and clustering.  I think some of that is how we got into that boat
with SIP because it was only later that this was tried and then workarounds
abounded for years.

On Mon, Mar 23, 2020 at 12:09 PM Justin Uberti <juberti=
40google.com@dmarc.ietf.org> wrote:

> In addition to Eric's comments, I think the biggest question regarding
> goals and scope is whether we focus on both client-to-server and
> server-to-server.
>
> If we are taking a fresh look at how all of IP calling works, we could
> find ourselves looking at:
> a) a replacement for SIP "RPCs" (i.e, how are commands reliably exchanged)
> b) a replacement for SIP operations and routing (i.e., the specific
> commands and routing mechanisms)
> c) a replacement for offer-answer (i.e., how do both sides agree on media
> formats)
> d) a replacement for SDP (i.e., the details of how media formats are
> described)
> e) a replacement for SRTP (i.e., how media is transmitted from point A to
> B)
> f) a replacement for RTP packetization (i.e., how media is encapsulated
> when transmitted)
>
> I think that server-to-server, essentially a NNI, requires us to consider
> all of these items, and that may ultimately make sense. However, if we
> focus initially on client-to-server, we can get by by only focusing on e)
> and f), as a-d) are under application control
>
> This could still provide a lot of value, since moving to a world where
> even private RTC services can be built atop HTTP would still be quite an
> accomplishment. (Compared to the current state of the world, where people
> deploying cloud RTC services need to source and deploy their own
> ICE/DTLS/SCTP/SRTP stacks and all the load
> balancing/authentication/failover tech that Jonathan mentions). It would
> also be complementary to WebRTC, which works pretty well for p2p
> applications, but struggles in c2s environments for the aforementioned
> reasons.
>
>
> On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg <jdrosen@five9.com>
> wrote:
>
>> With our BoF coming up this Thursday, I thought the time is right to
>> restart discussions with a focus in particular on SCOPE of the effort.
>>
>> For me its defined entirely by the GOAL. The goal is:
>>
>> It is possible to implement a ript server in public cloud, such that it
>> is nothing more than a web application as far as public cloud is concerned,
>> and therefore works with web-centric features such as Kubernetes, global
>> http load balancers, and service meshes - to name a few. The server
>> implements everything needed to make and receive voice and video calls.
>>
>> I do think the scope needs to be inclusive of server to server peering,
>> not just client to server. The client-to-server stuff defines the core, but
>> for me I'm really interested in building an alternative to sip trunking and
>> that requires solving for additional things (inbound load balancing,
>> inbound authentication, number routing). I think those can easily be in a
>> separate draft but I think they should be in scope.
>>
>> Thx,
>> Jonathan R.
>>
>> --
>> Jonathan Rosenberg
>> CTO and Head of AI, Five9
>>
>>
>> ________________________________
>>
>> CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain
>> confidential information of Five9 and/or its affiliated entities. Access by
>> the intended recipient only is authorized. Any liability arising from any
>> party acting, or refraining from acting, on any information contained in
>> this e-mail is hereby excluded. If you are not the intended recipient,
>> please notify the sender immediately, destroy the original transmission and
>> its attachments and do not disclose the contents to any other person, use
>> it for any purpose, or store or copy the information in any medium.
>> Copyright in this e-mail and any attachments belongs to Five9 and/or its
>> affiliated entities..
>>
>> --
>> V3 mailing list
>> V3@ietf.org
>> https://www.ietf.org/mailman/listinfo/v3
>>
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>


-- 


Anthony Minessale  | Founder and CEO

SignalWire: *228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303
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--000000000000ed4f8605a18d1216
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div dir=3D"auto">Also more consideration for live transla=
tion to sip/rtp/webrtc and back-to-back scenarios.</div><div dir=3D"auto"><=
br></div><div>If calls are being loaded balanced to many servers you still =
probably will want a way to keep track of what calls are on what server in =
a way that goes out of scope of HTTP so the application doing the clusterin=
g will bear a pretty big burden of solving that problem from behind the clu=
ster.</div><div dir=3D"auto"><br></div><div>I read the memo about replacing=
 SIP and was slightly disturbed that RFC2833 (4733/4734) is the one thing s=
alvaged ;)</div><div>That would be the first to go IMHO, its the leading ca=
use of interop issues in all of telephony and very rarely implemented to it=
s full potential not to mention there are some timestamp translation issues=
 I pointed out on the github discussion so that is it&#39;s own can of worm=
s.</div><div><br></div><div>Also, the description seems to be evolving more=
 towards considerations about making calls behave like a web app which is f=
ine because that&#39;s the GOAL but there is little consideration of actual=
 trunking so far.=C2=A0 SIP trunking is just a made-up marketing term to mi=
rror original TDM trunking which was actual trunking (bundling many calls i=
nto one stream to reduce overhead) In reality SIP trunking is often just (a=
 lot of SIP calls) We have our SIP setup so you can send many calls to many=
 different phone numbers over a single registration and that causes introp =
issues because most people don&#39;t comprehend this lol.</div><div><br></d=
iv><div>So it seems like there could be some improvements that could clean =
up issues with WebRTC and how web-app type environments can bust through na=
t etc.</div><div>WebRTC is still compatible with SIP in the sense that you =
can carry a WebRTC SDP over sip and set up calls but most of the time its S=
IP over WebSockets (though FS and Kamailio can translate and negotiate WebR=
TC sessions to other SIP transports)<br></div><div><br></div><div>From a se=
rver perspective with density being a major concern, removing all the need =
for dialogs and route headers is a good thing. What I also think is critica=
l to preserve is the server&#39;s ability to limit the media formats etc se=
nt by the sender.=C2=A0 Killing Offer/Answer is ok but the replacement shou=
ld at least preserve the notions that a client can offer what it can do, th=
e server can narrow that scope with its own requirements and the client mus=
t work within those shared formats.=C2=A0 The server is the one who has to =
create thousands of connections etc and must not be treated as a simple pee=
r-to-peer agent.</div><div><br></div><div>Removing SRTP is another concern =
because it&#39;s pretty widely used now, it&#39;s required in WebRTC and de=
cryption in the middle was something I was hoping we would go the other dir=
ection on and find ways to route encrypted packets without needing to decod=
e them.</div><div><br></div><div>I think if you want to replace SIP trunkin=
g more considerations have to be made for the mass translation of TDM/SIP/S=
S7 and how that works with routing and clustering.=C2=A0 I think some of th=
at is how we got into that boat with SIP because it was only later that thi=
s was tried and then workarounds abounded for years.</div></div><br><div cl=
ass=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Mar 23, 2=
020 at 12:09 PM Justin Uberti &lt;juberti=3D<a href=3D"mailto:40google.com@=
dmarc.ietf.org">40google.com@dmarc.ietf.org</a>&gt; wrote:<br></div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1p=
x solid rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr">In addition to =
Eric&#39;s comments, I think the biggest question regarding goals and scope=
 is whether we focus on both client-to-server and server-to-server.=C2=A0<d=
iv><br></div><div>If we are taking a fresh look at how all of IP calling wo=
rks, we could find ourselves looking at:</div><div><div></div><div></div><d=
iv>a) a replacement for=C2=A0<span>SIP</span>=C2=A0&quot;RPCs&quot; (i.e, h=
ow are commands reliably exchanged)</div><div>b) a replacement for SIP oper=
ations and routing (i.e., the specific commands and routing mechanisms)</di=
v><div>c) a replacement for offer-answer (i.e., how do both sides agree on =
media formats)<br></div><div>d) a replacement for=C2=A0<span>SDP (i.e., the=
 details of how media formats are described)</span><br></div><div>e) a repl=
acement for SRTP (i.e., how media is transmitted from point A to B)</div><d=
iv>f) a replacement for RTP packetization (i.e., how media is encapsulated =
when transmitted)</div><div><br></div><div>I think that server-to-server, e=
ssentially a NNI, requires us to consider all of these items, and that may =
ultimately make sense. However, if we focus initially on client-to-server, =
we can get by by only focusing on e) and f), as a-d) are under application =
control</div><div><br></div><div>This could still provide a lot of value, s=
ince moving to a world where even private RTC services can be built atop HT=
TP would still be quite an accomplishment. (Compared to the current state o=
f the world, where people deploying cloud RTC services need to source and d=
eploy their own ICE/DTLS/SCTP/SRTP stacks and all the load balancing/authen=
tication/failover tech that Jonathan mentions). It would also be complement=
ary to WebRTC, which works pretty well for p2p applications, but struggles =
in c2s environments for the aforementioned reasons.</div></div><div><br></d=
iv></div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_att=
r">On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg &lt;<a href=3D"mailto=
:jdrosen@five9.com" target=3D"_blank">jdrosen@five9.com</a>&gt; wrote:<br><=
/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">With our BoF coming =
up this Thursday, I thought the time is right to restart discussions with a=
 focus in particular on SCOPE of the effort.<br>
<br>
For me its defined entirely by the GOAL. The goal is:<br>
<br>
It is possible to implement a ript server in public cloud, such that it is =
nothing more than a web application as far as public cloud is concerned, an=
d therefore works with web-centric features such as Kubernetes, global http=
 load balancers, and service meshes - to name a few. The server implements =
everything needed to make and receive voice and video calls.<br>
<br>
I do think the scope needs to be inclusive of server to server peering, not=
 just client to server. The client-to-server stuff defines the core, but fo=
r me I&#39;m really interested in building an alternative to sip trunking a=
nd that requires solving for additional things (inbound load balancing, inb=
ound authentication, number routing). I think those can easily be in a sepa=
rate draft but I think they should be in scope.<br>
<br>
Thx,<br>
Jonathan R.<br>
<br>
--<br>
Jonathan Rosenberg<br>
CTO and Head of AI, Five9<br>
<br>
<br>
________________________________<br>
<br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
idential information of Five9 and/or its affiliated entities. Access by the=
 intended recipient only is authorized. Any liability arising from any part=
y acting, or refraining from acting, on any information contained in this e=
-mail is hereby excluded. If you are not the intended recipient, please not=
ify the sender immediately, destroy the original transmission and its attac=
hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
..<br>
<br>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>
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From: Justin Uberti <juberti@google.com>
Date: Mon, 23 Mar 2020 15:43:00 -0700
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Subject: Re: [V3] Thoughts on scope for ript
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On Mon, Mar 23, 2020 at 3:24 PM Anthony Minessale <anthm@signalwire.com>
wrote:

> Also more consideration for live translation to sip/rtp/webrtc and
> back-to-back scenarios.
>
> If calls are being loaded balanced to many servers you still probably will
> want a way to keep track of what calls are on what server in a way that
> goes out of scope of HTTP so the application doing the clustering will bear
> a pretty big burden of solving that problem from behind the cluster.
>
> I read the memo about replacing SIP and was slightly disturbed that
> RFC2833 (4733/4734) is the one thing salvaged ;)
> That would be the first to go IMHO, its the leading cause of interop
> issues in all of telephony and very rarely implemented to its full
> potential not to mention there are some timestamp translation issues I
> pointed out on the github discussion so that is it's own can of worms.
>
> Also, the description seems to be evolving more towards considerations
> about making calls behave like a web app which is fine because that's the
> GOAL but there is little consideration of actual trunking so far.  SIP
> trunking is just a made-up marketing term to mirror original TDM trunking
> which was actual trunking (bundling many calls into one stream to reduce
> overhead) In reality SIP trunking is often just (a lot of SIP calls) We
> have our SIP setup so you can send many calls to many different phone
> numbers over a single registration and that causes introp issues because
> most people don't comprehend this lol.
>


> So it seems like there could be some improvements that could clean up
> issues with WebRTC and how web-app type environments can bust through nat
> etc.
> WebRTC is still compatible with SIP in the sense that you can carry a
> WebRTC SDP over sip and set up calls but most of the time its SIP over
> WebSockets (though FS and Kamailio can translate and negotiate WebRTC
> sessions to other SIP transports)
>
> From a server perspective with density being a major concern, removing all
> the need for dialogs and route headers is a good thing. What I also think
> is critical to preserve is the server's ability to limit the media formats
> etc sent by the sender.  Killing Offer/Answer is ok but the replacement
> should at least preserve the notions that a client can offer what it can
> do, the server can narrow that scope with its own requirements and the
> client must work within those shared formats.  The server is the one who
> has to create thousands of connections etc and must not be treated as a
> simple peer-to-peer agent.
>
> Removing SRTP is another concern because it's pretty widely used now, it's
> required in WebRTC and decryption in the middle was something I was hoping
> we would go the other direction on and find ways to route encrypted packets
> without needing to decode them.
>

One reason we are focusing on client-to-server vs peer-to-peer is that it
avoids the interop issues associated with p2p (i.e., the difference between
connecting to the server that you control vs various peers that you don't
control). Also, WebRTC already works well for the peer-to-peer case.

I also think that whatever we come up with to replace SRTP will probably be
able to gateway to SRTP easily for systems that expect SRTP.

>
> I think if you want to replace SIP trunking more considerations have to be
> made for the mass translation of TDM/SIP/SS7 and how that works with
> routing and clustering.  I think some of that is how we got into that boat
> with SIP because it was only later that this was tried and then workarounds
> abounded for years.
>
> On Mon, Mar 23, 2020 at 12:09 PM Justin Uberti <juberti=
> 40google.com@dmarc.ietf.org> wrote:
>
>> In addition to Eric's comments, I think the biggest question regarding
>> goals and scope is whether we focus on both client-to-server and
>> server-to-server.
>>
>> If we are taking a fresh look at how all of IP calling works, we could
>> find ourselves looking at:
>> a) a replacement for SIP "RPCs" (i.e, how are commands reliably
>> exchanged)
>> b) a replacement for SIP operations and routing (i.e., the specific
>> commands and routing mechanisms)
>> c) a replacement for offer-answer (i.e., how do both sides agree on media
>> formats)
>> d) a replacement for SDP (i.e., the details of how media formats are
>> described)
>> e) a replacement for SRTP (i.e., how media is transmitted from point A to
>> B)
>> f) a replacement for RTP packetization (i.e., how media is encapsulated
>> when transmitted)
>>
>> I think that server-to-server, essentially a NNI, requires us to consider
>> all of these items, and that may ultimately make sense. However, if we
>> focus initially on client-to-server, we can get by by only focusing on e)
>> and f), as a-d) are under application control
>>
>> This could still provide a lot of value, since moving to a world where
>> even private RTC services can be built atop HTTP would still be quite an
>> accomplishment. (Compared to the current state of the world, where people
>> deploying cloud RTC services need to source and deploy their own
>> ICE/DTLS/SCTP/SRTP stacks and all the load
>> balancing/authentication/failover tech that Jonathan mentions). It would
>> also be complementary to WebRTC, which works pretty well for p2p
>> applications, but struggles in c2s environments for the aforementioned
>> reasons.
>>
>>
>> On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg <jdrosen@five9.com>
>> wrote:
>>
>>> With our BoF coming up this Thursday, I thought the time is right to
>>> restart discussions with a focus in particular on SCOPE of the effort.
>>>
>>> For me its defined entirely by the GOAL. The goal is:
>>>
>>> It is possible to implement a ript server in public cloud, such that it
>>> is nothing more than a web application as far as public cloud is concerned,
>>> and therefore works with web-centric features such as Kubernetes, global
>>> http load balancers, and service meshes - to name a few. The server
>>> implements everything needed to make and receive voice and video calls.
>>>
>>> I do think the scope needs to be inclusive of server to server peering,
>>> not just client to server. The client-to-server stuff defines the core, but
>>> for me I'm really interested in building an alternative to sip trunking and
>>> that requires solving for additional things (inbound load balancing,
>>> inbound authentication, number routing). I think those can easily be in a
>>> separate draft but I think they should be in scope.
>>>
>>> Thx,
>>> Jonathan R.
>>>
>>> --
>>> Jonathan Rosenberg
>>> CTO and Head of AI, Five9
>>>
>>>
>>> ________________________________
>>>
>>> CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain
>>> confidential information of Five9 and/or its affiliated entities. Access by
>>> the intended recipient only is authorized. Any liability arising from any
>>> party acting, or refraining from acting, on any information contained in
>>> this e-mail is hereby excluded. If you are not the intended recipient,
>>> please notify the sender immediately, destroy the original transmission and
>>> its attachments and do not disclose the contents to any other person, use
>>> it for any purpose, or store or copy the information in any medium.
>>> Copyright in this e-mail and any attachments belongs to Five9 and/or its
>>> affiliated entities...
>>>
>>> --
>>> V3 mailing list
>>> V3@ietf.org
>>> https://www.ietf.org/mailman/listinfo/v3
>>>
>> --
>> V3 mailing list
>> V3@ietf.org
>> https://www.ietf.org/mailman/listinfo/v3
>>
>
>
> --
>
>
> Anthony Minessale  | Founder and CEO
>
> SignalWire: *228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303
> <https://maps.google.com/?q=228+Hamilton+Ave+3rd+Floor,+Palo+Alto,+CA+94303&entry=gmail&source=g>*
>
> *https://youtu.be/WKjZkzgK4pc <https://youtu.be/WKjZkzgK4pc>*
>
> https://youtu.be/MwqpSl7luSM
>
> https://youtu.be/l82JtCECkog <https://www.youtube.com/watch?v=l82JtCECkog>
>
> Email: anthm@signalwire.com
>
> Phone: 262-309-8501 <(262)%20309-8501>
>
> Website: <https://www.freeswitch.com/>https://www.signalwire.com
> <https://www.facebook.com/search/top/?q=signalwire>
> <https://twitter.com/SignalWire>
>
>
>

--000000000000246d5005a18d5c42
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div></div><br><div class=3D"gmail_quote"><div dir=3D"ltr"=
 class=3D"gmail_attr">On Mon, Mar 23, 2020 at 3:24 PM Anthony Minessale &lt=
;<a href=3D"mailto:anthm@signalwire.com">anthm@signalwire.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"><div dir=3D"lt=
r"><div dir=3D"auto">Also more consideration for live translation to sip/rt=
p/webrtc and back-to-back scenarios.</div><div dir=3D"auto"><br></div><div>=
If calls are being loaded balanced to many servers you still probably will =
want a way to keep track of what calls are on what server in a way that goe=
s out of scope of HTTP so the application doing the clustering will bear a =
pretty big burden of solving that problem from behind the cluster.</div><di=
v dir=3D"auto"><br></div><div>I read the memo about replacing SIP and was s=
lightly disturbed that RFC2833 (4733/4734) is the one thing salvaged ;)</di=
v><div>That would be the first to go IMHO, its the leading cause of interop=
 issues in all of telephony and very rarely implemented to its full potenti=
al not to mention there are some timestamp translation issues I pointed out=
 on the github discussion so that is it&#39;s own can of worms.</div><div><=
br></div><div>Also, the description seems to be evolving more towards consi=
derations about making calls behave like a web app which is fine because th=
at&#39;s the GOAL but there is little consideration of actual trunking so f=
ar.=C2=A0 SIP trunking is just a made-up marketing term to mirror original =
TDM trunking which was actual trunking (bundling many calls into one stream=
 to reduce overhead) In reality SIP trunking is often just (a lot of SIP ca=
lls) We have our SIP setup so you can send many calls to many different pho=
ne numbers over a single registration and that causes introp issues because=
 most people don&#39;t comprehend this lol.</div></div></blockquote><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 dir=3D"l=
tr"><div></div><div>So it seems like there could be some improvements that =
could clean up issues with WebRTC and how web-app type environments can bus=
t through nat etc.</div><div>WebRTC is still compatible with SIP in the sen=
se that you can carry a WebRTC SDP over sip and set up calls but most of th=
e time its SIP over WebSockets (though FS and Kamailio can translate and ne=
gotiate WebRTC sessions to other SIP transports)<br></div><div><br></div><d=
iv>From a server perspective with density being a major concern, removing a=
ll the need for dialogs and route headers is a good thing. What I also thin=
k is critical to preserve is the server&#39;s ability to limit the media fo=
rmats etc sent by the sender.=C2=A0 Killing Offer/Answer is ok but the repl=
acement should at least preserve the notions that a client can offer what i=
t can do, the server can narrow that scope with its own requirements and th=
e client must work within those shared formats.=C2=A0 The server is the one=
 who has to create thousands of connections etc and must not be treated as =
a simple peer-to-peer agent.</div><div><br></div><div>Removing SRTP is anot=
her concern because it&#39;s pretty widely used now, it&#39;s required in W=
ebRTC and decryption in the middle was something I was hoping we would go t=
he other direction on and find ways to route encrypted packets without need=
ing to decode them.</div></div></blockquote><div><br></div><div>One reason =
we are focusing on client-to-server vs peer-to-peer is that it avoids the i=
nterop issues associated with p2p (i.e., the difference between connecting =
to the server that you control vs various peers that you don&#39;t control)=
. Also, WebRTC already works well for the peer-to-peer case.=C2=A0</div><di=
v><br></div><div>I also think that whatever we come up with to replace SRTP=
 will probably be able to gateway to SRTP easily for systems that expect SR=
TP.</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr"=
><div><br></div><div>I think if you want to replace SIP trunking more consi=
derations have to be made for the mass translation of TDM/SIP/SS7 and how t=
hat works with routing and clustering.=C2=A0 I think some of that is how we=
 got into that boat with SIP because it was only later that this was tried =
and then workarounds abounded for years.</div></div><br><div class=3D"gmail=
_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Mar 23, 2020 at 12:09=
 PM Justin Uberti &lt;juberti=3D<a href=3D"mailto:40google.com@dmarc.ietf.o=
rg" target=3D"_blank">40google.com@dmarc.ietf.org</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 dir=3D"ltr">In additi=
on to Eric&#39;s comments, I think the biggest question regarding goals and=
 scope is whether we focus on both client-to-server and server-to-server.=
=C2=A0<div><br></div><div>If we are taking a fresh look at how all of IP ca=
lling works, we could find ourselves looking at:</div><div><div></div><div>=
</div><div>a) a replacement for=C2=A0<span>SIP</span>=C2=A0&quot;RPCs&quot;=
 (i.e, how are commands reliably exchanged)</div><div>b) a replacement for =
SIP operations and routing (i.e., the specific commands and routing mechani=
sms)</div><div>c) a replacement for offer-answer (i.e., how do both sides a=
gree on media formats)<br></div><div>d) a replacement for=C2=A0<span>SDP (i=
.e., the details of how media formats are described)</span><br></div><div>e=
) a replacement for SRTP (i.e., how media is transmitted from point A to B)=
</div><div>f) a replacement for RTP packetization (i.e., how media is encap=
sulated when transmitted)</div><div><br></div><div>I think that server-to-s=
erver, essentially a NNI, requires us to consider all of these items, and t=
hat may ultimately make sense. However, if we focus initially on client-to-=
server, we can get by by only focusing on e) and f), as a-d) are under appl=
ication control</div><div><br></div><div>This could still provide a lot of =
value, since moving to a world where even private RTC services can be built=
 atop HTTP would still be quite an accomplishment. (Compared to the current=
 state of the world, where people deploying cloud RTC services need to sour=
ce and deploy their own ICE/DTLS/SCTP/SRTP stacks and all the load balancin=
g/authentication/failover tech that Jonathan mentions). It would also be co=
mplementary to WebRTC, which works pretty well for p2p applications, but st=
ruggles in c2s environments for the aforementioned reasons.</div></div><div=
><br></div></div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"g=
mail_attr">On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg &lt;<a href=
=3D"mailto:jdrosen@five9.com" target=3D"_blank">jdrosen@five9.com</a>&gt; w=
rote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0p=
x 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">With our B=
oF coming up this Thursday, I thought the time is right to restart discussi=
ons with a focus in particular on SCOPE of the effort.<br>
<br>
For me its defined entirely by the GOAL. The goal is:<br>
<br>
It is possible to implement a ript server in public cloud, such that it is =
nothing more than a web application as far as public cloud is concerned, an=
d therefore works with web-centric features such as Kubernetes, global http=
 load balancers, and service meshes - to name a few. The server implements =
everything needed to make and receive voice and video calls.<br>
<br>
I do think the scope needs to be inclusive of server to server peering, not=
 just client to server. The client-to-server stuff defines the core, but fo=
r me I&#39;m really interested in building an alternative to sip trunking a=
nd that requires solving for additional things (inbound load balancing, inb=
ound authentication, number routing). I think those can easily be in a sepa=
rate draft but I think they should be in scope.<br>
<br>
Thx,<br>
Jonathan R.<br>
<br>
--<br>
Jonathan Rosenberg<br>
CTO and Head of AI, Five9<br>
<br>
<br>
________________________________<br>
<br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
idential information of Five9 and/or its affiliated entities. Access by the=
 intended recipient only is authorized. Any liability arising from any part=
y acting, or refraining from acting, on any information contained in this e=
-mail is hereby excluded. If you are not the intended recipient, please not=
ify the sender immediately, destroy the original transmission and its attac=
hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
...<br>
<br>
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<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>
-- <br>
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<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div><br clear=3D"all"><div><br></div>-- <br><div dir=3D"ltr"=
><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=
=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr">=
<div><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bottom=
:0pt;margin-left:5pt"><img src=3D"https://drive.google.com/a/signalwire.com=
/uc?id=3D1BSHviBrhSusZ8tCNr3UYzQbWxD6RUudb&amp;export=3Ddownload"><br></p><=
p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;m=
argin-left:5pt"><span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,=
0,0);background-color:transparent;vertical-align:baseline;white-space:pre-w=
rap">Anthony Minessale =C2=A0| Founder and CEO</span></p><p dir=3D"ltr" sty=
le=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt"><=
span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,0,0);background-c=
olor:transparent;vertical-align:baseline;white-space:pre-wrap">SignalWire: =
</span><span style=3D"color:rgb(0,0,0);font-size:9pt;font-family:Arial;back=
ground-color:transparent;vertical-align:baseline;white-space:pre-wrap"><u><=
a href=3D"https://maps.google.com/?q=3D228+Hamilton+Ave+3rd+Floor,+Palo+Alt=
o,+CA+94303&amp;entry=3Dgmail&amp;source=3Dg" style=3D"color:rgb(17,85,204)=
" target=3D"_blank">228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303</a></u>=
</span></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-=
bottom:0pt;margin-left:5pt"><span style=3D"background-color:transparent;ver=
tical-align:baseline"><font color=3D"#0000ee" face=3D"Arial"><span style=3D=
"font-size:12px;white-space:pre-wrap"><u><a href=3D"https://youtu.be/WKjZkz=
gK4pc" target=3D"_blank">https://youtu.be/WKjZkzgK4pc</a></u></span></font>=
<br></span></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;mar=
gin-bottom:0pt;margin-left:5pt"><span style=3D"background-color:transparent=
;vertical-align:baseline"><font color=3D"#0000ee" face=3D"Arial"><span styl=
e=3D"font-size:12px;white-space:pre-wrap"><a href=3D"https://youtu.be/MwqpS=
l7luSM" target=3D"_blank">https://youtu.be/MwqpSl7luSM</a></span><br></font=
></span></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin=
-bottom:0pt;margin-left:5pt"><span style=3D"background-color:transparent;ve=
rtical-align:baseline"><a href=3D"https://www.youtube.com/watch?v=3Dl82JtCE=
Ckog" style=3D"color:rgb(17,85,204)" target=3D"_blank">https://youtu.be/l82=
JtCECkog</a><font color=3D"#0000ee" face=3D"Arial"><br></font></span></p><p=
 dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;ma=
rgin-left:5pt"><span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,0=
,0);background-color:transparent;vertical-align:baseline;white-space:pre-wr=
ap">Email: <a href=3D"mailto:anthm@signalwire.com" target=3D"_blank">anthm@=
signalwire.com</a></span></p><p style=3D"line-height:1.656;margin-top:0pt;m=
argin-bottom:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font-family:=
Arial;color:rgb(0,0,0);background-color:transparent;vertical-align:baseline=
;white-space:pre-wrap">Phone:  <a href=3D"tel:(262)%20309-8501" value=3D"+1=
2623098501" target=3D"_blank">262-309-8501</a> </span></p><p dir=3D"ltr" st=
yle=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt">=
<span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,0,0);background-=
color:transparent;vertical-align:baseline;white-space:pre-wrap">Website:</s=
pan><a href=3D"https://www.freeswitch.com/" style=3D"color:rgb(17,85,204)" =
target=3D"_blank"><span style=3D"font-size:9pt;font-family:Arial;color:rgb(=
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-wrap"> </span></a><a href=3D"https://www.signalwire.com/" style=3D"color:r=
gb(17,85,204)" target=3D"_blank"><span style=3D"font-size:9pt;font-family:A=
rial;background-color:transparent;vertical-align:baseline;white-space:pre-w=
rap">https://www.signalwire.com</span></a></p><div style=3D"font-size:small=
"><a href=3D"https://www.facebook.com/search/top/?q=3Dsignalwire" style=3D"=
color:rgb(17,85,204)" target=3D"_blank"><img src=3D"cid:ii_jm25x25z1_165d8b=
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ll"><br></div></div></div></div></div></div></div></div></div></div></div><=
/div></div></div></div>
</blockquote></div></div>

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From: Anthony Minessale <anthm@signalwire.com>
Date: Mon, 23 Mar 2020 18:38:24 -0500
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To: Justin Uberti <juberti@google.com>
Cc: Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] Thoughts on scope for ript
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On Mon, Mar 23, 2020 at 5:43 PM Justin Uberti <juberti@google.com> wrote:

>
> On Mon, Mar 23, 2020 at 3:24 PM Anthony Minessale <anthm@signalwire.com>
> wrote:
>
>> Also more consideration for live translation to sip/rtp/webrtc and
>> back-to-back scenarios.
>>
>> If calls are being loaded balanced to many servers you still probably
>> will want a way to keep track of what calls are on what server in a way
>> that goes out of scope of HTTP so the application doing the clustering will
>> bear a pretty big burden of solving that problem from behind the cluster.
>>
>> I read the memo about replacing SIP and was slightly disturbed that
>> RFC2833 (4733/4734) is the one thing salvaged ;)
>> That would be the first to go IMHO, its the leading cause of interop
>> issues in all of telephony and very rarely implemented to its full
>> potential not to mention there are some timestamp translation issues I
>> pointed out on the github discussion so that is it's own can of worms.
>>
>> Also, the description seems to be evolving more towards considerations
>> about making calls behave like a web app which is fine because that's the
>> GOAL but there is little consideration of actual trunking so far.  SIP
>> trunking is just a made-up marketing term to mirror original TDM trunking
>> which was actual trunking (bundling many calls into one stream to reduce
>> overhead) In reality SIP trunking is often just (a lot of SIP calls) We
>> have our SIP setup so you can send many calls to many different phone
>> numbers over a single registration and that causes introp issues because
>> most people don't comprehend this lol.
>>
>
>
>> So it seems like there could be some improvements that could clean up
>> issues with WebRTC and how web-app type environments can bust through nat
>> etc.
>> WebRTC is still compatible with SIP in the sense that you can carry a
>> WebRTC SDP over sip and set up calls but most of the time its SIP over
>> WebSockets (though FS and Kamailio can translate and negotiate WebRTC
>> sessions to other SIP transports)
>>
>> From a server perspective with density being a major concern, removing
>> all the need for dialogs and route headers is a good thing. What I also
>> think is critical to preserve is the server's ability to limit the media
>> formats etc sent by the sender.  Killing Offer/Answer is ok but the
>> replacement should at least preserve the notions that a client can offer
>> what it can do, the server can narrow that scope with its own requirements
>> and the client must work within those shared formats.  The server is the
>> one who has to create thousands of connections etc and must not be treated
>> as a simple peer-to-peer agent.
>>
>> Removing SRTP is another concern because it's pretty widely used now,
>> it's required in WebRTC and decryption in the middle was something I was
>> hoping we would go the other direction on and find ways to route encrypted
>> packets without needing to decode them.
>>
>
> One reason we are focusing on client-to-server vs peer-to-peer is that it
> avoids the interop issues associated with p2p (i.e., the difference between
> connecting to the server that you control vs various peers that you don't
> control). Also, WebRTC already works well for the peer-to-peer case.
>

As a server, I appreciate that because it makes scaling much easier to have
the power to dictate things.


>
> I also think that whatever we come up with to replace SRTP will probably
> be able to gateway to SRTP easily for systems that expect SRTP.
>

>> I think if you want to replace SIP trunking more considerations have to
>> be made for the mass translation of TDM/SIP/SS7 and how that works with
>> routing and clustering.  I think some of that is how we got into that boat
>> with SIP because it was only later that this was tried and then workarounds
>> abounded for years.
>>
>> On Mon, Mar 23, 2020 at 12:09 PM Justin Uberti <juberti=
>> 40google.com@dmarc.ietf.org> wrote:
>>
>>> In addition to Eric's comments, I think the biggest question regarding
>>> goals and scope is whether we focus on both client-to-server and
>>> server-to-server.
>>>
>>> If we are taking a fresh look at how all of IP calling works, we could
>>> find ourselves looking at:
>>> a) a replacement for SIP "RPCs" (i.e, how are commands reliably
>>> exchanged)
>>> b) a replacement for SIP operations and routing (i.e., the specific
>>> commands and routing mechanisms)
>>> c) a replacement for offer-answer (i.e., how do both sides agree on
>>> media formats)
>>> d) a replacement for SDP (i.e., the details of how media formats are
>>> described)
>>> e) a replacement for SRTP (i.e., how media is transmitted from point A
>>> to B)
>>> f) a replacement for RTP packetization (i.e., how media is encapsulated
>>> when transmitted)
>>>
>>> I think that server-to-server, essentially a NNI, requires us to
>>> consider all of these items, and that may ultimately make sense. However,
>>> if we focus initially on client-to-server, we can get by by only focusing
>>> on e) and f), as a-d) are under application control
>>>
>>> This could still provide a lot of value, since moving to a world where
>>> even private RTC services can be built atop HTTP would still be quite an
>>> accomplishment. (Compared to the current state of the world, where people
>>> deploying cloud RTC services need to source and deploy their own
>>> ICE/DTLS/SCTP/SRTP stacks and all the load
>>> balancing/authentication/failover tech that Jonathan mentions). It would
>>> also be complementary to WebRTC, which works pretty well for p2p
>>> applications, but struggles in c2s environments for the aforementioned
>>> reasons.
>>>
>>>
>>> On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg <jdrosen@five9.com>
>>> wrote:
>>>
>>>> With our BoF coming up this Thursday, I thought the time is right to
>>>> restart discussions with a focus in particular on SCOPE of the effort.
>>>>
>>>> For me its defined entirely by the GOAL. The goal is:
>>>>
>>>> It is possible to implement a ript server in public cloud, such that it
>>>> is nothing more than a web application as far as public cloud is concerned,
>>>> and therefore works with web-centric features such as Kubernetes, global
>>>> http load balancers, and service meshes - to name a few. The server
>>>> implements everything needed to make and receive voice and video calls.
>>>>
>>>> I do think the scope needs to be inclusive of server to server peering,
>>>> not just client to server. The client-to-server stuff defines the core, but
>>>> for me I'm really interested in building an alternative to sip trunking and
>>>> that requires solving for additional things (inbound load balancing,
>>>> inbound authentication, number routing). I think those can easily be in a
>>>> separate draft but I think they should be in scope.
>>>>
>>>> Thx,
>>>> Jonathan R.
>>>>
>>>> --
>>>> Jonathan Rosenberg
>>>> CTO and Head of AI, Five9
>>>>
>>>>
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>>>>
>>> --
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>>>
>>
>>
>> --
>>
>>
>> Anthony Minessale  | Founder and CEO
>>
>> SignalWire: *228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303
>> <https://maps.google.com/?q=228+Hamilton+Ave+3rd+Floor,+Palo+Alto,+CA+94303&entry=gmail&source=g>*
>>
>> *https://youtu.be/WKjZkzgK4pc <https://youtu.be/WKjZkzgK4pc>*
>>
>> https://youtu.be/MwqpSl7luSM
>>
>> https://youtu.be/l82JtCECkog
>> <https://www.youtube.com/watch?v=l82JtCECkog>
>>
>> Email: anthm@signalwire.com
>>
>> Phone: 262-309-8501 <(262)%20309-8501>
>>
>> Website: <https://www.freeswitch.com/>https://www.signalwire.com
>> <https://www.facebook.com/search/top/?q=signalwire>
>> <https://twitter.com/SignalWire>
>>
>>
>>

-- 


Anthony Minessale  | Founder and CEO

SignalWire: *228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303
<https://maps.google.com/?q=228+Hamilton+Ave+3rd+Floor,+Palo+Alto,+CA+94303&entry=gmail&source=g>*

*https://youtu.be/WKjZkzgK4pc <https://youtu.be/WKjZkzgK4pc>*

https://youtu.be/MwqpSl7luSM

https://youtu.be/l82JtCECkog <https://www.youtube.com/watch?v=l82JtCECkog>

Email: anthm@signalwire.com

Phone: 262-309-8501

Website: <https://www.freeswitch.com/>https://www.signalwire.com
<https://www.facebook.com/search/top/?q=signalwire>
<https://twitter.com/SignalWire>

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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, Mar 23, 2020 at 5:43 PM Justi=
n Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&g=
t; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div d=
ir=3D"ltr"><div></div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=
=3D"gmail_attr">On Mon, Mar 23, 2020 at 3:24 PM Anthony Minessale &lt;<a hr=
ef=3D"mailto:anthm@signalwire.com" target=3D"_blank">anthm@signalwire.com</=
a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0p=
x 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><d=
iv dir=3D"ltr"><div dir=3D"auto">Also more consideration for live translati=
on to sip/rtp/webrtc and back-to-back scenarios.</div><div dir=3D"auto"><br=
></div><div>If calls are being loaded balanced to many servers you still pr=
obably will want a way to keep track of what calls are on what server in a =
way that goes out of scope of HTTP so the application doing the clustering =
will bear a pretty big burden of solving that problem from behind the clust=
er.</div><div dir=3D"auto"><br></div><div>I read the memo about replacing S=
IP and was slightly disturbed that RFC2833 (4733/4734) is the one thing sal=
vaged ;)</div><div>That would be the first to go IMHO, its the leading caus=
e of interop issues in all of telephony and very rarely implemented to its =
full potential not to mention there are some timestamp translation issues I=
 pointed out on the github discussion so that is it&#39;s own can of worms.=
</div><div><br></div><div>Also, the description seems to be evolving more t=
owards considerations about making calls behave like a web app which is fin=
e because that&#39;s the GOAL but there is little consideration of actual t=
runking so far.=C2=A0 SIP trunking is just a made-up marketing term to mirr=
or original TDM trunking which was actual trunking (bundling many calls int=
o one stream to reduce overhead) In reality SIP trunking is often just (a l=
ot of SIP calls) We have our SIP setup so you can send many calls to many d=
ifferent phone numbers over a single registration and that causes introp is=
sues because most people don&#39;t comprehend this lol.</div></div></blockq=
uote><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"><di=
v dir=3D"ltr"><div></div><div>So it seems like there could be some improvem=
ents that could clean up issues with WebRTC and how web-app type environmen=
ts can bust through nat etc.</div><div>WebRTC is still compatible with SIP =
in the sense that you can carry a WebRTC SDP over sip and set up calls but =
most of the time its SIP over WebSockets (though FS and Kamailio can transl=
ate and negotiate WebRTC sessions to other SIP transports)<br></div><div><b=
r></div><div>From a server perspective with density being a major concern, =
removing all the need for dialogs and route headers is a good thing. What I=
 also think is critical to preserve is the server&#39;s ability to limit th=
e media formats etc sent by the sender.=C2=A0 Killing Offer/Answer is ok bu=
t the replacement should at least preserve the notions that a client can of=
fer what it can do, the server can narrow that scope with its own requireme=
nts and the client must work within those shared formats.=C2=A0 The server =
is the one who has to create thousands of connections etc and must not be t=
reated as a simple peer-to-peer agent.</div><div><br></div><div>Removing SR=
TP is another concern because it&#39;s pretty widely used now, it&#39;s req=
uired in WebRTC and decryption in the middle was something I was hoping we =
would go the other direction on and find ways to route encrypted packets wi=
thout needing to decode them.</div></div></blockquote><div><br></div><div>O=
ne reason we are focusing on client-to-server vs peer-to-peer is that it av=
oids the interop issues associated with p2p (i.e., the difference between c=
onnecting to the server that you control vs various peers that you don&#39;=
t control). Also, WebRTC already works well for the peer-to-peer case.=C2=
=A0</div></div></div></blockquote><div><br></div><div>As a server, I apprec=
iate that because it makes scaling much easier to have the power to dictate=
 things.</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 dir=3D"ltr"><div class=3D"gmail_quote"><div><br></div><div>I als=
o think that whatever we come up with to replace SRTP will probably be able=
 to gateway to SRTP easily for systems that expect SRTP.</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 dir=3D"l=
tr"><div class=3D"gmail_quote"><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 dir=3D"ltr"><div><br></div><div>I think if you want to replace S=
IP trunking more considerations have to be made for the mass translation of=
 TDM/SIP/SS7 and how that works with routing and clustering.=C2=A0 I think =
some of that is how we got into that boat with SIP because it was only late=
r that this was tried and then workarounds abounded for years.</div></div><=
br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon,=
 Mar 23, 2020 at 12:09 PM Justin Uberti &lt;juberti=3D<a href=3D"mailto:40g=
oogle.com@dmarc.ietf.org" target=3D"_blank">40google.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"><div=
 dir=3D"ltr">In addition to Eric&#39;s comments, I think the biggest questi=
on regarding goals and scope is whether we focus on both client-to-server a=
nd server-to-server.=C2=A0<div><br></div><div>If we are taking a fresh look=
 at how all of IP calling works, we could find ourselves looking at:</div><=
div><div></div><div></div><div>a) a replacement for=C2=A0<span>SIP</span>=
=C2=A0&quot;RPCs&quot; (i.e, how are commands reliably exchanged)</div><div=
>b) a replacement for SIP operations and routing (i.e., the specific comman=
ds and routing mechanisms)</div><div>c) a replacement for offer-answer (i.e=
., how do both sides agree on media formats)<br></div><div>d) a replacement=
 for=C2=A0<span>SDP (i.e., the details of how media formats are described)<=
/span><br></div><div>e) a replacement for SRTP (i.e., how media is transmit=
ted from point A to B)</div><div>f) a replacement for RTP packetization (i.=
e., how media is encapsulated when transmitted)</div><div><br></div><div>I =
think that server-to-server, essentially a NNI, requires us to consider all=
 of these items, and that may ultimately make sense. However, if we focus i=
nitially on client-to-server, we can get by by only focusing on e) and f), =
as a-d) are under application control</div><div><br></div><div>This could s=
till provide a lot of value, since moving to a world where even private RTC=
 services can be built atop HTTP would still be quite an accomplishment. (C=
ompared to the current state of the world, where people deploying cloud RTC=
 services need to source and deploy their own ICE/DTLS/SCTP/SRTP stacks and=
 all the load balancing/authentication/failover tech that Jonathan mentions=
). It would also be complementary to WebRTC, which works pretty well for p2=
p applications, but struggles in c2s environments for the aforementioned re=
asons.</div></div><div><br></div></div><br><div class=3D"gmail_quote"><div =
dir=3D"ltr" class=3D"gmail_attr">On Mon, Mar 23, 2020 at 7:21 AM Jonathan R=
osenberg &lt;<a href=3D"mailto:jdrosen@five9.com" target=3D"_blank">jdrosen=
@five9.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">With our BoF coming up this Thursday, I thought the time is righ=
t to restart discussions with a focus in particular on SCOPE of the effort.=
<br>
<br>
For me its defined entirely by the GOAL. The goal is:<br>
<br>
It is possible to implement a ript server in public cloud, such that it is =
nothing more than a web application as far as public cloud is concerned, an=
d therefore works with web-centric features such as Kubernetes, global http=
 load balancers, and service meshes - to name a few. The server implements =
everything needed to make and receive voice and video calls.<br>
<br>
I do think the scope needs to be inclusive of server to server peering, not=
 just client to server. The client-to-server stuff defines the core, but fo=
r me I&#39;m really interested in building an alternative to sip trunking a=
nd that requires solving for additional things (inbound load balancing, inb=
ound authentication, number routing). I think those can easily be in a sepa=
rate draft but I think they should be in scope.<br>
<br>
Thx,<br>
Jonathan R.<br>
<br>
--<br>
Jonathan Rosenberg<br>
CTO and Head of AI, Five9<br>
<br>
<br>
________________________________<br>
<br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
idential information of Five9 and/or its affiliated entities. Access by the=
 intended recipient only is authorized. Any liability arising from any part=
y acting, or refraining from acting, on any information contained in this e=
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ify the sender immediately, destroy the original transmission and its attac=
hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
...<br>
<br>
-- <br>
V3 mailing list<br>
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<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div><br clear=3D"all"><div><br></div>-- <br><div dir=3D"ltr"=
><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=
=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr">=
<div><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bottom=
:0pt;margin-left:5pt"><img src=3D"https://drive.google.com/a/signalwire.com=
/uc?id=3D1BSHviBrhSusZ8tCNr3UYzQbWxD6RUudb&amp;export=3Ddownload"><br></p><=
p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;m=
argin-left:5pt"><span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,=
0,0);background-color:transparent;vertical-align:baseline;white-space:pre-w=
rap">Anthony Minessale =C2=A0| Founder and CEO</span></p><p dir=3D"ltr" sty=
le=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt"><=
span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,0,0);background-c=
olor:transparent;vertical-align:baseline;white-space:pre-wrap">SignalWire: =
</span><span style=3D"color:rgb(0,0,0);font-size:9pt;font-family:Arial;back=
ground-color:transparent;vertical-align:baseline;white-space:pre-wrap"><u><=
a href=3D"https://maps.google.com/?q=3D228+Hamilton+Ave+3rd+Floor,+Palo+Alt=
o,+CA+94303&amp;entry=3Dgmail&amp;source=3Dg" style=3D"color:rgb(17,85,204)=
" target=3D"_blank">228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303</a></u>=
</span></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-=
bottom:0pt;margin-left:5pt"><span style=3D"background-color:transparent;ver=
tical-align:baseline"><font color=3D"#0000ee" face=3D"Arial"><span style=3D=
"font-size:12px;white-space:pre-wrap"><u><a href=3D"https://youtu.be/WKjZkz=
gK4pc" target=3D"_blank">https://youtu.be/WKjZkzgK4pc</a></u></span></font>=
<br></span></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;mar=
gin-bottom:0pt;margin-left:5pt"><span style=3D"background-color:transparent=
;vertical-align:baseline"><font color=3D"#0000ee" face=3D"Arial"><span styl=
e=3D"font-size:12px;white-space:pre-wrap"><a href=3D"https://youtu.be/MwqpS=
l7luSM" target=3D"_blank">https://youtu.be/MwqpSl7luSM</a></span><br></font=
></span></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin=
-bottom:0pt;margin-left:5pt"><span style=3D"background-color:transparent;ve=
rtical-align:baseline"><a href=3D"https://www.youtube.com/watch?v=3Dl82JtCE=
Ckog" style=3D"color:rgb(17,85,204)" target=3D"_blank">https://youtu.be/l82=
JtCECkog</a><font color=3D"#0000ee" face=3D"Arial"><br></font></span></p><p=
 dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;ma=
rgin-left:5pt"><span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,0=
,0);background-color:transparent;vertical-align:baseline;white-space:pre-wr=
ap">Email: <a href=3D"mailto:anthm@signalwire.com" target=3D"_blank">anthm@=
signalwire.com</a></span></p><p style=3D"line-height:1.656;margin-top:0pt;m=
argin-bottom:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font-family:=
Arial;color:rgb(0,0,0);background-color:transparent;vertical-align:baseline=
;white-space:pre-wrap">Phone:  <a href=3D"tel:(262)%20309-8501" value=3D"+1=
2623098501" target=3D"_blank">262-309-8501</a> </span></p><p dir=3D"ltr" st=
yle=3D"line-height:1.656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt">=
<span style=3D"font-size:9pt;font-family:Arial;color:rgb(0,0,0);background-=
color:transparent;vertical-align:baseline;white-space:pre-wrap">Website:</s=
pan><a href=3D"https://www.freeswitch.com/" style=3D"color:rgb(17,85,204)" =
target=3D"_blank"><span style=3D"font-size:9pt;font-family:Arial;color:rgb(=
0,0,0);background-color:transparent;vertical-align:baseline;white-space:pre=
-wrap"> </span></a><a href=3D"https://www.signalwire.com/" style=3D"color:r=
gb(17,85,204)" target=3D"_blank"><span style=3D"font-size:9pt;font-family:A=
rial;background-color:transparent;vertical-align:baseline;white-space:pre-w=
rap">https://www.signalwire.com</span></a></p><div style=3D"font-size:small=
"><a href=3D"https://www.facebook.com/search/top/?q=3Dsignalwire" style=3D"=
color:rgb(17,85,204)" target=3D"_blank"><img src=3D"cid:ii_jm25x25z1_165d8b=
71f782610b" width=3D"32" height=3D"32"></a>=C2=A0<a href=3D"https://twitter=
.com/SignalWire" style=3D"color:rgb(17,85,204)" target=3D"_blank"><img src=
=3D"cid:ii_jm25y73l3_165d8b7eee16c0ac" width=3D"32" height=3D"32"></a></div=
></div><div style=3D"font-size:small"><br></div><div style=3D"font-size:sma=
ll"><br></div></div></div></div></div></div></div></div></div></div></div><=
/div></div></div></div>
</blockquote></div></div>
</blockquote></div><br clear=3D"all"><div><br></div>-- <br><div dir=3D"ltr"=
 class=3D"gmail_signature"><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div=
 dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"l=
tr"><div><div dir=3D"ltr"><div><p dir=3D"ltr" style=3D"line-height:1.656;ma=
rgin-top:0pt;margin-bottom:0pt;margin-left:5pt"><img src=3D"https://drive.g=
oogle.com/a/signalwire.com/uc?id=3D1BSHviBrhSusZ8tCNr3UYzQbWxD6RUudb&amp;ex=
port=3Ddownload"><br></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-t=
op:0pt;margin-bottom:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font=
-family:Arial;color:rgb(0,0,0);background-color:transparent;vertical-align:=
baseline;white-space:pre-wrap">Anthony Minessale =C2=A0| Founder and CEO</s=
pan></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bot=
tom:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font-family:Arial;col=
or:rgb(0,0,0);background-color:transparent;vertical-align:baseline;white-sp=
ace:pre-wrap">SignalWire: </span><span style=3D"color:rgb(0,0,0);font-size:=
9pt;font-family:Arial;background-color:transparent;vertical-align:baseline;=
white-space:pre-wrap"><u><a href=3D"https://maps.google.com/?q=3D228+Hamilt=
on+Ave+3rd+Floor,+Palo+Alto,+CA+94303&amp;entry=3Dgmail&amp;source=3Dg" sty=
le=3D"color:rgb(17,85,204)" target=3D"_blank">228 Hamilton Ave 3rd Floor, P=
alo Alto, CA 94303</a></u></span></p><p dir=3D"ltr" style=3D"line-height:1.=
656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt"><span style=3D"backgr=
ound-color:transparent;vertical-align:baseline"><font color=3D"#0000ee" fac=
e=3D"Arial"><span style=3D"font-size:12px;white-space:pre-wrap"><u><a href=
=3D"https://youtu.be/WKjZkzgK4pc" target=3D"_blank">https://youtu.be/WKjZkz=
gK4pc</a></u></span></font><br></span></p><p dir=3D"ltr" style=3D"line-heig=
ht:1.656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt"><span style=3D"b=
ackground-color:transparent;vertical-align:baseline"><font color=3D"#0000ee=
" face=3D"Arial"><span style=3D"font-size:12px;white-space:pre-wrap"><a hre=
f=3D"https://youtu.be/MwqpSl7luSM" target=3D"_blank">https://youtu.be/MwqpS=
l7luSM</a></span><br></font></span></p><p dir=3D"ltr" style=3D"line-height:=
1.656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt"><span style=3D"back=
ground-color:transparent;vertical-align:baseline"><a href=3D"https://www.yo=
utube.com/watch?v=3Dl82JtCECkog" style=3D"color:rgb(17,85,204)" target=3D"_=
blank">https://youtu.be/l82JtCECkog</a><font color=3D"#0000ee" face=3D"Aria=
l"><br></font></span></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-t=
op:0pt;margin-bottom:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font=
-family:Arial;color:rgb(0,0,0);background-color:transparent;vertical-align:=
baseline;white-space:pre-wrap">Email: <a href=3D"mailto:anthm@signalwire.co=
m" target=3D"_blank">anthm@signalwire.com</a></span></p><p style=3D"line-he=
ight:1.656;margin-top:0pt;margin-bottom:0pt;margin-left:5pt"><span style=3D=
"font-size:9pt;font-family:Arial;color:rgb(0,0,0);background-color:transpar=
ent;vertical-align:baseline;white-space:pre-wrap">Phone:  262-309-8501 </sp=
an></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bott=
om:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font-family:Arial;colo=
r:rgb(0,0,0);background-color:transparent;vertical-align:baseline;white-spa=
ce:pre-wrap">Website:</span><a href=3D"https://www.freeswitch.com/" style=
=3D"color:rgb(17,85,204)" target=3D"_blank"><span style=3D"font-size:9pt;fo=
nt-family:Arial;color:rgb(0,0,0);background-color:transparent;vertical-alig=
n:baseline;white-space:pre-wrap"> </span></a><a href=3D"https://www.signalw=
ire.com/" style=3D"color:rgb(17,85,204)" target=3D"_blank"><span style=3D"f=
ont-size:9pt;font-family:Arial;background-color:transparent;vertical-align:=
baseline;white-space:pre-wrap">https://www.signalwire.com</span></a></p><di=
v style=3D"font-size:small"><a href=3D"https://www.facebook.com/search/top/=
?q=3Dsignalwire" style=3D"color:rgb(17,85,204)" target=3D"_blank"><img src=
=3D"cid:ii_jm25x25z1_165d8b71f782610b" width=3D"32" height=3D"32"></a>=C2=
=A0<a href=3D"https://twitter.com/SignalWire" style=3D"color:rgb(17,85,204)=
" target=3D"_blank"><img src=3D"cid:ii_jm25y73l3_165d8b7eee16c0ac" width=3D=
"32" height=3D"32"></a></div></div><div style=3D"font-size:small"><br></div=
><div style=3D"font-size:small"><br></div></div></div></div></div></div></d=
iv></div></div></div></div></div></div></div></div></div>

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Subject: [V3] E2E Security and P2P media
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> On Feb 22, 2020, at 3:00 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
> LACK OF E2E SECURITY
> Your charter specifies e2e security, but AFAICT, this is not
> provided by this specification, nor is it obvious how to enhance
> it to provide it. Indeed, it seems like there's not any support
> for a disjoint media path. You may recall that I made similar
> comments at the DISPATCH session in SIN, and I had understood
> that it was your intention to accommodate these cases, but
> I don't see anything here, which is somewhat disappointing.

This was in some version but accidentally got deleted out in editing but =
what I think we should do is something along the lines of following:

1.Include End-2-End encryption and authentication of the media packets =
in the base design. (and there would also still be Hop-bop provided by =
transport layer)=20

2. Design it to work well if you are using MLS keying or something like =
it but the keying would be out of scope.=20

That design would well to do things like cover the PERC WG uses cases. =
(but i am not advocating a solution like double - just we have one E2E =
that is keyed by something like MLS and a separate HbH that is keyed by =
TLS)=20

I don=E2=80=99t think we should directly take on P2P media to start with =
due to the complexity that an ICE like protocol adds. Most the uses =
cases driving this work fine in the same sort of way that HTTP works =
without any P2P.  Much of the =E2=80=9CP2P=E2=80=9D SIP and WeBRTC =
traffic actually goes via a TURN server or other tunnel and RIPT would =
support that sort of relay topology as long as we had a E2E crypto.=20

I do think it might be possible that we will be able to use WebRTC set =
up a data channel then run RIPT  over that data channel.=20



--Apple-Mail=_667CBC2B-630E-4225-B67C-684D853736E4
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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""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On Feb 22, 2020, at 3:00 PM, Eric Rescorla &lt;<a =
href=3D"mailto:ekr@rtfm.com" class=3D"">ekr@rtfm.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: =
14px; 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"">LACK OF E2E SECURITY</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">Your charter specifies e2e =
security, but AFAICT, this is not</span><br style=3D"caret-color: rgb(0, =
0, 0); font-family: Helvetica; font-size: 14px; 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""><span style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 14px; 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"">provided by this specification, nor is it obvious how to =
enhance</span><br style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 14px; 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""><span style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 14px; 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"">it to provide =
it. Indeed, it seems like there's not any support</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">for a disjoint media path. You =
may recall that I made similar</span><br style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 14px; 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""><span style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 14px; 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"">comments at the DISPATCH session in SIN, and I had =
understood</span><br style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 14px; 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""><span style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 14px; 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"">that it was =
your intention to accommodate these cases, but</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">I don't see anything here, which =
is somewhat disappointing.</span></div></blockquote></div><br =
class=3D""><div class=3D"">This was in some version but accidentally got =
deleted out in editing but what I think we should do is something along =
the lines of following:</div><div class=3D""><br class=3D""></div><div =
class=3D"">1.Include End-2-End encryption and authentication of the =
media packets in the base design. (and there would also still be Hop-bop =
provided by transport layer)&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D"">2. Design it to work well if you are =
using MLS keying or something like it but the keying would be out of =
scope.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">That design would well to do things like cover the PERC WG =
uses cases. (but i am not advocating a solution like double - just we =
have one E2E that is keyed by something like MLS and a separate HbH that =
is keyed by TLS)&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">I don=E2=80=99t think we should directly take on P2P media to =
start with due to the complexity that an ICE like protocol adds. Most =
the uses cases driving this work fine in the same sort of way that HTTP =
works without any P2P. &nbsp;Much of the =E2=80=9CP2P=E2=80=9D SIP and =
WeBRTC traffic actually goes via a TURN server or other tunnel and RIPT =
would support that sort of relay topology as long as we had a E2E =
crypto.&nbsp;</div><div class=3D""><br class=3D""></div><div class=3D"">I =
do think it might be possible that we will be able to use WebRTC set up =
a data channel then run RIPT &nbsp;over that data =
channel.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D""><br class=3D""></div></body></html>=

--Apple-Mail=_667CBC2B-630E-4225-B67C-684D853736E4--


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From: Jonathan Rosenberg <jdrosen@five9.com>
To: "v3@ietf.org" <v3@ietf.org>
CC: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: ekr's comment #1: load balancing
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Archived-At: <https://mailarchive.ietf.org/arch/msg/v3/haqjKwqWIHtEiUZiB-W_6n4SQ2Y>
Subject: [V3] ekr's comment #1: load balancing
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I figured for ease of following I'd take ekr's review and split it up into =
a few separate mails.
https://mailarchive.ietf.org/arch/msg/v3/V2BuGEZJpEMcAKcYLNHDMtV3i70/



ekr says:
-----------
> WEB ARCHITECTURE
This document talks multiple times about how it's designed to work
with existing HTTP infrastructure and there are a number of features
of this design that seem to be dictated by that. However, it seems to
me that you are making a number of surprising (and perhaps incorrect)
assumptions about architecture and in general this seems misguided.

- Transport Connections and HTTP Requests Need Not Fate Share There
seems to be an implicit assumption in this design that HTTP LBs have
semi-persistent routing, such that if request 1 and request 2 go to
different servers this is a "migration" event (S 8.9). This may be the
case in some architectures but it's not by any means guaranteed. The
LB is free to send every request to a different server, and Web
applications need to be designed to accommodate this.
(see, for instance:
https://www.oreilly.com/library/view/programming-google-app/9781491900246/c=
h01.html
.
section The Runtime Environment).

While this might in principle be compatible with RIPT as designed, it seems
unlikely to work well if each byway goes to a different server.
------

My assumption is that cloud load balancers can be configured to perform sti=
cky routing, *not* that all load balancers will do it by default. In other =
words, the assumption is that the ript server is being deployed into a clou=
d environment which is under the control (via configuration) of the company=
 that is deploying the server, and therefore its settings can be tuned to m=
ake ript work properly. Sticky routing is broadly supported. It is supporte=
d by AWS (https://docs.aws.amazon.com/elasticloadbalancing/latest/classic/e=
lb-sticky-sessions.html#enable-sticky-sessions-application), Google (https:=
//cloud.google.com/load-balancing/docs/backend-service#session_affinity) , =
Azure (https://docs.microsoft.com/en-us/azure/load-balancer/load-balancer-d=
istribution-mode) . I suspect the others do too but I didn't check. Many we=
b apps that were built on-prem and were stateful, but are now being run in =
cloud compute environments, have an identical need so I don't expect this f=
eature to be removed.

You are correct that ript wont work well if each byway aka media packet goe=
s to a different server. Though, this may depend on the application. For ap=
ps that don't really process the media in a highly stateful way, it might a=
ctually work fine. For stateful operations like transcoding it would obviou=
sly be bad, yes.

That said, I think what is needed here is a better statement on the constra=
ints. Instead of making things a hard rule like, "must work with existing c=
loud services without change", I think a better statement is perhaps someth=
ing like, "aligns with web design principals, protocols, and features of cl=
oud services in such a way that it minimizes the set of features which are =
unique to real-time".  Would that work better?



Thx,
Jonathan R.

--
Jonathan Rosenberg
CTO and Head of AI, Five9


________________________________

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Date: Wed, 25 Mar 2020 16:10:27 +1100
From: "Martin Thomson" <mt@lowentropy.net>
To: v3@ietf.org
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Subject: [V3] RIPT virtual meeting info for IETF 107
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Props to Sean Turner for setting a good example here.  Here is the info you will need for the meeting on Wed/Thu:

- Webex Link:
  https://ietf.webex.com/ietf/j.php?MTID=m77168c5c7592f30d7f369b49b47cfcf3
- Make sure your video is OFF
- Mute your mic unless you are speaking
- Use headsets to avoid echo
- Use Webex chat only to join the mic queue:
  +q adds you to the queue 
  -q removes you from the queue
- Add your name and affiliation to the virtual blue sheet in the Etherpad:
  https://etherpad.ietf.org/p/notes-ietf-107-ript?useMonospaceFont=true
- Join the Jabber room:
  xmpp:ript@jabber.ietf.org?join


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <mt@lowentropy.net>, "v3@ietf.org" <v3@ietf.org>
Thread-Topic: [V3] RIPT virtual meeting info for IETF 107
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Date: Wed, 25 Mar 2020 21:20:58 +1100
From: "Martin Thomson" <mt@lowentropy.net>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] RIPT virtual meeting info for IETF 107
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On Wed, Mar 25, 2020, at 18:31, Christer Holmberg wrote:
> >    Props to Sean Turner for setting a good example here.  Here is the info you will need for the meeting on Wed/Thu:
>   
> Just to verify: Is there going to be a V3 meeting on Wednesday, or are 
> you referring to some other meeting?

I am sorry, the meeting is on Thursday for a little over half the planet and Friday for the remainder.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <mt@lowentropy.net>, "v3@ietf.org" <v3@ietf.org>
Thread-Topic: [V3] RIPT virtual meeting info for IETF 107
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Date: Wed, 25 Mar 2020 10:48:36 +0000
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From: Victor Vasiliev <vasilvv@google.com>
Date: Wed, 25 Mar 2020 11:30:46 -0400
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Subject: Re: [V3] Thoughts on scope for ript
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I would like to second Justin's point here about the value of a
client-server RTP streaming by itself (e-f, though in-band c-d are also
valuable).  There's a lot of value in the world where I can take an HTTP
URL to some endpoint in the cloud and give it to, say, an IoT camera device
to use as a realtime media sink, or pass it to an off-the-shelf streaming
library to show a realtime media source on screen.

On Mon, Mar 23, 2020 at 1:09 PM Justin Uberti <juberti=
40google.com@dmarc.ietf.org> wrote:

> In addition to Eric's comments, I think the biggest question regarding
> goals and scope is whether we focus on both client-to-server and
> server-to-server.
>
> If we are taking a fresh look at how all of IP calling works, we could
> find ourselves looking at:
> a) a replacement for SIP "RPCs" (i.e, how are commands reliably exchanged)
> b) a replacement for SIP operations and routing (i.e., the specific
> commands and routing mechanisms)
> c) a replacement for offer-answer (i.e., how do both sides agree on media
> formats)
> d) a replacement for SDP (i.e., the details of how media formats are
> described)
> e) a replacement for SRTP (i.e., how media is transmitted from point A to
> B)
> f) a replacement for RTP packetization (i.e., how media is encapsulated
> when transmitted)
>
> I think that server-to-server, essentially a NNI, requires us to consider
> all of these items, and that may ultimately make sense. However, if we
> focus initially on client-to-server, we can get by by only focusing on e)
> and f), as a-d) are under application control
>
> This could still provide a lot of value, since moving to a world where
> even private RTC services can be built atop HTTP would still be quite an
> accomplishment. (Compared to the current state of the world, where people
> deploying cloud RTC services need to source and deploy their own
> ICE/DTLS/SCTP/SRTP stacks and all the load
> balancing/authentication/failover tech that Jonathan mentions). It would
> also be complementary to WebRTC, which works pretty well for p2p
> applications, but struggles in c2s environments for the aforementioned
> reasons.
>
>
> On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rosenberg <jdrosen@five9.com>
> wrote:
>
>> With our BoF coming up this Thursday, I thought the time is right to
>> restart discussions with a focus in particular on SCOPE of the effort.
>>
>> For me its defined entirely by the GOAL. The goal is:
>>
>> It is possible to implement a ript server in public cloud, such that it
>> is nothing more than a web application as far as public cloud is concerned,
>> and therefore works with web-centric features such as Kubernetes, global
>> http load balancers, and service meshes - to name a few. The server
>> implements everything needed to make and receive voice and video calls.
>>
>> I do think the scope needs to be inclusive of server to server peering,
>> not just client to server. The client-to-server stuff defines the core, but
>> for me I'm really interested in building an alternative to sip trunking and
>> that requires solving for additional things (inbound load balancing,
>> inbound authentication, number routing). I think those can easily be in a
>> separate draft but I think they should be in scope.
>>
>> Thx,
>> Jonathan R.
>>
>> --
>> Jonathan Rosenberg
>> CTO and Head of AI, Five9
>>
>>
>> ________________________________
>>
>> CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain
>> confidential information of Five9 and/or its affiliated entities. Access by
>> the intended recipient only is authorized. Any liability arising from any
>> party acting, or refraining from acting, on any information contained in
>> this e-mail is hereby excluded. If you are not the intended recipient,
>> please notify the sender immediately, destroy the original transmission and
>> its attachments and do not disclose the contents to any other person, use
>> it for any purpose, or store or copy the information in any medium.
>> Copyright in this e-mail and any attachments belongs to Five9 and/or its
>> affiliated entities..
>>
>> --
>> V3 mailing list
>> V3@ietf.org
>> https://www.ietf.org/mailman/listinfo/v3
>>
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>

--00000000000023856305a1af8ece
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<div dir=3D"ltr">I would like to second Justin&#39;s point here about the v=
alue of a client-server RTP streaming by itself (e-f, though in-band c-d ar=
e also valuable).=C2=A0 There&#39;s a lot of value in the world where I can=
 take an HTTP URL to some endpoint in the cloud and give it to, say, an IoT=
 camera device to use as a realtime media sink, or pass it to an off-the-sh=
elf streaming library to show a realtime media source on screen.</div><br><=
div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Mar=
 23, 2020 at 1:09 PM Justin Uberti &lt;juberti=3D<a href=3D"mailto:40google=
.com@dmarc.ietf.org" target=3D"_blank">40google.com@dmarc.ietf.org</a>&gt; =
wrote:<br></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 dir=
=3D"ltr">In addition to Eric&#39;s comments, I think the biggest question r=
egarding goals and scope is whether we focus on both client-to-server and s=
erver-to-server.=C2=A0<div><br></div><div>If we are taking a fresh look at =
how all of IP calling works, we could find ourselves looking at:</div><div>=
<div></div><div></div><div>a) a replacement for=C2=A0<span>SIP</span>=C2=A0=
&quot;RPCs&quot; (i.e, how are commands reliably exchanged)</div><div>b) a =
replacement for SIP operations and routing (i.e., the specific commands and=
 routing mechanisms)</div><div>c) a replacement for offer-answer (i.e., how=
 do both sides agree on media formats)<br></div><div>d) a replacement for=
=C2=A0<span>SDP (i.e., the details of how media formats are described)</spa=
n><br></div><div>e) a replacement for SRTP (i.e., how media is transmitted =
from point A to B)</div><div>f) a replacement for RTP packetization (i.e., =
how media is encapsulated when transmitted)</div><div><br></div><div>I thin=
k that server-to-server, essentially a NNI, requires us to consider all of =
these items, and that may ultimately make sense. However, if we focus initi=
ally on client-to-server, we can get by by only focusing on e) and f), as a=
-d) are under application control</div><div><br></div><div>This could still=
 provide a lot of value, since moving to a world where even private RTC ser=
vices can be built atop HTTP would still be quite an accomplishment. (Compa=
red to the current state of the world, where people deploying cloud RTC ser=
vices need to source and deploy their own ICE/DTLS/SCTP/SRTP stacks and all=
 the load balancing/authentication/failover tech that Jonathan mentions). I=
t would also be complementary to WebRTC, which works pretty well for p2p ap=
plications, but struggles in c2s environments for the aforementioned reason=
s.</div></div><div><br></div></div><br><div class=3D"gmail_quote"><div dir=
=3D"ltr" class=3D"gmail_attr">On Mon, Mar 23, 2020 at 7:21 AM Jonathan Rose=
nberg &lt;<a href=3D"mailto:jdrosen@five9.com" target=3D"_blank">jdrosen@fi=
ve9.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-lef=
t:1ex">With our BoF coming up this Thursday, I thought the time is right to=
 restart discussions with a focus in particular on SCOPE of the effort.<br>
<br>
For me its defined entirely by the GOAL. The goal is:<br>
<br>
It is possible to implement a ript server in public cloud, such that it is =
nothing more than a web application as far as public cloud is concerned, an=
d therefore works with web-centric features such as Kubernetes, global http=
 load balancers, and service meshes - to name a few. The server implements =
everything needed to make and receive voice and video calls.<br>
<br>
I do think the scope needs to be inclusive of server to server peering, not=
 just client to server. The client-to-server stuff defines the core, but fo=
r me I&#39;m really interested in building an alternative to sip trunking a=
nd that requires solving for additional things (inbound load balancing, inb=
ound authentication, number routing). I think those can easily be in a sepa=
rate draft but I think they should be in scope.<br>
<br>
Thx,<br>
Jonathan R.<br>
<br>
--<br>
Jonathan Rosenberg<br>
CTO and Head of AI, Five9<br>
<br>
<br>
________________________________<br>
<br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain conf=
idential information of Five9 and/or its affiliated entities. Access by the=
 intended recipient only is authorized. Any liability arising from any part=
y acting, or refraining from acting, on any information contained in this e=
-mail is hereby excluded. If you are not the intended recipient, please not=
ify the sender immediately, destroy the original transmission and its attac=
hments and do not disclose the contents to any other person, use it for any=
 purpose, or store or copy the information in any medium. Copyright in this=
 e-mail and any attachments belongs to Five9 and/or its affiliated entities=
..<br>
<br>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>

--00000000000023856305a1af8ece--


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From: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
Date: Wed, 25 Mar 2020 11:27:51 -0500
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To: Victor Vasiliev <vasilvv=40google.com@dmarc.ietf.org>
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Subject: Re: [V3] Thoughts on scope for ript
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This is interesting ...

On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=
40google.com@dmarc.ietf.org> wrote:

> I would like to second Justin's point here about the value of a
> client-server RTP streaming by itself (e-f, though in-band c-d are also
> valuable).  There's a lot of value in the world where I can take an HTTP
> URL to some endpoint in the cloud and give it to, say, an IoT camera device
> to use as a realtime media sink, or pass it to an off-the-shelf streaming
> library to show a realtime media source on screen.
>

I know most of my own concerns about RIPT media have started out with "this
won't be as good as RTP for media", but what I've been getting from RIPT
discussions is, that may not matter as much as I expected.

For a use case like Victor's, great media quality would be great, but good
enough media quality may be ... good enough.

Am I putting words in the mouths of the proponents?

Best,

Spencer

--00000000000006474b05a1b05b85
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<div dir=3D"ltr"><div dir=3D"ltr">This is interesting ...</div><br><div cla=
ss=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Wed, Mar 25, 20=
20 at 10:31 AM Victor Vasiliev &lt;vasilvv=3D<a href=3D"mailto:40google.com=
@dmarc.ietf.org">40google.com@dmarc.ietf.org</a>&gt; wrote:<br></div><block=
quote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1=
px solid rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr">I would like t=
o second Justin&#39;s point here about the value of a client-server RTP str=
eaming by itself (e-f, though in-band c-d are also valuable).=C2=A0 There&#=
39;s a lot of value in the world where I can take an HTTP URL to some endpo=
int in the cloud and give it to, say, an IoT camera device to use as a real=
time media sink, or pass it to an off-the-shelf streaming library to show a=
 realtime media source on screen.</div></blockquote><div><br></div><div>I k=
now most of my own concerns about RIPT media have started out with &quot;th=
is won&#39;t be as good as RTP for media&quot;, but what I&#39;ve been gett=
ing from RIPT discussions is, that may not matter as much as I expected.=C2=
=A0</div><div><br></div><div>For a use case like Victor&#39;s, great media =
quality would be great, but good enough media quality may be ... good enoug=
h.=C2=A0</div><div><br></div><div>Am I putting words in the mouths of the p=
roponents?</div><div><br></div><div>Best,</div><div><br></div><div>Spencer=
=C2=A0</div></div></div>

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From: Anthony Minessale <anthm@signalwire.com>
Date: Wed, 25 Mar 2020 11:53:57 -0500
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Cc: Victor Vasiliev <vasilvv=40google.com@dmarc.ietf.org>, "v3@ietf.org" <v3@ietf.org>, Jonathan Rosenberg <jdrosen@five9.com>, Justin Uberti <juberti=40google.com@dmarc.ietf.org>
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Subject: Re: [V3] Thoughts on scope for ript
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On Wed, Mar 25, 2020 at 11:28 AM Spencer Dawkins at IETF <
spencerdawkins.ietf@gmail.com> wrote:

> This is interesting ...
>
> On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=
> 40google.com@dmarc.ietf.org> wrote:
>
>> I would like to second Justin's point here about the value of a
>> client-server RTP streaming by itself (e-f, though in-band c-d are also
>> valuable).  There's a lot of value in the world where I can take an HTTP
>> URL to some endpoint in the cloud and give it to, say, an IoT camera device
>> to use as a realtime media sink, or pass it to an off-the-shelf streaming
>> library to show a realtime media source on screen.
>>
>
> I know most of my own concerns about RIPT media have started out with
> "this won't be as good as RTP for media", but what I've been getting from
> RIPT discussions is, that may not matter as much as I expected.
>
> For a use case like Victor's, great media quality would be great, but good
> enough media quality may be ... good enough.
>
> Am I putting words in the mouths of the proponents?
>
>
Being "as good as RTP" is relative and so is "quality" There is quality
like how it sounds and quality like (how many packets did I get that I am
supposed to be getting and are they on time, are they dejittered and are
they dropped when they are too old)
If the media signal is in good shape then the second type makes all the
difference in preserving that.

One thing about RTP that is helpful but often not taken advantage of is
that you can pass it through as is even out of order and messed up and the
final thing to receive the stream can put things back together.
If something in the middle just cleanses it and rewrites all the timestamps
you get a permanently messed-up stream that can never be fixed.
Sometimes you want to dejitter and rewrite the packets when you know the
next hop has no way to do it but otherwise usually only when you are the
final terminating leg.

Everything RTP can do to make sure you have these features should be
duplicated or maybe an encapsulation mode to just carry old school RTP as a
payload so 2 aincent devices can still exchange RTP across a RIPT-wormhole.







> Best,
>
> Spencer
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>


-- 


Anthony Minessale  | Founder and CEO

SignalWire: *228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303
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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 Wed, Mar 25, 2020 at 11:28 AM Spen=
cer Dawkins at IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com">sp=
encerdawkins.ietf@gmail.com</a>&gt; wrote:<br></div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,=
204,204);padding-left:1ex"><div dir=3D"ltr"><div dir=3D"ltr">This is intere=
sting ...</div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gma=
il_attr">On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev &lt;vasilvv=3D<a =
href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_blank">40google.com@=
dmarc.ietf.org</a>&gt; wrote:<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 dir=3D"ltr">I would like to second Justin&#39;s point he=
re about the value of a client-server RTP streaming by itself (e-f, though =
in-band c-d are also valuable).=C2=A0 There&#39;s a lot of value in the wor=
ld where I can take an HTTP URL to some endpoint in the cloud and give it t=
o, say, an IoT camera device to use as a realtime media sink, or pass it to=
 an off-the-shelf streaming library to show a realtime media source on scre=
en.</div></blockquote><div><br></div><div>I know most of my own concerns ab=
out RIPT media have started out with &quot;this won&#39;t be as good as RTP=
 for media&quot;, but what I&#39;ve been getting from RIPT discussions is, =
that may not matter as much as I expected.=C2=A0</div><div><br></div><div>F=
or a use case like Victor&#39;s, great media quality would be great, but go=
od enough media quality may be ... good enough.=C2=A0</div><div><br></div><=
div>Am I putting words in the mouths of the proponents?</div><div><br></div=
></div></div></blockquote><div><br></div><div>Being &quot;as good as RTP&qu=
ot; is relative and so is &quot;quality&quot; There is quality like how it =
sounds and quality like (how many packets did I get that I am supposed to b=
e getting and are they on time, are they dejittered and are they dropped wh=
en they are too old)</div><div>If the media signal is in good shape then th=
e second type makes all the difference in preserving that.</div><div><br></=
div><div>One thing about RTP that is helpful but often not taken advantage =
of is that you can pass it through as is even out of order and messed up an=
d the final thing to receive the stream can put things back together.</div>=
<div>If something in the middle just cleanses it and rewrites all the times=
tamps you get a permanently messed-up stream that can never be fixed.</div>=
<div>Sometimes you want to dejitter and rewrite the packets when you know t=
he next hop has no way to do it but otherwise usually only when you are the=
 final terminating leg.</div><div><br></div><div>Everything RTP can do to m=
ake sure you have these features should be duplicated or maybe an encapsula=
tion mode to just carry old school RTP as a payload so 2 aincent=C2=A0devic=
es can still exchange RTP across a RIPT-wormhole.</div><div><br></div><div>=
<br></div><div><br></div><div><br></div><div><br></div><div>=C2=A0</div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-lef=
t:1px solid rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr"><div class=
=3D"gmail_quote"><div></div><div>Best,</div><div><br></div><div>Spencer=C2=
=A0</div></div></div>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div><br clear=3D"all"><div><br></div>-- <br><div dir=3D"ltr"=
 class=3D"gmail_signature"><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div=
 dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=3D"l=
tr"><div><div dir=3D"ltr"><div><p dir=3D"ltr" style=3D"line-height:1.656;ma=
rgin-top:0pt;margin-bottom:0pt;margin-left:5pt"><img src=3D"https://drive.g=
oogle.com/a/signalwire.com/uc?id=3D1BSHviBrhSusZ8tCNr3UYzQbWxD6RUudb&amp;ex=
port=3Ddownload"><br></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-t=
op:0pt;margin-bottom:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font=
-family:Arial;color:rgb(0,0,0);background-color:transparent;vertical-align:=
baseline;white-space:pre-wrap">Anthony Minessale =C2=A0| Founder and CEO</s=
pan></p><p dir=3D"ltr" style=3D"line-height:1.656;margin-top:0pt;margin-bot=
tom:0pt;margin-left:5pt"><span style=3D"font-size:9pt;font-family:Arial;col=
or:rgb(0,0,0);background-color:transparent;vertical-align:baseline;white-sp=
ace:pre-wrap">SignalWire: </span><span style=3D"color:rgb(0,0,0);font-size:=
9pt;font-family:Arial;background-color:transparent;vertical-align:baseline;=
white-space:pre-wrap"><u><a href=3D"https://maps.google.com/?q=3D228+Hamilt=
on+Ave+3rd+Floor,+Palo+Alto,+CA+94303&amp;entry=3Dgmail&amp;source=3Dg" sty=
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From: Justin Uberti <juberti@google.com>
Date: Wed, 25 Mar 2020 09:55:28 -0700
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To: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
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Subject: Re: [V3] Thoughts on scope for ript
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On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at IETF <
spencerdawkins.ietf@gmail.com> wrote:

> This is interesting ...
>
> On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=
> 40google.com@dmarc.ietf.org> wrote:
>
>> I would like to second Justin's point here about the value of a
>> client-server RTP streaming by itself (e-f, though in-band c-d are also
>> valuable).  There's a lot of value in the world where I can take an HTTP
>> URL to some endpoint in the cloud and give it to, say, an IoT camera device
>> to use as a realtime media sink, or pass it to an off-the-shelf streaming
>> library to show a realtime media source on screen.
>>
>
> I know most of my own concerns about RIPT media have started out with
> "this won't be as good as RTP for media", but what I've been getting from
> RIPT discussions is, that may not matter as much as I expected.
>
> For a use case like Victor's, great media quality would be great, but good
> enough media quality may be ... good enough.
>
> Am I putting words in the mouths of the proponents?
>
> Some applications may tolerate somewhat lower quality in fallback
scenarios, but I think we should be aiming for RTP performance with HTTP
ease of deployment.

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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 Wed, Mar 25, 2020 at 9:28 AM Spenc=
er Dawkins at IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" tar=
get=3D"_blank">spencerdawkins.ietf@gmail.com</a>&gt; wrote:<br></div><block=
quote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1=
px solid rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr"><div dir=3D"lt=
r">This is interesting ...</div><br><div class=3D"gmail_quote"><div dir=3D"=
ltr" class=3D"gmail_attr">On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev =
&lt;vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_bla=
nk">40google.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 rg=
b(204,204,204);padding-left:1ex"><div dir=3D"ltr">I would like to second Ju=
stin&#39;s point here about the value of a client-server RTP streaming by i=
tself (e-f, though in-band c-d are also valuable).=C2=A0 There&#39;s a lot =
of value in the world where I can take an HTTP URL to some endpoint in the =
cloud and give it to, say, an IoT camera device to use as a realtime media =
sink, or pass it to an off-the-shelf streaming library to show a realtime m=
edia source on screen.</div></blockquote><div><br></div><div>I know most of=
 my own concerns about RIPT media have started out with &quot;this won&#39;=
t be as good as RTP for media&quot;, but what I&#39;ve been getting from RI=
PT discussions is, that may not matter as much as I expected.=C2=A0</div><d=
iv><br></div><div>For a use case like Victor&#39;s, great media quality wou=
ld be great, but good enough media quality may be ... good enough.=C2=A0</d=
iv><div><br></div><div>Am I putting words in the mouths of the proponents?<=
/div><div><br></div></div></div></blockquote><div>Some applications may tol=
erate somewhat lower quality in fallback scenarios, but I think we should b=
e aiming for RTP performance with HTTP ease of deployment.</div><div>=C2=A0=
</div></div></div>

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From: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
Date: Wed, 25 Mar 2020 14:25:45 -0500
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To: Justin Uberti <juberti@google.com>
Cc: Victor Vasiliev <vasilvv@google.com>, Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] Thoughts on scope for ript
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Replying to Justin, after reading Victor's reply - thank you both for
replying ...

On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti <juberti@google.com> wrote:

>
>
> On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at IETF <
> spencerdawkins.ietf@gmail.com> wrote:
>
>> This is interesting ...
>>
>> On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=
>> 40google.com@dmarc.ietf.org> wrote:
>>
>>> I would like to second Justin's point here about the value of a
>>> client-server RTP streaming by itself (e-f, though in-band c-d are also
>>> valuable).  There's a lot of value in the world where I can take an HTTP
>>> URL to some endpoint in the cloud and give it to, say, an IoT camera device
>>> to use as a realtime media sink, or pass it to an off-the-shelf streaming
>>> library to show a realtime media source on screen.
>>>
>>
>> I know most of my own concerns about RIPT media have started out with
>> "this won't be as good as RTP for media", but what I've been getting from
>> RIPT discussions is, that may not matter as much as I expected.
>>
>> For a use case like Victor's, great media quality would be great, but
>> good enough media quality may be ... good enough.
>>
>> Am I putting words in the mouths of the proponents?
>>
>> Some applications may tolerate somewhat lower quality in fallback
> scenarios, but I think we should be aiming for RTP performance with HTTP
> ease of deployment.
>

I was actually hoping that wasn't the case, because ISTM that the question
whether RTP performance over HTTP is possible/likely is going to suck up a
LOT of the discussion time during the BOF ...

Best,

Spencer

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<div dir=3D"ltr"><div dir=3D"ltr">Replying to Justin, after reading Victor&=
#39;s reply - thank you both for replying ...</div><br><div class=3D"gmail_=
quote"><div dir=3D"ltr" class=3D"gmail_attr">On Wed, Mar 25, 2020 at 11:55 =
AM Justin Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.c=
om</a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex=
"><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 Wed, Mar 25, 2020 at 9:28 AM Spe=
ncer Dawkins at IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" t=
arget=3D"_blank">spencerdawkins.ietf@gmail.com</a>&gt; wrote:<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 dir=3D"ltr"><div dir=3D"=
ltr">This is interesting ...</div><br><div class=3D"gmail_quote"><div dir=
=3D"ltr" class=3D"gmail_attr">On Wed, Mar 25, 2020 at 10:31 AM Victor Vasil=
iev &lt;vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"=
_blank">40google.com@dmarc.ietf.org</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 dir=3D"ltr">I would like to second =
Justin&#39;s point here about the value of a client-server RTP streaming by=
 itself (e-f, though in-band c-d are also valuable).=C2=A0 There&#39;s a lo=
t of value in the world where I can take an HTTP URL to some endpoint in th=
e cloud and give it to, say, an IoT camera device to use as a realtime medi=
a sink, or pass it to an off-the-shelf streaming library to show a realtime=
 media source on screen.</div></blockquote><div><br></div><div>I know most =
of my own concerns about RIPT media have started out with &quot;this won&#3=
9;t be as good as RTP for media&quot;, but what I&#39;ve been getting from =
RIPT discussions is, that may not matter as much as I expected.=C2=A0</div>=
<div><br></div><div>For a use case like Victor&#39;s, great media quality w=
ould be great, but good enough media quality may be ... good enough.=C2=A0<=
/div><div><br></div><div>Am I putting words in the mouths of the proponents=
?</div><div><br></div></div></div></blockquote><div>Some applications may t=
olerate somewhat lower quality in fallback scenarios, but I think we should=
 be aiming for RTP performance with HTTP ease of deployment.</div></div></d=
iv></blockquote><div><br></div><div>I was actually hoping that wasn&#39;t t=
he case, because ISTM that the question whether RTP performance over HTTP i=
s possible/likely is going to suck up a LOT of the discussion time during t=
he BOF ...=C2=A0=C2=A0</div><div><br></div><div>Best,</div><div><br></div><=
div>Spencer</div></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Wed, 25 Mar 2020 12:31:37 -0700
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To: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
Cc: Victor Vasiliev <vasilvv@google.com>, Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] Thoughts on scope for ript
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On Wed, Mar 25, 2020 at 12:26 PM Spencer Dawkins at IETF <
spencerdawkins.ietf@gmail.com> wrote:

> Replying to Justin, after reading Victor's reply - thank you both for
> replying ...
>
> On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti <juberti@google.com> wrote:
>
>>
>>
>> On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at IETF <
>> spencerdawkins.ietf@gmail.com> wrote:
>>
>>> This is interesting ...
>>>
>>> On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=
>>> 40google.com@dmarc.ietf.org> wrote:
>>>
>>>> I would like to second Justin's point here about the value of a
>>>> client-server RTP streaming by itself (e-f, though in-band c-d are also
>>>> valuable).  There's a lot of value in the world where I can take an HTTP
>>>> URL to some endpoint in the cloud and give it to, say, an IoT camera device
>>>> to use as a realtime media sink, or pass it to an off-the-shelf streaming
>>>> library to show a realtime media source on screen.
>>>>
>>>
>>> I know most of my own concerns about RIPT media have started out with
>>> "this won't be as good as RTP for media", but what I've been getting from
>>> RIPT discussions is, that may not matter as much as I expected.
>>>
>>> For a use case like Victor's, great media quality would be great, but
>>> good enough media quality may be ... good enough.
>>>
>>> Am I putting words in the mouths of the proponents?
>>>
>>> Some applications may tolerate somewhat lower quality in fallback
>> scenarios, but I think we should be aiming for RTP performance with HTTP
>> ease of deployment.
>>
>
> I was actually hoping that wasn't the case, because ISTM that the question
> whether RTP performance over HTTP is possible/likely is going to suck up a
> LOT of the discussion time during the BOF ...
>

To Eric's point, "H3 is not widely deployed yet, and so we should think
about how we want things to look". I think it's entirely possible to
achieve RTP performance over H3, and hopefully also have a reasonable
fallback when only H2 is available.

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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 Wed, Mar 25, 2020 at 12:26 PM Spen=
cer Dawkins at IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" ta=
rget=3D"_blank">spencerdawkins.ietf@gmail.com</a>&gt; wrote:<br></div><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 dir=3D"ltr"><div dir=3D"l=
tr">Replying to Justin, after reading Victor&#39;s reply - thank you both f=
or replying ...</div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=
=3D"gmail_attr">On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti &lt;<a href=
=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.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"><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 Wed, Mar 25, 2020 at 9:28 AM Spencer Dawki=
ns at IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" target=3D"_=
blank">spencerdawkins.ietf@gmail.com</a>&gt; wrote:<br></div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid=
 rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr"><div dir=3D"ltr">This =
is interesting ...</div><br><div class=3D"gmail_quote"><div dir=3D"ltr" cla=
ss=3D"gmail_attr">On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev &lt;vasi=
lvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_blank">40go=
ogle.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"><div dir=3D"ltr">I would like to second Justin&#39;s=
 point here about the value of a client-server RTP streaming by itself (e-f=
, though in-band c-d are also valuable).=C2=A0 There&#39;s a lot of value i=
n the world where I can take an HTTP URL to some endpoint in the cloud and =
give it to, say, an IoT camera device to use as a realtime media sink, or p=
ass it to an off-the-shelf streaming library to show a realtime media sourc=
e on screen.</div></blockquote><div><br></div><div>I know most of my own co=
ncerns about RIPT media have started out with &quot;this won&#39;t be as go=
od as RTP for media&quot;, but what I&#39;ve been getting from RIPT discuss=
ions is, that may not matter as much as I expected.=C2=A0</div><div><br></d=
iv><div>For a use case like Victor&#39;s, great media quality would be grea=
t, but good enough media quality may be ... good enough.=C2=A0</div><div><b=
r></div><div>Am I putting words in the mouths of the proponents?</div><div>=
<br></div></div></div></blockquote><div>Some applications may tolerate some=
what lower quality in fallback scenarios, but I think we should be aiming f=
or RTP performance with HTTP ease of deployment.</div></div></div></blockqu=
ote><div><br></div><div>I was actually hoping that wasn&#39;t the case, bec=
ause ISTM that the question whether RTP performance over HTTP is possible/l=
ikely is going to suck up a LOT of the discussion time during the BOF ...=
=C2=A0=C2=A0</div></div></div></blockquote><div><br></div><div>To Eric&#39;=
s point, &quot;H3 is not widely deployed yet, and so we should think about =
how we want things to look&quot;. I think it&#39;s entirely possible to ach=
ieve RTP performance over H3, and hopefully also have a reasonable fallback=
 when only H2 is available.</div></div></div>

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Thread-Topic: [V3] Thoughts on scope for ript
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Subject: Re: [V3] Thoughts on scope for ript
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From: Justin Uberti <juberti@google.com>
Date: Wed, 25 Mar 2020 12:49:55 -0700
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To: Stephan Wenger <stewe@stewe.org>
Cc: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>, Victor Vasiliev <vasilvv@google.com>,  Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] Thoughts on scope for ript
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On Wed, Mar 25, 2020 at 12:38 PM Stephan Wenger <stewe@stewe.org> wrote:

> Hi Justin, all:
>
>
>
> *From: *V3 <v3-bounces@ietf.org> on behalf of Justin Uberti <juberti=3D
> 40google.com@dmarc.ietf.org>
> *Date: *Wednesday, March 25, 2020 at 12:32
> *To: *Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
> *Cc: *Victor Vasiliev <vasilvv@google.com>, Jonathan Rosenberg <
> jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
> *Subject: *Re: [V3] Thoughts on scope for ript
>
>
>
>
>
>
>
> On Wed, Mar 25, 2020 at 12:26 PM Spencer Dawkins at IETF <
> spencerdawkins.ietf@gmail.com> wrote:
>
> Replying to Justin, after reading Victor's reply - thank you both for
> replying ...
>
>
>
> On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti <juberti@google.com> wrote=
:
>
>
>
>
>
> On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at IETF <
> spencerdawkins.ietf@gmail.com> wrote:
>
> This is interesting ...
>
>
>
> On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=3D
> 40google.com@dmarc.ietf.org> wrote:
>
> I would like to second Justin's point here about the value of a
> client-server RTP streaming by itself (e-f, though in-band c-d are also
> valuable).  There's a lot of value in the world where I can take an HTTP
> URL to some endpoint in the cloud and give it to, say, an IoT camera devi=
ce
> to use as a realtime media sink, or pass it to an off-the-shelf streaming
> library to show a realtime media source on screen.
>
>
>
> I know most of my own concerns about RIPT media have started out with
> "this won't be as good as RTP for media", but what I've been getting from
> RIPT discussions is, that may not matter as much as I expected.
>
>
>
> For a use case like Victor's, great media quality would be great, but goo=
d
> enough media quality may be ... good enough.
>
>
>
> Am I putting words in the mouths of the proponents?
>
>
>
> Some applications may tolerate somewhat lower quality in fallback
> scenarios, but I think we should be aiming for RTP performance with HTTP
> ease of deployment.
>
>
>
> I was actually hoping that wasn't the case, because ISTM that the questio=
n
> whether RTP performance over HTTP is possible/likely is going to suck up =
a
> LOT of the discussion time during the BOF ...
>
>
>
> To Eric's point, "H3 is not widely deployed yet, and so we should think
> about how we want things to look". I think it's entirely possible to
> achieve RTP performance over H3, and hopefully also have a reasonable
> fallback when only H2 is available.
>
>
>
> [Stephan]: many video conferencing technologies (including Vidyo, Zoom),
> have previously, or are currently supporting http-based media transport,
> often for years or even decades.  They are almost never enabled by any
> professional deployer, because the resulting QoS is user-perceived as so
> bad that the marketing guys decided that offering no video/voice (and
> instead make people pick of the phone) is better for product reputation
> than offer the degraded QoS.
>
> Do any of you guys have hard numbers what sort of delay h3-based transpor=
t
> can achieve on today=E2=80=99s infrastructure?
>

With H3 we have lots of options; there are lots of details to work through
here but given that the substrate is QUIC there is no theoretical reason
why RTP performance isn't possible.

Regarding HTTP media transport in existing products (i.e., non-H3), you are
correct that there are indeed limitations and as such providers do what
they can to avoid this option, but it is nowhere near as grim as you make
it out to be. Recall that the first web version of Zoom exclusively used
websockets for transport <https://webrtchacks.com/zoom-avoids-using-webrtc/=
>
.

>

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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 Wed, Mar 25, 2020 at 12:38 PM Step=
han Wenger &lt;<a href=3D"mailto:stewe@stewe.org">stewe@stewe.org</a>&gt; w=
rote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0p=
x 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">





<div lang=3D"EN-US">
<div class=3D"gmail-m_-9050459078766760524WordSection1">
<p class=3D"MsoNormal">Hi Justin, all:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div style=3D"border-right:none;border-bottom:none;border-left:none;border-=
top:1pt solid rgb(181,196,223);padding:3pt 0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:12pt;color:black">From: =
</span></b><span style=3D"font-size:12pt;color:black">V3 &lt;<a href=3D"mai=
lto:v3-bounces@ietf.org" target=3D"_blank">v3-bounces@ietf.org</a>&gt; on b=
ehalf of Justin Uberti &lt;juberti=3D<a href=3D"mailto:40google.com@dmarc.i=
etf.org" target=3D"_blank">40google.com@dmarc.ietf.org</a>&gt;<br>
<b>Date: </b>Wednesday, March 25, 2020 at 12:32<br>
<b>To: </b>Spencer Dawkins at IETF &lt;<a href=3D"mailto:spencerdawkins.iet=
f@gmail.com" target=3D"_blank">spencerdawkins.ietf@gmail.com</a>&gt;<br>
<b>Cc: </b>Victor Vasiliev &lt;<a href=3D"mailto:vasilvv@google.com" target=
=3D"_blank">vasilvv@google.com</a>&gt;, Jonathan Rosenberg &lt;<a href=3D"m=
ailto:jdrosen@five9.com" target=3D"_blank">jdrosen@five9.com</a>&gt;, &quot=
;<a href=3D"mailto:v3@ietf.org" target=3D"_blank">v3@ietf.org</a>&quot; &lt=
;<a href=3D"mailto:v3@ietf.org" target=3D"_blank">v3@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [V3] Thoughts on scope for ript<u></u><u></u></span></p=
>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 12:26 PM Spencer Dawkins at =
IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" target=3D"_blank"=
>spencerdawkins.ietf@gmail.com</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0in 0in 0in 6pt;margin-left:4=
.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal">Replying to Justin, after reading Victor&#39;s reply=
 - thank you both for replying ...<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti &lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0in 0in 0in 6pt;margin-left:4=
.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at I=
ETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" target=3D"_blank">=
spencerdawkins.ietf@gmail.com</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0in 0in 0in 6pt;margin-left:4=
.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal">This is interesting ...<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev &lt=
;vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_blank"=
>40google.com@dmarc.ietf.org</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote style=3D"border-top:none;border-right:none;border-bottom:none;b=
order-left:1pt solid rgb(204,204,204);padding:0in 0in 0in 6pt;margin-left:4=
.8pt;margin-right:0in">
<div>
<p class=3D"MsoNormal">I would like to second Justin&#39;s point here about=
 the value of a client-server RTP streaming by itself (e-f, though in-band =
c-d are also valuable).=C2=A0 There&#39;s a lot of value in the world where=
 I can take an HTTP URL to some endpoint in the
 cloud and give it to, say, an IoT camera device to use as a realtime media=
 sink, or pass it to an off-the-shelf streaming library to show a realtime =
media source on screen.<u></u><u></u></p>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I know most of my own concerns about RIPT media have=
 started out with &quot;this won&#39;t be as good as RTP for media&quot;, b=
ut what I&#39;ve been getting from RIPT discussions is, that may not matter=
 as much as I expected.=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">For a use case like Victor&#39;s, great media qualit=
y would be great, but good enough media quality may be ... good enough.=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">Am I putting words in the mouths of the proponents?<=
u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal">Some applications may tolerate somewhat lower qualit=
y in fallback scenarios, but I think we should be aiming for RTP performanc=
e with HTTP ease of deployment.<u></u><u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I was actually hoping that wasn&#39;t the case, beca=
use ISTM that the question whether RTP performance over HTTP is possible/li=
kely is going to suck up a LOT of the discussion time during the BOF ...=C2=
=A0=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To Eric&#39;s point, &quot;H3 is not widely deployed=
 yet, and so we should think about how we want things to look&quot;. I thin=
k it&#39;s entirely possible to achieve RTP performance over H3, and hopefu=
lly also have a reasonable fallback when only H2 is
 available.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">[Stephan]: many video conferencing technologies (inc=
luding Vidyo, Zoom), have previously, or are currently supporting http-base=
d media transport, often for years or even decades.=C2=A0 They are almost n=
ever enabled by any professional deployer,
 because the resulting QoS is user-perceived as so bad that the marketing g=
uys decided that offering no video/voice (and instead make people pick of t=
he phone) is better for product reputation than offer the degraded QoS.=C2=
=A0
<u></u><u></u></p>
<p class=3D"MsoNormal">Do any of you guys have hard numbers what sort of de=
lay h3-based transport can achieve on today=E2=80=99s infrastructure?</p></=
div></div></div></div></div></blockquote><div><br></div><div>With H3 we hav=
e lots of options; there are lots of details to work through here but given=
 that the substrate is QUIC there is no theoretical reason why RTP performa=
nce isn&#39;t possible.</div><div><br></div><div>Regarding HTTP media trans=
port in existing products (i.e., non-H3), you are correct that there are in=
deed limitations and as such providers do what they can to avoid this optio=
n, but it is nowhere near as grim as you make it out to be. Recall that the=
 first web version of Zoom exclusively=C2=A0<a href=3D"https://webrtchacks.=
com/zoom-avoids-using-webrtc/">used websockets for transport</a>.</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 lang=3D"EN-US"><div clas=
s=3D"gmail-m_-9050459078766760524WordSection1"><div><div><div>
</div>
</div>
</div>
</div>
</div>

</blockquote></div></div>

--000000000000d5ab4605a1b32cf9--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 25 Mar 2020 12:56:20 -0700
Message-ID: <CABcZeBNYFpV519O0--d8OpQQQq0epSrSOXi8TFEvWNmDrNzTcQ@mail.gmail.com>
To: Justin Uberti <juberti=40google.com@dmarc.ietf.org>
Cc: Stephan Wenger <stewe@stewe.org>, Victor Vasiliev <vasilvv@google.com>,  Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>,  Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
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Subject: Re: [V3] Thoughts on scope for ript
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On Wed, Mar 25, 2020 at 12:52 PM Justin Uberti <juberti=3D
40google.com@dmarc.ietf.org> wrote:

>
>
> On Wed, Mar 25, 2020 at 12:38 PM Stephan Wenger <stewe@stewe.org> wrote:
>
>> Hi Justin, all:
>>
>>
>>
>> *From: *V3 <v3-bounces@ietf.org> on behalf of Justin Uberti <juberti=3D
>> 40google.com@dmarc.ietf.org>
>> *Date: *Wednesday, March 25, 2020 at 12:32
>> *To: *Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
>> *Cc: *Victor Vasiliev <vasilvv@google.com>, Jonathan Rosenberg <
>> jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
>> *Subject: *Re: [V3] Thoughts on scope for ript
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 25, 2020 at 12:26 PM Spencer Dawkins at IETF <
>> spencerdawkins.ietf@gmail.com> wrote:
>>
>> Replying to Justin, after reading Victor's reply - thank you both for
>> replying ...
>>
>>
>>
>> On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti <juberti@google.com>
>> wrote:
>>
>>
>>
>>
>>
>> On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at IETF <
>> spencerdawkins.ietf@gmail.com> wrote:
>>
>> This is interesting ...
>>
>>
>>
>> On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=3D
>> 40google.com@dmarc.ietf.org> wrote:
>>
>> I would like to second Justin's point here about the value of a
>> client-server RTP streaming by itself (e-f, though in-band c-d are also
>> valuable).  There's a lot of value in the world where I can take an HTTP
>> URL to some endpoint in the cloud and give it to, say, an IoT camera dev=
ice
>> to use as a realtime media sink, or pass it to an off-the-shelf streamin=
g
>> library to show a realtime media source on screen.
>>
>>
>>
>> I know most of my own concerns about RIPT media have started out with
>> "this won't be as good as RTP for media", but what I've been getting fro=
m
>> RIPT discussions is, that may not matter as much as I expected.
>>
>>
>>
>> For a use case like Victor's, great media quality would be great, but
>> good enough media quality may be ... good enough.
>>
>>
>>
>> Am I putting words in the mouths of the proponents?
>>
>>
>>
>> Some applications may tolerate somewhat lower quality in fallback
>> scenarios, but I think we should be aiming for RTP performance with HTTP
>> ease of deployment.
>>
>>
>>
>> I was actually hoping that wasn't the case, because ISTM that the
>> question whether RTP performance over HTTP is possible/likely is going t=
o
>> suck up a LOT of the discussion time during the BOF ...
>>
>>
>>
>> To Eric's point, "H3 is not widely deployed yet, and so we should think
>> about how we want things to look". I think it's entirely possible to
>> achieve RTP performance over H3, and hopefully also have a reasonable
>> fallback when only H2 is available.
>>
>>
>>
>> [Stephan]: many video conferencing technologies (including Vidyo, Zoom),
>> have previously, or are currently supporting http-based media transport,
>> often for years or even decades.  They are almost never enabled by any
>> professional deployer, because the resulting QoS is user-perceived as so
>> bad that the marketing guys decided that offering no video/voice (and
>> instead make people pick of the phone) is better for product reputation
>> than offer the degraded QoS.
>>
>> Do any of you guys have hard numbers what sort of delay h3-based
>> transport can achieve on today=E2=80=99s infrastructure?
>>
>
> With H3 we have lots of options; there are lots of details to work throug=
h
> here but given that the substrate is QUIC there is no theoretical reason
> why RTP performance isn't possible.
>
> Regarding HTTP media transport in existing products (i.e., non-H3), you
> are correct that there are indeed limitations and as such providers do wh=
at
> they can to avoid this option, but it is nowhere near as grim as you make
> it out to be. Recall that the first web version of Zoom exclusively used
> websockets for transport
> <https://webrtchacks.com/zoom-avoids-using-webrtc/>.
>

I mostly concur with Justin here: I would expect QUIC (which is to say
probably H3 + Datagram) to be roughly comparable to RTP [0]. Anything over
TCP will obviously be non-awesome, but we already do have some nontrivial
fraction of media going over TCP to TURN relays, so it's not like it
doesn't work at all, especially in good network conditions.

-Ekr

[0] There are some open issues with how QUIC congestion control and the
media rate control interact, but I imagine they are mostly soluble.

> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>

--0000000000003e346b05a1b34503
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div dir=3D"ltr"></div><br><div class=3D"gmail_quote"><div=
 dir=3D"ltr" class=3D"gmail_attr">On Wed, Mar 25, 2020 at 12:52 PM Justin U=
berti &lt;juberti=3D<a href=3D"mailto:40google.com@dmarc.ietf.org">40google=
.com@dmarc.ietf.org</a>&gt; wrote:<br></div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204)=
;padding-left:1ex"><div dir=3D"ltr"><div dir=3D"ltr"><br></div><br><div cla=
ss=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Wed, Mar 25, 20=
20 at 12:38 PM Stephan Wenger &lt;<a href=3D"mailto:stewe@stewe.org" target=
=3D"_blank">stewe@stewe.org</a>&gt; wrote:<br></div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,=
204,204);padding-left:1ex">





<div lang=3D"EN-US">
<div>
<p class=3D"MsoNormal">Hi Justin, all:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div style=3D"border-color:rgb(181,196,223) currentcolor currentcolor;borde=
r-style:solid none none;border-width:1pt medium medium;padding:3pt 0in 0in"=
>
<p class=3D"MsoNormal"><b><span style=3D"font-size:12pt;color:black">From: =
</span></b><span style=3D"font-size:12pt;color:black">V3 &lt;<a href=3D"mai=
lto:v3-bounces@ietf.org" target=3D"_blank">v3-bounces@ietf.org</a>&gt; on b=
ehalf of Justin Uberti &lt;juberti=3D<a href=3D"mailto:40google.com@dmarc.i=
etf.org" target=3D"_blank">40google.com@dmarc.ietf.org</a>&gt;<br>
<b>Date: </b>Wednesday, March 25, 2020 at 12:32<br>
<b>To: </b>Spencer Dawkins at IETF &lt;<a href=3D"mailto:spencerdawkins.iet=
f@gmail.com" target=3D"_blank">spencerdawkins.ietf@gmail.com</a>&gt;<br>
<b>Cc: </b>Victor Vasiliev &lt;<a href=3D"mailto:vasilvv@google.com" target=
=3D"_blank">vasilvv@google.com</a>&gt;, Jonathan Rosenberg &lt;<a href=3D"m=
ailto:jdrosen@five9.com" target=3D"_blank">jdrosen@five9.com</a>&gt;, &quot=
;<a href=3D"mailto:v3@ietf.org" target=3D"_blank">v3@ietf.org</a>&quot; &lt=
;<a href=3D"mailto:v3@ietf.org" target=3D"_blank">v3@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [V3] Thoughts on scope for ript<u></u><u></u></span></p=
>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 12:26 PM Spencer Dawkins at =
IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" target=3D"_blank"=
>spencerdawkins.ietf@gmail.com</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<div>
<p class=3D"MsoNormal">Replying to Justin, after reading Victor&#39;s reply=
 - thank you both for replying ...<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti &lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at I=
ETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" target=3D"_blank">=
spencerdawkins.ietf@gmail.com</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<div>
<p class=3D"MsoNormal">This is interesting ...<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev &lt=
;vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_blank"=
>40google.com@dmarc.ietf.org</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<p class=3D"MsoNormal">I would like to second Justin&#39;s point here about=
 the value of a client-server RTP streaming by itself (e-f, though in-band =
c-d are also valuable).=C2=A0 There&#39;s a lot of value in the world where=
 I can take an HTTP URL to some endpoint in the
 cloud and give it to, say, an IoT camera device to use as a realtime media=
 sink, or pass it to an off-the-shelf streaming library to show a realtime =
media source on screen.<u></u><u></u></p>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I know most of my own concerns about RIPT media have=
 started out with &quot;this won&#39;t be as good as RTP for media&quot;, b=
ut what I&#39;ve been getting from RIPT discussions is, that may not matter=
 as much as I expected.=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">For a use case like Victor&#39;s, great media qualit=
y would be great, but good enough media quality may be ... good enough.=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">Am I putting words in the mouths of the proponents?<=
u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal">Some applications may tolerate somewhat lower qualit=
y in fallback scenarios, but I think we should be aiming for RTP performanc=
e with HTTP ease of deployment.<u></u><u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I was actually hoping that wasn&#39;t the case, beca=
use ISTM that the question whether RTP performance over HTTP is possible/li=
kely is going to suck up a LOT of the discussion time during the BOF ...=C2=
=A0=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To Eric&#39;s point, &quot;H3 is not widely deployed=
 yet, and so we should think about how we want things to look&quot;. I thin=
k it&#39;s entirely possible to achieve RTP performance over H3, and hopefu=
lly also have a reasonable fallback when only H2 is
 available.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">[Stephan]: many video conferencing technologies (inc=
luding Vidyo, Zoom), have previously, or are currently supporting http-base=
d media transport, often for years or even decades.=C2=A0 They are almost n=
ever enabled by any professional deployer,
 because the resulting QoS is user-perceived as so bad that the marketing g=
uys decided that offering no video/voice (and instead make people pick of t=
he phone) is better for product reputation than offer the degraded QoS.=C2=
=A0
<u></u><u></u></p>
<p class=3D"MsoNormal">Do any of you guys have hard numbers what sort of de=
lay h3-based transport can achieve on today=E2=80=99s infrastructure?</p></=
div></div></div></div></div></blockquote><div><br></div><div>With H3 we hav=
e lots of options; there are lots of details to work through here but given=
 that the substrate is QUIC there is no theoretical reason why RTP performa=
nce isn&#39;t possible.</div><div><br></div><div>Regarding HTTP media trans=
port in existing products (i.e., non-H3), you are correct that there are in=
deed limitations and as such providers do what they can to avoid this optio=
n, but it is nowhere near as grim as you make it out to be. Recall that the=
 first web version of Zoom exclusively=C2=A0<a href=3D"https://webrtchacks.=
com/zoom-avoids-using-webrtc/" target=3D"_blank">used websockets for transp=
ort</a>.</div></div></div></blockquote><div><br></div><div>I mostly concur =
with Justin here: I would expect QUIC (which is to say probably H3 + Datagr=
am) to be roughly comparable to RTP [0]. Anything over TCP will obviously b=
e non-awesome, but we already do have some nontrivial fraction of media goi=
ng over TCP to TURN relays, so it&#39;s not like it doesn&#39;t work at all=
, especially in good network conditions.<br></div><div><br></div><div>-Ekr<=
/div><div><br></div><div>[0] There are some open issues with how QUIC conge=
stion control and the media rate control interact, but I imagine they are m=
ostly soluble.<br></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 dir=3D"ltr"><div class=3D"gmail_quote"><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 lang=3D"EN-US"><div><div><div><div>
</div>
</div>
</div>
</div>
</div>

</blockquote></div></div>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Wed, 25 Mar 2020 13:18:24 -0700
Message-ID: <CAOJ7v-1s+i5400qFAsskSOwuafeYHckHrRkpbEe6WXrc9cTgkw@mail.gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
Cc: Stephan Wenger <stewe@stewe.org>, Victor Vasiliev <vasilvv@google.com>,  Jonathan Rosenberg <jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>,  Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
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Subject: Re: [V3] Thoughts on scope for ript
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On Wed, Mar 25, 2020 at 12:58 PM Eric Rescorla <ekr@rtfm.com> wrote:

>
> On Wed, Mar 25, 2020 at 12:52 PM Justin Uberti <juberti=3D
> 40google..com@dmarc.ietf.org <40google.com@dmarc.ietf.org>> wrote:
>
>>
>>
>> On Wed, Mar 25, 2020 at 12:38 PM Stephan Wenger <stewe@stewe.org> wrote:
>>
>>> Hi Justin, all:
>>>
>>>
>>>
>>> *From: *V3 <v3-bounces@ietf.org> on behalf of Justin Uberti <juberti=3D
>>> 40google.com@dmarc.ietf.org>
>>> *Date: *Wednesday, March 25, 2020 at 12:32
>>> *To: *Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
>>> *Cc: *Victor Vasiliev <vasilvv@google.com>, Jonathan Rosenberg <
>>> jdrosen@five9.com>, "v3@ietf.org" <v3@ietf.org>
>>> *Subject: *Re: [V3] Thoughts on scope for ript
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 25, 2020 at 12:26 PM Spencer Dawkins at IETF <
>>> spencerdawkins.ietf@gmail.com> wrote:
>>>
>>> Replying to Justin, after reading Victor's reply - thank you both for
>>> replying ...
>>>
>>>
>>>
>>> On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti <juberti@google.com>
>>> wrote:
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at IETF <
>>> spencerdawkins.ietf@gmail.com> wrote:
>>>
>>> This is interesting ...
>>>
>>>
>>>
>>> On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev <vasilvv=3D
>>> 40google.com@dmarc.ietf.org> wrote:
>>>
>>> I would like to second Justin's point here about the value of a
>>> client-server RTP streaming by itself (e-f, though in-band c-d are also
>>> valuable).  There's a lot of value in the world where I can take an HTT=
P
>>> URL to some endpoint in the cloud and give it to, say, an IoT camera de=
vice
>>> to use as a realtime media sink, or pass it to an off-the-shelf streami=
ng
>>> library to show a realtime media source on screen.
>>>
>>>
>>>
>>> I know most of my own concerns about RIPT media have started out with
>>> "this won't be as good as RTP for media", but what I've been getting fr=
om
>>> RIPT discussions is, that may not matter as much as I expected.
>>>
>>>
>>>
>>> For a use case like Victor's, great media quality would be great, but
>>> good enough media quality may be ... good enough.
>>>
>>>
>>>
>>> Am I putting words in the mouths of the proponents?
>>>
>>>
>>>
>>> Some applications may tolerate somewhat lower quality in fallback
>>> scenarios, but I think we should be aiming for RTP performance with HTT=
P
>>> ease of deployment.
>>>
>>>
>>>
>>> I was actually hoping that wasn't the case, because ISTM that the
>>> question whether RTP performance over HTTP is possible/likely is going =
to
>>> suck up a LOT of the discussion time during the BOF ...
>>>
>>>
>>>
>>> To Eric's point, "H3 is not widely deployed yet, and so we should think
>>> about how we want things to look". I think it's entirely possible to
>>> achieve RTP performance over H3, and hopefully also have a reasonable
>>> fallback when only H2 is available.
>>>
>>>
>>>
>>> [Stephan]: many video conferencing technologies (including Vidyo, Zoom)=
,
>>> have previously, or are currently supporting http-based media transport=
,
>>> often for years or even decades.  They are almost never enabled by any
>>> professional deployer, because the resulting QoS is user-perceived as s=
o
>>> bad that the marketing guys decided that offering no video/voice (and
>>> instead make people pick of the phone) is better for product reputation
>>> than offer the degraded QoS.
>>>
>>> Do any of you guys have hard numbers what sort of delay h3-based
>>> transport can achieve on today=E2=80=99s infrastructure?
>>>
>>
>> With H3 we have lots of options; there are lots of details to work
>> through here but given that the substrate is QUIC there is no theoretica=
l
>> reason why RTP performance isn't possible.
>>
>> Regarding HTTP media transport in existing products (i.e., non-H3), you
>> are correct that there are indeed limitations and as such providers do w=
hat
>> they can to avoid this option, but it is nowhere near as grim as you mak=
e
>> it out to be. Recall that the first web version of Zoom exclusively used
>> websockets for transport
>> <https://webrtchacks.com/zoom-avoids-using-webrtc/>.
>>
>
> I mostly concur with Justin here: I would expect QUIC (which is to say
> probably H3 + Datagram) to be roughly comparable to RTP [0]. Anything ove=
r
> TCP will obviously be non-awesome, but we already do have some nontrivial
> fraction of media going over TCP to TURN relays, so it's not like it
> doesn't work at all, especially in good network conditions.
>
> -Ekr
>
> [0] There are some open issues with how QUIC congestion control and the
> media rate control interact, but I imagine they are mostly soluble.
>

Yes, the CC stuff is one of the details I was thinking of. The other main
one is the right abstractions in cloud providers to allow more direct
access to the underlying transport.

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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 Wed, Mar 25, 2020 at 12:58 PM Eric=
 Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<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 dir=3D"ltr">=
<div dir=3D"ltr"></div><br><div class=3D"gmail_quote"><div dir=3D"ltr" clas=
s=3D"gmail_attr">On Wed, Mar 25, 2020 at 12:52 PM Justin Uberti &lt;juberti=
=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_blank">40googl=
e..com@dmarc.ietf.org</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"><div dir=3D"ltr"><div dir=3D"ltr"><br></div><br><div c=
lass=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Wed, Mar 25, =
2020 at 12:38 PM Stephan Wenger &lt;<a href=3D"mailto:stewe@stewe.org" targ=
et=3D"_blank">stewe@stewe.org</a>&gt; wrote:<br></div><blockquote class=3D"=
gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(20=
4,204,204);padding-left:1ex">





<div lang=3D"EN-US">
<div>
<p class=3D"MsoNormal">Hi Justin, all:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div style=3D"border-color:rgb(181,196,223) currentcolor currentcolor;borde=
r-style:solid none none;border-width:1pt medium medium;padding:3pt 0in 0in"=
>
<p class=3D"MsoNormal"><b><span style=3D"font-size:12pt;color:black">From: =
</span></b><span style=3D"font-size:12pt;color:black">V3 &lt;<a href=3D"mai=
lto:v3-bounces@ietf.org" target=3D"_blank">v3-bounces@ietf.org</a>&gt; on b=
ehalf of Justin Uberti &lt;juberti=3D<a href=3D"mailto:40google.com@dmarc.i=
etf.org" target=3D"_blank">40google.com@dmarc.ietf.org</a>&gt;<br>
<b>Date: </b>Wednesday, March 25, 2020 at 12:32<br>
<b>To: </b>Spencer Dawkins at IETF &lt;<a href=3D"mailto:spencerdawkins.iet=
f@gmail.com" target=3D"_blank">spencerdawkins.ietf@gmail.com</a>&gt;<br>
<b>Cc: </b>Victor Vasiliev &lt;<a href=3D"mailto:vasilvv@google.com" target=
=3D"_blank">vasilvv@google.com</a>&gt;, Jonathan Rosenberg &lt;<a href=3D"m=
ailto:jdrosen@five9.com" target=3D"_blank">jdrosen@five9.com</a>&gt;, &quot=
;<a href=3D"mailto:v3@ietf.org" target=3D"_blank">v3@ietf.org</a>&quot; &lt=
;<a href=3D"mailto:v3@ietf.org" target=3D"_blank">v3@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [V3] Thoughts on scope for ript<u></u><u></u></span></p=
>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 12:26 PM Spencer Dawkins at =
IETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" target=3D"_blank"=
>spencerdawkins.ietf@gmail.com</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<div>
<p class=3D"MsoNormal">Replying to Justin, after reading Victor&#39;s reply=
 - thank you both for replying ...<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 11:55 AM Justin Uberti &lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 9:28 AM Spencer Dawkins at I=
ETF &lt;<a href=3D"mailto:spencerdawkins.ietf@gmail.com" target=3D"_blank">=
spencerdawkins.ietf@gmail.com</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<div>
<p class=3D"MsoNormal">This is interesting ...<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Mar 25, 2020 at 10:31 AM Victor Vasiliev &lt=
;vasilvv=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_blank"=
>40google.com@dmarc.ietf.org</a>&gt; wrote:<u></u><u></u></p>
</div>
<blockquote>
<div>
<p class=3D"MsoNormal">I would like to second Justin&#39;s point here about=
 the value of a client-server RTP streaming by itself (e-f, though in-band =
c-d are also valuable).=C2=A0 There&#39;s a lot of value in the world where=
 I can take an HTTP URL to some endpoint in the
 cloud and give it to, say, an IoT camera device to use as a realtime media=
 sink, or pass it to an off-the-shelf streaming library to show a realtime =
media source on screen.<u></u><u></u></p>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I know most of my own concerns about RIPT media have=
 started out with &quot;this won&#39;t be as good as RTP for media&quot;, b=
ut what I&#39;ve been getting from RIPT discussions is, that may not matter=
 as much as I expected.=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">For a use case like Victor&#39;s, great media qualit=
y would be great, but good enough media quality may be ... good enough.=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">Am I putting words in the mouths of the proponents?<=
u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal">Some applications may tolerate somewhat lower qualit=
y in fallback scenarios, but I think we should be aiming for RTP performanc=
e with HTTP ease of deployment.<u></u><u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I was actually hoping that wasn&#39;t the case, beca=
use ISTM that the question whether RTP performance over HTTP is possible/li=
kely is going to suck up a LOT of the discussion time during the BOF ...=C2=
=A0=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To Eric&#39;s point, &quot;H3 is not widely deployed=
 yet, and so we should think about how we want things to look&quot;. I thin=
k it&#39;s entirely possible to achieve RTP performance over H3, and hopefu=
lly also have a reasonable fallback when only H2 is
 available.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">[Stephan]: many video conferencing technologies (inc=
luding Vidyo, Zoom), have previously, or are currently supporting http-base=
d media transport, often for years or even decades.=C2=A0 They are almost n=
ever enabled by any professional deployer,
 because the resulting QoS is user-perceived as so bad that the marketing g=
uys decided that offering no video/voice (and instead make people pick of t=
he phone) is better for product reputation than offer the degraded QoS.=C2=
=A0
<u></u><u></u></p>
<p class=3D"MsoNormal">Do any of you guys have hard numbers what sort of de=
lay h3-based transport can achieve on today=E2=80=99s infrastructure?</p></=
div></div></div></div></div></blockquote><div><br></div><div>With H3 we hav=
e lots of options; there are lots of details to work through here but given=
 that the substrate is QUIC there is no theoretical reason why RTP performa=
nce isn&#39;t possible.</div><div><br></div><div>Regarding HTTP media trans=
port in existing products (i.e., non-H3), you are correct that there are in=
deed limitations and as such providers do what they can to avoid this optio=
n, but it is nowhere near as grim as you make it out to be. Recall that the=
 first web version of Zoom exclusively=C2=A0<a href=3D"https://webrtchacks.=
com/zoom-avoids-using-webrtc/" target=3D"_blank">used websockets for transp=
ort</a>.</div></div></div></blockquote><div><br></div><div>I mostly concur =
with Justin here: I would expect QUIC (which is to say probably H3 + Datagr=
am) to be roughly comparable to RTP [0]. Anything over TCP will obviously b=
e non-awesome, but we already do have some nontrivial fraction of media goi=
ng over TCP to TURN relays, so it&#39;s not like it doesn&#39;t work at all=
, especially in good network conditions.<br></div><div><br></div><div>-Ekr<=
/div><div><br></div><div>[0] There are some open issues with how QUIC conge=
stion control and the media rate control interact, but I imagine they are m=
ostly soluble.</div></div></div></blockquote><div><br></div><div>Yes, the C=
C stuff is one of the details I was thinking of. The other main one is the =
right abstractions in cloud providers to allow more direct access to the un=
derlying transport.=C2=A0</div></div></div>

--000000000000bdb59e05a1b39232--


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "v3@ietf.org" <v3@ietf.org>
CC: Martin Thomson <mt@lowentropy.net>
Thread-Topic: [V3] Virtual IETF 107 Meeting WebEx Invitation: RIPT
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Subject: Re: [V3] Virtual IETF 107 Meeting WebEx Invitation: RIPT
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The RIPT BOF will begin in 2 hours at the Webex link below.
https://ietf.webex.com/ietf/j.php?MTID=3Dm77168c5c7592f30d7f369b49b47cfcf3


Please sign the virtual blue sheet in Etherpad:
https://etherpad.ietf.org/p/notes-ietf-107-ript?useMonospaceFont=3Dtrue

Please join Jabber for general chat: xmpp:ript@jabber.ietf.org?join
Webex chat is only to join the mic queue ("+q", or "-q" to cancel).

We need volunteer Note Takers and a Jabber Scribe; please unicast the
chairs.

Thanks,
Mo & Martin

On 3/20/20, 1:29 PM, "V3 on behalf of IETF Agenda" <v3-bounces@ietf.org on
behalf of agenda@ietf.org> wrote:

>RIPT Session at IETF 107
>
>Thursday, March 26, 2020
>22:10-00:10 UTC
>
>Join meeting:=20
><https://ietf.webex.com/ietf/j.php?MTID=3Dm77168c5c7592f30d7f369b49b47cfcf=
3>
>
>Meeting number (access code): 646 916 587
>Meeting password: avEzuWXp427
>
>IETF 107 Agenda: <https://datatracker.ietf.org/meeting/107/agenda>
>
>--=20
>V3 mailing list
>V3@ietf.org
>https://www.ietf.org/mailman/listinfo/v3


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Date: Fri, 27 Mar 2020 12:03:25 +1100
From: "Martin Thomson" <mt@lowentropy.net>
To: v3@ietf.org
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Subject: [V3] Minutes from BoF session
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https://datatracker.ietf.org/doc/minutes-107-ript/ contains both the minutes from Dan York that were collected in Etherpad and some bonus minutes prepared by Jean Mahoney.

Thank you both very much for your contribution and the excellent notes.  And thank you everyone who contributed to a productive discussion.

We have some more work to do on refining the scope of this work.  I encourage those people who had specific issues with scope to again raise them on the list so that we can work through them.

Related to the scoping question, it seems to me that the question of how to structure any effort in this space is going to be quite important.  If people have concrete suggestions about how that might be done, then input on that would also be valuable.


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
To: "v3@ietf.org" <v3@ietf.org>
Thread-Topic: High level comments on the RIPT charter and scope
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Subject: [V3] High level comments on the RIPT charter and scope
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Hi,

So this is my personal opinion and based on my long work with RTP.=20

Yes, there are clearly interest in chartering some work in IETF. Howver, I =
think
there is significant diversity in what people are interested in actually
happening.=20

Clearly there is the signalling protocol for seession/call establishment an=
d
control between peering provider nodes. Lets call this the outbound part. T=
he
main issue I see in this area is the quetion of feature creap. It would be =
good
to figure out what core functionality is really needed and actually get tha=
t
described in a charter. A Question is if conference applications impacts th=
is
scope or not.=20

The next part is the inbound problem, i.e. having end-user devices/applicat=
ion
being able to register to their provider as being ready to receive/answer c=
alls.
This may actually be sufficient similar to the outbound part that they can =
be
done in the same WG.=20

However, an interesting question for both of the above is if the solution i=
s
required to be able to be implemented in a browser, at least the client par=
t. I
think the signalling is fairly straight forward to be able to implement in =
JS as
long as we ignore the media.=20

The third big part is the media transport. I think we need to strongly cons=
ider
if this can be broken out in to it seperate WG. There are several aspects h=
ere.
First of the split in expertise between the media transport and the signall=
ing
folks. The next aspect is the resuability of the media transport, that anyw=
ay
need to be define as individual reusable protocol component that is not
dependent on the signalling. I don't see a huge problem with being able to
initiate a media transport channel into the relevant server node using an
initial HTTP request. I am thinking WebSocket like establishment of the med=
ia
transport channel here. And possibly there will be need for a fallback to H=
TTP
only, but also that should be further discussed.=20

I am also very worried that if the model for the media transport is only po=
int
to point calls then we get something that will become obsolete or at least =
need
hacks immediately. This thing need to support centralized conferences with
cascaded or meshed central media nodes. So I think it is cruical that one d=
efine
the usage model with switched central nodes and have that in mind so that o=
ne
can actually support the applications needed. I think by defining a strict =
model
one can actually avoid a lot of feature creep in the media transport.=20

We also have use cases that has practical total overlapp, namely real-time
sensor streaming services, where cameras, and other frequently sampling sen=
sors
send data of real-time control loop nature. There is also the media industr=
ies
need for content contribution, i.e. how to get camera's media into the
production centers. These are similar enough and need of standardized proto=
cols.
SRT in Dispatch is only one aspect of this. What we don't need to take on f=
or
this is the one-to-many distribution only. So broadcast and multicast can b=
e
declared out of scope.=20

I think E2E media secruity need to be defined from day one. It will be a ce=
ntral
part of these systems even if in the legacy interconnect there will be the
termination in gateways. Also media processing nodes can be included but th=
e
inclusion of them in the trust will need to be explicit in this model.=20

People have been voicing that we don't need things to work as well as RTP. =
I
think in fact that a dedicated real-time media over QUIC (with datagram) ca=
n
actually achieve better than RTP performance by using a mix of reliable and
semi-reliable transport primitives on applciation data unit level, i.e. aud=
io
and video frames.=20

The next media transport issue that was barely discussed in the BOF is the
legacy (RTP) interoperate question. This is going to come back in the scope
discussion and do needs at least a direction.=20

Coming back to my initial statement about diversity, I think there at least=
 two
communities that are interested in this. And I think we actually need to at=
tempt
to make relevant protocol work for both these communities. For the moment t=
he
most outspoken people in this effort have been the signalling people. The m=
edia
transport crowd have been waiting for their turn until QUIC to matured
sufficiently to be a realistic start. But as people know some discussion ha=
ve
happened. However we are this point now I think where work can start in ern=
est
also on this topic.=20

So can we seriously consider writing two different charter that together en=
able
the intended use cases?=20

Cheers

Magnus Westerlund=20


----------------------------------------------------------------------
Networks, Ericsson Research
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Torshamnsgatan 23           | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From: Cullen Jennings <fluffy@iii.ca>
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To: Magnus Westerlund <magnus.westerlund=40ericsson.com@dmarc.ietf.org>
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Subject: [V3] Centralized conference media
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> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund =
<magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
>=20
> I am also very worried that if the model for the media transport is =
only point
> to point calls then we get something that will become obsolete or at =
least need
> hacks immediately. This thing need to support centralized conferences =
with
> cascaded or meshed central media nodes. So I think it is cruical that =
one define
> the usage model with switched central nodes and have that in mind so =
that one
> can actually support the applications needed. I think by defining a =
strict model
> one can actually avoid a lot of feature creep in the media transport.=20=



I think it does make sense to support some meta data on a media flow for =
centralized conferencing support. What semantic level information do you =
think is needed in the media ?




--Apple-Mail=_1269667F-8794-4909-987C-CF24C4464EE5
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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""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On Mar 27, 2020, at 8:23 AM, Magnus Westerlund &lt;<a =
href=3D"mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org" =
class=3D"">magnus.westerlund=3D40ericsson.com@dmarc.ietf.org</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: =
14px; 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"">I am also very worried that if =
the model for the media transport is only point</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">to point calls then we get =
something that will become obsolete or at least need</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">hacks immediately. This thing =
need to support centralized conferences with</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">cascaded or meshed central media =
nodes. So I think it is cruical that one define</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">the usage model with switched =
central nodes and have that in mind so that one</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">can actually support the =
applications needed. I think by defining a strict model</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">one can actually avoid a lot of =
feature creep in the media transport.<span =
class=3D"Apple-converted-space">&nbsp;</span></span></div></blockquote></d=
iv><div class=3D""><br class=3D""></div><div class=3D"">I think it does =
make sense to support some meta data on a media flow for centralized =
conferencing support. What semantic level information do you think is =
needed in the media ?</div><div class=3D""><br class=3D""></div><div =
class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""></div></body></html>=

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> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund =
<magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
>=20
> The third big part is the media transport. I think we need to strongly =
consider
> if this can be broken out in to it seperate WG.

I think the WG we are talking about here is the WG to do media of H3. So =
we can talk about other things being in or out of it but that is the =
core thing.=20

I would like to point out that one of the design properties that has =
been making this easier is that media is self describing. What I mean by =
that is that if a device receives media that is one of the supported =
formats and within in the limits of that device supports, the media has =
all the information the device needs to decode it. No out of band =
information needed. That is key to making this easier to deal with HA =
and scalability on cloud side. I think that is a good property for =
modern designs. We have been going that way for a long time with things =
like opus and it works out well.=20

> here are several aspects here.
> First of the split in expertise between the media transport and the =
signalling
> folks.=20

I=E2=80=99m not sure what you mean by signalling but the the singling =
here was mostly start media , stop media. It is not call routing, =
forking etc that is done by SIP.=20




--Apple-Mail=_21EC913E-35C6-4463-9068-08200770284A
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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 Mar 27, 2020, at 8:23 AM, Magnus Westerlund &lt;<a =
href=3D"mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org" =
class=3D"">magnus.westerlund=3D40ericsson.com@dmarc.ietf.org</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: =
14px; 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"">The third big part is the media =
transport. I think we need to strongly consider</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">if this can be broken out in to =
it seperate WG. </span></div></blockquote><div><br class=3D""></div><div>I=
 think the WG we are talking about here is the WG to do media of H3. So =
we can talk about other things being in or out of it but that is the =
core thing.&nbsp;</div><div><br class=3D""></div><div>I would like to =
point out that one of the design properties that has been making this =
easier is that media is self describing. What I mean by that is that if =
a device receives media that is one of the supported formats and within =
in the limits of that device supports, the media has all the information =
the device needs to decode it. No out of band information needed. That =
is key to making this easier to deal with HA and scalability on cloud =
side. I think that is a good property for modern designs. We have been =
going that way for a long time with things like opus and it works out =
well.&nbsp;</div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><span style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 14px; 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"">here are =
several aspects here.</span><br style=3D"caret-color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 14px; 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""><span style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 14px; 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"">First of the split in expertise between the media transport =
and the signalling</span><br style=3D"caret-color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 14px; 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""><span style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 14px; 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"">folks.<span =
class=3D"Apple-converted-space">&nbsp;</span></span></div></blockquote><br=
 class=3D""></div><div>I=E2=80=99m not sure what you mean by signalling =
but the the singling here was mostly start media , stop media. It is not =
call routing, forking etc that is done by SIP.&nbsp;</div><div><br =
class=3D""></div><div><br class=3D""></div><br class=3D""></body></html>=

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From: Cullen Jennings <fluffy@iii.ca>
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To: Magnus Westerlund <magnus.westerlund=40ericsson.com@dmarc.ietf.org>
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> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund =
<magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
>=20
> The next media transport issue that was barely discussed in the BOF is =
the
> legacy (RTP) interoperate question. This is going to come back in the =
scope
> discussion and do needs at least a direction.=20

In WebRTC, we decided to be able to gateway SIP to WebRTC without =
processing media. That turned out to be a major limitation to how WebRTC =
worked. That is good and we have WebRTC but for this, I am storngly of =
the view we should not have that design constraint.=20

I think it is fine to require a media gateway to go from RIPT to RTP. =
Sure that should be pretty easy - one of the major use cases for this =
work will likely involve building SBCs that talk RIPT on one side and =
SIP on the other. Pretty sure that a bunch of other people are thinking =
about the same.=20

That sound right to you are or are you arguing for something else ?



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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""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On Mar 27, 2020, at 8:23 AM, Magnus Westerlund &lt;<a =
href=3D"mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org" =
class=3D"">magnus.westerlund=3D40ericsson.com@dmarc.ietf.org</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: =
14px; 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"">The next media transport issue =
that was barely discussed in the BOF is the</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
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-webkit-text-stroke-width: 0px; text-decoration: none;" class=3D""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">legacy (RTP) interoperate =
question. This is going to come back in the scope</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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"">discussion and do needs at least =
a direction.<span =
class=3D"Apple-converted-space">&nbsp;</span></span></div></blockquote></d=
iv><br class=3D""><div class=3D"">In WebRTC, we decided to be able to =
gateway SIP to WebRTC without processing media. That turned out to be a =
major limitation to how WebRTC worked. That is good and we have WebRTC =
but for this, I am storngly of the view we should not have that design =
constraint.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">I think it is fine to require a media gateway to go from RIPT =
to RTP. Sure that should be pretty easy - one of the major use cases for =
this work will likely involve building SBCs that talk RIPT on one side =
and SIP on the other. Pretty sure that a bunch of other people are =
thinking about the same.&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D"">That sound right to you are or are you =
arguing for something else ?</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div></body></html>=

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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Mar 2020 09:28:15 -0700
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To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [V3] Centralized conference media
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I think Magnus is trying to establish the conference topology, which has
ramifications on how we want to handle things like RTCP.

One strawman could be to focus on RTCP-terminating SFUs, i.e., cementing
the notion that RIPT media flows are indeed point-to-point (i.e.,
client-to-SFU, as opposed to client-to-client1, client2, client3).

On Fri, Mar 27, 2020 at 9:11 AM Cullen Jennings <fluffy@iii.ca> wrote:

>
>
> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
> magnus.westerlund=40ericsson.com@dmarc.ietf.org> wrote:
>
> I am also very worried that if the model for the media transport is only
> point
> to point calls then we get something that will become obsolete or at least
> need
> hacks immediately. This thing need to support centralized conferences with
> cascaded or meshed central media nodes. So I think it is cruical that one
> define
> the usage model with switched central nodes and have that in mind so that
> one
> can actually support the applications needed. I think by defining a strict
> model
> one can actually avoid a lot of feature creep in the media transport.
>
>
> I think it does make sense to support some meta data on a media flow for
> centralized conferencing support. What semantic level information do you
> think is needed in the media ?
>
>
>
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>

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<div dir=3D"ltr">I think Magnus is trying to establish the conference topol=
ogy, which has ramifications on how we want to handle things like RTCP.<div=
><br></div><div>One strawman could be to focus on RTCP-terminating SFUs, i.=
e., cementing the notion that RIPT media flows are indeed point-to-point (i=
.e., client-to-SFU, as opposed to client-to-client1, client2, client3).</di=
v></div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr=
">On Fri, Mar 27, 2020 at 9:11 AM Cullen Jennings &lt;<a href=3D"mailto:flu=
ffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<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 style=3D"overflow-wrap: break-word;"><br><div=
><br><blockquote type=3D"cite"><div>On Mar 27, 2020, at 8:23 AM, Magnus Wes=
terlund &lt;<a href=3D"mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf=
.org" target=3D"_blank">magnus.westerlund=3D40ericsson.com@dmarc.ietf.org</=
a>&gt; wrote:</div><br><div><span style=3D"font-family:Helvetica;font-size:=
14px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-s=
pacing:normal;text-align:start;text-indent:0px;text-transform:none;white-sp=
ace:normal;word-spacing:0px;text-decoration:none;float:none;display:inline"=
>I am also very worried that if the model for the media transport is only p=
oint</span><br style=3D"font-family:Helvetica;font-size:14px;font-style:nor=
mal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-=
align:start;text-indent:0px;text-transform:none;white-space:normal;word-spa=
cing:0px;text-decoration:none"><span style=3D"font-family:Helvetica;font-si=
ze:14px;font-style:normal;font-variant-caps:normal;font-weight:normal;lette=
r-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white=
-space:normal;word-spacing:0px;text-decoration:none;float:none;display:inli=
ne">to point calls then we get something that will become obsolete or at le=
ast need</span><br style=3D"font-family:Helvetica;font-size:14px;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"><span style=3D"font-family:Helvetica;fon=
t-size:14px;font-style:normal;font-variant-caps:normal;font-weight:normal;l=
etter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;w=
hite-space:normal;word-spacing:0px;text-decoration:none;float:none;display:=
inline">hacks immediately. This thing need to support centralized conferenc=
es with</span><br style=3D"font-family:Helvetica;font-size:14px;font-style:=
normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;te=
xt-align:start;text-indent:0px;text-transform:none;white-space:normal;word-=
spacing:0px;text-decoration:none"><span style=3D"font-family:Helvetica;font=
-size:14px;font-style:normal;font-variant-caps:normal;font-weight:normal;le=
tter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;wh=
ite-space:normal;word-spacing:0px;text-decoration:none;float:none;display:i=
nline">cascaded or meshed central media nodes. So I think it is cruical tha=
t one define</span><br style=3D"font-family:Helvetica;font-size:14px;font-s=
tyle:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:norm=
al;text-align:start;text-indent:0px;text-transform:none;white-space:normal;=
word-spacing:0px;text-decoration:none"><span style=3D"font-family:Helvetica=
;font-size:14px;font-style:normal;font-variant-caps:normal;font-weight:norm=
al;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:no=
ne;white-space:normal;word-spacing:0px;text-decoration:none;float:none;disp=
lay:inline">the usage model with switched central nodes and have that in mi=
nd so that one</span><br style=3D"font-family:Helvetica;font-size:14px;font=
-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:no=
rmal;text-align:start;text-indent:0px;text-transform:none;white-space:norma=
l;word-spacing:0px;text-decoration:none"><span style=3D"font-family:Helveti=
ca;font-size:14px;font-style:normal;font-variant-caps:normal;font-weight:no=
rmal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:=
none;white-space:normal;word-spacing:0px;text-decoration:none;float:none;di=
splay:inline">can actually support the applications needed. I think by defi=
ning a strict model</span><br style=3D"font-family:Helvetica;font-size:14px=
;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spaci=
ng:normal;text-align:start;text-indent:0px;text-transform:none;white-space:=
normal;word-spacing:0px;text-decoration:none"><span style=3D"font-family:He=
lvetica;font-size:14px;font-style:normal;font-variant-caps:normal;font-weig=
ht:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-trans=
form:none;white-space:normal;word-spacing:0px;text-decoration:none;float:no=
ne;display:inline">one can actually avoid a lot of feature creep in the med=
ia transport.<span>=C2=A0</span></span></div></blockquote></div><div><br></=
div><div>I think it does make sense to support some meta data on a media fl=
ow for centralized conferencing support. What semantic level information do=
 you think is needed in the media ?</div><div><br></div><div><br></div><div=
><br></div></div>-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>

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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Mar 2020 13:22:07 -0700
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To: Cullen Jennings <fluffy@iii.ca>
Cc: Magnus Westerlund <magnus.westerlund=40ericsson.com@dmarc.ietf.org>,  "v3@ietf.org" <v3@ietf.org>
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It's hard to imagine that we can avoid this if we are in fact going to use
H3 as a substrate.

I do think it would be useful to ensure the RIPT->RTP gateway can be
stateless and transparent (i.e., you can transform a RIPT frame into a RTP
packet with reasonable fidelity), so that said gateway is all you need in
order for existing services to participate in the RIPT ecosystem.

On Fri, Mar 27, 2020 at 9:32 AM Cullen Jennings <fluffy@iii.ca> wrote:

>
>
> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
> magnus.westerlund=40ericsson.com@dmarc.ietf.org> wrote:
>
> The next media transport issue that was barely discussed in the BOF is the
> legacy (RTP) interoperate question. This is going to come back in the scope
> discussion and do needs at least a direction.
>
>
> In WebRTC, we decided to be able to gateway SIP to WebRTC without
> processing media. That turned out to be a major limitation to how WebRTC
> worked. That is good and we have WebRTC but for this, I am storngly of the
> view we should not have that design constraint.
>
> I think it is fine to require a media gateway to go from RIPT to RTP. Sure
> that should be pretty easy - one of the major use cases for this work will
> likely involve building SBCs that talk RIPT on one side and SIP on the
> other. Pretty sure that a bunch of other people are thinking about the
> same.
>
> That sound right to you are or are you arguing for something else ?
>
>
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>

--000000000000b1f1b605a1dbdb75
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<div dir=3D"ltr">It&#39;s hard to imagine that we can avoid this if we are =
in fact going to use H3 as a substrate.=C2=A0<div><br></div><div>I do think=
 it would be useful to ensure the RIPT-&gt;RTP gateway can be stateless and=
 transparent (i.e., you can transform a RIPT frame into a RTP packet with r=
easonable fidelity), so that said gateway is all you need in order for exis=
ting services to participate=C2=A0in the RIPT ecosystem.</div></div><br><di=
v class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Fri, Mar 2=
7, 2020 at 9:32 AM Cullen Jennings &lt;<a href=3D"mailto:fluffy@iii.ca">flu=
ffy@iii.ca</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 style=3D"overflow-wrap: break-word;"><br><div><br><blockquo=
te type=3D"cite"><div>On Mar 27, 2020, at 8:23 AM, Magnus Westerlund &lt;<a=
 href=3D"mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org" target=
=3D"_blank">magnus.westerlund=3D40ericsson.com@dmarc.ietf.org</a>&gt; wrote=
:</div><br><div><span style=3D"font-family:Helvetica;font-size:14px;font-st=
yle:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:norma=
l;text-align:start;text-indent:0px;text-transform:none;white-space:normal;w=
ord-spacing:0px;text-decoration:none;float:none;display:inline">The next me=
dia transport issue that was barely discussed in the BOF is the</span><br s=
tyle=3D"font-family:Helvetica;font-size:14px;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-de=
coration:none"><span style=3D"font-family:Helvetica;font-size:14px;font-sty=
le:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal=
;text-align:start;text-indent:0px;text-transform:none;white-space:normal;wo=
rd-spacing:0px;text-decoration:none;float:none;display:inline">legacy (RTP)=
 interoperate question. This is going to come back in the scope</span><br s=
tyle=3D"font-family:Helvetica;font-size:14px;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-de=
coration:none"><span style=3D"font-family:Helvetica;font-size:14px;font-sty=
le:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal=
;text-align:start;text-indent:0px;text-transform:none;white-space:normal;wo=
rd-spacing:0px;text-decoration:none;float:none;display:inline">discussion a=
nd do needs at least a direction.<span>=C2=A0</span></span></div></blockquo=
te></div><br><div>In WebRTC, we decided to be able to gateway SIP to WebRTC=
 without processing media. That turned out to be a major limitation to how =
WebRTC worked. That is good and we have WebRTC but for this, I am storngly =
of the view we should not have that design constraint.=C2=A0</div><div><br>=
</div><div>I think it is fine to require a media gateway to go from RIPT to=
 RTP. Sure that should be pretty easy - one of the major use cases for this=
 work will likely involve building SBCs that talk RIPT on one side and SIP =
on the other. Pretty sure that a bunch of other people are thinking about t=
he same.=C2=A0</div><div><br></div><div>That sound right to you are or are =
you arguing for something else ?</div><div><br></div><div><br></div></div>-=
- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>

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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
To: "juberti=40google.com@dmarc.ietf.org" <juberti=40google.com@dmarc.ietf.org>, "fluffy@iii.ca" <fluffy@iii.ca>
CC: "v3@ietf.org" <v3@ietf.org>
Thread-Topic: [V3] Centralized conference media
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Archived-At: <https://mailarchive.ietf.org/arch/msg/v3/-gapJj4bcHLUhQCLp2MlXc5DXHI>
Subject: Re: [V3] Centralized conference media
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Hi,

So yes, I think there are the aspect of the topology as well as how one han=
dle
the media related controll messages, i.e. the equivalent of RTCP. For examp=
le
things like FIR, PAUSE/Resume.=20

So, if one with RTCP terminating middleboxes means that the ID spaces are s=
oley
locally that could be a choice. I will note that in some cases one still ne=
ed to
have the complete set of media sources from all end-points in a conference =
be
identified at a specific point to point leg. Even if only a subset is curre=
ntly
transmitted over this leg. There are both benefits and downsides I think of
using a local namespace and having each node translate these.

I will note that one may still need to have end-to-end identies on media AD=
Us if
one are going to have any end-to-end security.=20

Cheers

Magnus Westerlund



On Fri, 2020-03-27 at 09:28 -0700, Justin Uberti wrote:
> I think Magnus is trying to establish the conference topology, which has
> ramifications on how we want to handle things like RTCP.
>=20
> One strawman could be to focus on RTCP-terminating SFUs, i.e., cementing =
the
> notion that RIPT media flows are indeed point-to-point (i..e., client-to-=
SFU,
> as opposed to client-to-client1, client2, client3).
>=20
> On Fri, Mar 27, 2020 at 9:11 AM Cullen Jennings <fluffy@iii.ca> wrote:
> >=20
> > > On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
> > > magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
> > >=20
> > > I am also very worried that if the model for the media transport is o=
nly
> > > point
> > > to point calls then we get something that will become obsolete or at =
least
> > > need
> > > hacks immediately. This thing need to support centralized conferences=
 with
> > > cascaded or meshed central media nodes. So I think it is cruical that=
 one
> > > define
> > > the usage model with switched central nodes and have that in mind so =
that
> > > one
> > > can actually support the applications needed. I think by defining a s=
trict
> > > model
> > > one can actually avoid a lot of feature creep in the media transport.=
=20
> >=20
> > I think it does make sense to support some meta data on a media flow fo=
r
> > centralized conferencing support. What semantic level information do yo=
u
> > think is needed in the media ?
> >=20
> >=20
> >=20
> > --=20
> > V3 mailing list
> > V3@ietf.org
> > https://www.ietf.org/mailman/listinfo/v3
--=20
Cheers

Magnus Westerlund=20


----------------------------------------------------------------------
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----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Torshamnsgatan 23           | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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Cc: Cullen Jennings <fluffy@iii.ca>, Magnus Westerlund <magnus.westerlund=40ericsson.com@dmarc.ietf.org>, "v3@ietf.org" <v3@ietf.org>
To: Justin Uberti <juberti=40google.com@dmarc.ietf.org>
References: <3595789d7a8d24671250e2ca04fcf78eb04179a4.camel@ericsson.com> <EFF58041-128E-42F6-9303-A058A5A02E22@iii.ca> <CAOJ7v-3x=MHUzqwSeXnkAveJ2vEvp6OS1_OTQvOARrnnmFC6fg@mail.gmail.com>
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Subject: Re: [V3] Gateway new to old
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It may be a reasonable goal to aim to re-use RTP payload formats as the =
media packetisation within the new transport. That is, different headers =
and transport, but format the media content the same way.

Since payload format packetisation already all have registered media =
types this would make the media content easy to identify, and save a lot =
of work in re-defining packetisation schemes.

Colin



> On 27 Mar 2020, at 20:22, Justin Uberti =
<juberti=3D40google.com@dmarc.ietf.org> wrote:
>=20
> It's hard to imagine that we can avoid this if we are in fact going to =
use H3 as a substrate.=20
>=20
> I do think it would be useful to ensure the RIPT->RTP gateway can be =
stateless and transparent (i.e., you can transform a RIPT frame into a =
RTP packet with reasonable fidelity), so that said gateway is all you =
need in order for existing services to participate in the RIPT =
ecosystem.
>=20
> On Fri, Mar 27, 2020 at 9:32 AM Cullen Jennings <fluffy@iii.ca =
<mailto:fluffy@iii.ca>> wrote:
>=20
>=20
>> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund =
<magnus.westerlund=3D40ericsson.com@dmarc.ietf.org =
<mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org>> wrote:
>>=20
>> The next media transport issue that was barely discussed in the BOF =
is the
>> legacy (RTP) interoperate question. This is going to come back in the =
scope
>> discussion and do needs at least a direction.=20
>=20
> In WebRTC, we decided to be able to gateway SIP to WebRTC without =
processing media. That turned out to be a major limitation to how WebRTC =
worked. That is good and we have WebRTC but for this, I am storngly of =
the view we should not have that design constraint.=20
>=20
> I think it is fine to require a media gateway to go from RIPT to RTP. =
Sure that should be pretty easy - one of the major use cases for this =
work will likely involve building SBCs that talk RIPT on one side and =
SIP on the other. Pretty sure that a bunch of other people are thinking =
about the same.=20
>=20
> That sound right to you are or are you arguing for something else ?
>=20
>=20
> --=20
> V3 mailing list
> V3@ietf.org <mailto:V3@ietf.org>
> https://www.ietf.org/mailman/listinfo/v3 =
<https://www.ietf.org/mailman/listinfo/v3>
> --=20
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3


--Apple-Mail=_B3625465-9EE6-4875-A2D6-955C0D79B77B
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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""><div =
class=3D"">It may be a reasonable goal to aim to re-use RTP payload =
formats as the media packetisation within the new transport. That is, =
different headers and transport, but format the media content the same =
way.</div><div class=3D""><br class=3D""></div><div class=3D"">Since =
payload format packetisation already all have registered media types =
this would make the media content easy to identify, and save a lot of =
work in re-defining packetisation schemes.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Colin</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""><div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">On 27 =
Mar 2020, at 20:22, Justin Uberti &lt;<a =
href=3D"mailto:juberti=3D40google.com@dmarc.ietf.org" =
class=3D"">juberti=3D40google.com@dmarc.ietf.org</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D"">It's hard to imagine that we can avoid this if we are in fact =
going to use H3 as a substrate.&nbsp;<div class=3D""><br =
class=3D""></div><div class=3D"">I do think it would be useful to ensure =
the RIPT-&gt;RTP gateway can be stateless and transparent (i.e., you can =
transform a RIPT frame into a RTP packet with reasonable fidelity), so =
that said gateway is all you need in order for existing services to =
participate&nbsp;in the RIPT ecosystem.</div></div><br class=3D""><div =
class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Fri, Mar =
27, 2020 at 9:32 AM Cullen Jennings &lt;<a href=3D"mailto:fluffy@iii.ca" =
class=3D"">fluffy@iii.ca</a>&gt; wrote:<br class=3D""></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;" class=3D""><br class=3D""><div class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">On Mar =
27, 2020, at 8:23 AM, Magnus Westerlund &lt;<a =
href=3D"mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org" =
target=3D"_blank" =
class=3D"">magnus.westerlund=3D40ericsson.com@dmarc.ietf.org</a>&gt; =
wrote:</div><br class=3D""><div class=3D""><span =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none;float:none;display:inline" class=3D"">The next media =
transport issue that was barely discussed in the BOF is the</span><br =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none" class=3D""><span =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none;float:none;display:inline" class=3D"">legacy (RTP) =
interoperate question. This is going to come back in the scope</span><br =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none" class=3D""><span =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none;float:none;display:inline" class=3D"">discussion and =
do needs at least a direction.<span =
class=3D"">&nbsp;</span></span></div></blockquote></div><br =
class=3D""><div class=3D"">In WebRTC, we decided to be able to gateway =
SIP to WebRTC without processing media. That turned out to be a major =
limitation to how WebRTC worked. That is good and we have WebRTC but for =
this, I am storngly of the view we should not have that design =
constraint.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">I think it is fine to require a media gateway to go from RIPT =
to RTP. Sure that should be pretty easy - one of the major use cases for =
this work will likely involve building SBCs that talk RIPT on one side =
and SIP on the other. Pretty sure that a bunch of other people are =
thinking about the same.&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D"">That sound right to you are or are you =
arguing for something else ?</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div></div>-- <br =
class=3D"">
V3 mailing list<br class=3D"">
<a href=3D"mailto:V3@ietf.org" target=3D"_blank" =
class=3D"">V3@ietf.org</a><br class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" =
target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/v3</a><br class=3D"">
</blockquote></div>
-- <br class=3D"">V3 mailing list<br class=3D""><a =
href=3D"mailto:V3@ietf.org" class=3D"">V3@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/v3<br =
class=3D""></div></blockquote></div><br class=3D""></div></body></html>=

--Apple-Mail=_B3625465-9EE6-4875-A2D6-955C0D79B77B--


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
To: "juberti@google.com" <juberti@google.com>, "csp@csperkins.org" <csp@csperkins.org>
CC: "v3@ietf.org" <v3@ietf.org>, "fluffy@iii.ca" <fluffy@iii.ca>
Thread-Topic: [V3] Gateway new to old
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Archived-At: <https://mailarchive.ietf.org/arch/msg/v3/rE_bFQvnencOolOonTWmRwSExGQ>
Subject: Re: [V3] Gateway new to old
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Hi,

(Still as invidual)

Attempting to answer Cullen's question I think there are significant points=
 in
trying to actually enabling the utilization of the advantages this redefini=
tion
can enable. For example to me it appears that one should ensure that the me=
dia
transport becomes more ADU focused rather than packet focused. With 1080p o=
r 4K
video in good quality the number of 1400 bytes packets needed to carry a si=
ngle
frame are more in the range of hundreds than in single digit. I think one s=
hould
try to ensure that this work as well as possible. Thus semi-reliable head o=
f
line blocking free transport of whole ADUs becomes of more interest.=20

I think there is a point to re-use the RTP payload format. If one has such
support in the end-point one can use the RTP payload fragmentation when nee=
ded
for interconnect with RTP systems and not use it if one can rely on QUIC+RI=
PT
primitives. Reusing RTP payload formats also have the advantage of avoiding=
 to
have to redefine all the codecs of interest.

Cheers

Magnus Westerlund


On Sun, 2020-03-29 at 23:05 +0100, Colin Perkins wrote:
> It may be a reasonable goal to aim to re-use RTP payload formats as the m=
edia
> packetisation within the new transport. That is, different headers and
> transport, but format the media content the same way.
>=20
> Since payload format packetisation already all have registered media type=
s
> this would make the media content easy to identify, and save a lot of wor=
k in
> re-defining packetisation schemes.
>=20
> Colin
>=20
>=20
>=20
> > On 27 Mar 2020, at 20:22, Justin Uberti <juberti=3D40google.com@dmarc.i=
etf.org
> > > wrote:
> >=20
> > It's hard to imagine that we can avoid this if we are in fact going to =
use
> > H3 as a substrate.=20
> >=20
> > I do think it would be useful to ensure the RIPT->RTP gateway can be
> > stateless and transparent (i.e., you can transform a RIPT frame into a =
RTP
> > packet with reasonable fidelity), so that said gateway is all you need =
in
> > order for existing services to participate in the RIPT ecosystem.
> >=20
> > On Fri, Mar 27, 2020 at 9:32 AM Cullen Jennings <fluffy@iii.ca> wrote:
> > >=20
> > > > On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
> > > > magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
> > > >=20
> > > > The next media transport issue that was barely discussed in the BOF=
 is
> > > > the
> > > > legacy (RTP) interoperate question. This is going to come back in t=
he
> > > > scope
> > > > discussion and do needs at least a direction.=20
> > >=20
> > > In WebRTC, we decided to be able to gateway SIP to WebRTC without
> > > processing media. That turned out to be a major limitation to how Web=
RTC
> > > worked. That is good and we have WebRTC but for this, I am storngly o=
f the
> > > view we should not have that design constraint.=20
> > >=20
> > > I think it is fine to require a media gateway to go from RIPT to RTP.=
 Sure
> > > that should be pretty easy - one of the major use cases for this work=
 will
> > > likely involve building SBCs that talk RIPT on one side and SIP on th=
e
> > > other. Pretty sure that a bunch of other people are thinking about th=
e
> > > same.=20
> > >=20
> > > That sound right to you are or are you arguing for something else ?
> > >=20
> > >=20
> > > --=20
> > > V3 mailing list
> > > V3@ietf.org
> > > https://www.ietf.org/mailman/listinfo/v3
> >=20
> > --=20
> > V3 mailing list
> > V3@ietf.org
> > https://www.ietf.org/mailman/listinfo/v3
>=20
>=20
--=20
Cheers

Magnus Westerlund=20


----------------------------------------------------------------------
Networks, Ericsson Research
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Torshamnsgatan 23           | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------



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From nobody Mon Mar 30 06:23:40 2020
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To: Magnus Westerlund <magnus.westerlund=40ericsson.com@dmarc.ietf.org>, "juberti@google.com" <juberti@google.com>, "csp@csperkins.org" <csp@csperkins.org>
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From: Joerg Ott <ott@in.tum.de>
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Subject: Re: [V3] Gateway new to old
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Hi,

one should add that the payload formats are mostly not overly complex
and their design is defined by the needs of the codecs -- plus mapping
to an unreliable packet transport.  The codecs don't change and the
unreliability still holds with QUIC underneath.  So, there would seem
to be little gain to begin with in attempting to redo all of this.

Cheers,
Jörg

On 30.03.20 15:16, Magnus Westerlund wrote:
> Hi,
> 
> (Still as invidual)
> 
> Attempting to answer Cullen's question I think there are significant points in
> trying to actually enabling the utilization of the advantages this redefinition
> can enable. For example to me it appears that one should ensure that the media
> transport becomes more ADU focused rather than packet focused. With 1080p or 4K
> video in good quality the number of 1400 bytes packets needed to carry a single
> frame are more in the range of hundreds than in single digit. I think one should
> try to ensure that this work as well as possible. Thus semi-reliable head of
> line blocking free transport of whole ADUs becomes of more interest.
> 
> I think there is a point to re-use the RTP payload format. If one has such
> support in the end-point one can use the RTP payload fragmentation when needed
> for interconnect with RTP systems and not use it if one can rely on QUIC+RIPT
> primitives. Reusing RTP payload formats also have the advantage of avoiding to
> have to redefine all the codecs of interest.
> 
> Cheers
> 
> Magnus Westerlund
> 
> 
> On Sun, 2020-03-29 at 23:05 +0100, Colin Perkins wrote:
>> It may be a reasonable goal to aim to re-use RTP payload formats as the media
>> packetisation within the new transport. That is, different headers and
>> transport, but format the media content the same way.
>>
>> Since payload format packetisation already all have registered media types
>> this would make the media content easy to identify, and save a lot of work in
>> re-defining packetisation schemes.
>>
>> Colin
>>
>>
>>
>>> On 27 Mar 2020, at 20:22, Justin Uberti <juberti=40google.com@dmarc.ietf.org
>>>> wrote:
>>>
>>> It's hard to imagine that we can avoid this if we are in fact going to use
>>> H3 as a substrate.
>>>
>>> I do think it would be useful to ensure the RIPT->RTP gateway can be
>>> stateless and transparent (i.e., you can transform a RIPT frame into a RTP
>>> packet with reasonable fidelity), so that said gateway is all you need in
>>> order for existing services to participate in the RIPT ecosystem.
>>>
>>> On Fri, Mar 27, 2020 at 9:32 AM Cullen Jennings <fluffy@iii.ca> wrote:
>>>>
>>>>> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
>>>>> magnus.westerlund=40ericsson.com@dmarc.ietf.org> wrote:
>>>>>
>>>>> The next media transport issue that was barely discussed in the BOF is
>>>>> the
>>>>> legacy (RTP) interoperate question. This is going to come back in the
>>>>> scope
>>>>> discussion and do needs at least a direction.
>>>>
>>>> In WebRTC, we decided to be able to gateway SIP to WebRTC without
>>>> processing media. That turned out to be a major limitation to how WebRTC
>>>> worked. That is good and we have WebRTC but for this, I am storngly of the
>>>> view we should not have that design constraint.
>>>>
>>>> I think it is fine to require a media gateway to go from RIPT to RTP. Sure
>>>> that should be pretty easy - one of the major use cases for this work will
>>>> likely involve building SBCs that talk RIPT on one side and SIP on the
>>>> other. Pretty sure that a bunch of other people are thinking about the
>>>> same.
>>>>
>>>> That sound right to you are or are you arguing for something else ?
>>>>
>>>>
>>>> -- 
>>>> V3 mailing list
>>>> V3@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/v3
>>>
>>> -- 
>>> V3 mailing list
>>> V3@ietf.org
>>> https://www.ietf.org/mailman/listinfo/v3
>>
>>
>>


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> On Mar 29, 2020, at 4:05 PM, Colin Perkins <csp@csperkins.org> wrote:
>=20
> It may be a reasonable goal to aim to re-use RTP payload formats as =
the media packetisation within the new transport. That is, different =
headers and transport, but format the media content the same way.
>=20
> Since payload format packetisation already all have registered media =
types this would make the media content easy to identify, and save a lot =
of work in re-defining packetisation schemes.
>=20
> Colin
>=20
>=20

+1 on reusing the RTP payload formats. As Joerg points out they are a =
good design for what is needed here.=20


--Apple-Mail=_35E70BE7-8B15-430D-B534-AD6EE2060C27
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	charset=us-ascii

<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""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On Mar 29, 2020, at 4:05 PM, Colin Perkins &lt;<a =
href=3D"mailto:csp@csperkins.org" class=3D"">csp@csperkins.org</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
class=3D"" style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; =
font-size: 14px; 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;">It may be a reasonable goal to aim to re-use RTP payload formats =
as the media packetisation within the new transport. That is, different =
headers and transport, but format the media content the same =
way.</div><div class=3D"" style=3D"caret-color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 14px; 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;"><br class=3D""></div><div class=3D"" =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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;">Since payload =
format packetisation already all have registered media types this would =
make the media content easy to identify, and save a lot of work in =
re-defining packetisation schemes.</div><div class=3D"" =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
14px; 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;"><br =
class=3D""></div><div class=3D"" style=3D"caret-color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 14px; 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;">Colin</div><div class=3D"" style=3D"caret-color: =
rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; 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;"><br class=3D""></div><br =
class=3D"Apple-interchange-newline"></div></blockquote></div><br =
class=3D""><div class=3D"">+1 on reusing the RTP payload formats. As =
Joerg points out they are a good design for what is needed =
here.&nbsp;</div><div class=3D""><br class=3D""></div></body></html>=

--Apple-Mail=_35E70BE7-8B15-430D-B534-AD6EE2060C27--


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Archived-At: <https://mailarchive.ietf.org/arch/msg/v3/3FyfYDn7cOLg0eJNIrKelIMJ3fo>
Subject: Re: [V3] High level comments on the RIPT charter and scope
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On Fri, 2020-03-27 at 10:19 -0600, Cullen Jennings wrote:
>=20
>=20
> > On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
> > magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
> >=20
> > The third big part is the media transport. I think we need to strongly
> > consider
> > if this can be broken out in to it seperate WG.
>=20
> I think the WG we are talking about here is the WG to do media of H3. So =
we
> can talk about other things being in or out of it but that is the core th=
ing.=20
>=20
> I would like to point out that one of the design properties that has been
> making this easier is that media is self describing. What I mean by that =
is
> that if a device receives media that is one of the supported formats and
> within in the limits of that device supports, the media has all the
> information the device needs to decode it. No out of band information nee=
ded.
> That is key to making this easier to deal with HA and scalability on clou=
d
> side. I think that is a good property for modern designs. We have been go=
ing
> that way for a long time with things like opus and it works out well.=20

So, let me ask a question here. How important is it be capable to transmitt
individual media ADU or even RTP payloads in a single HTTP request that rip=
ples
through the system independent compared to have a system that can in a 0-rt=
t
fashion (re-)establish an point to point by initiating an HTTP request?=20

Treating each media ADU as its own request potentially being routed in a no=
n-
sticky way will induce significant jitter would it not? The drafts talks ab=
out
sticky routing, and I did notice some comments about that assumption alread=
y.
Thus, from my perspective it appears that the high-performance target would=
 be
to establish an point to point quic based transport with the relevant serve=
r
node that handles media forwarding. If that fails have a mechanism that can
rapidly re-establish that connection with what ever node that is now the
appointed one for the given call/conference.=20

Then I do understand the desire to have a fallback, and maybe that is plain=
 HTTP
requests, but let that be the fallback and not the primary mechanism. Becas=
ue I
think that would unnecessary limit the potential of this system.=20

>=20
> > here are several aspects here.
> > First of the split in expertise between the media transport and the
> > signalling
> > folks.=20
>=20
> I=E2=80=99m not sure what you mean by signalling but the the singling her=
e was mostly
> start media , stop media. It is not call routing, forking etc that is don=
e by
> SIP.=20

Yes, but even session establishment and control is still signalling from my
perspective.=20

=20
Cheers

Magnus Westerlund=20


----------------------------------------------------------------------
Networks, Ericsson Research
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Torshamnsgatan 23           | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------



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From: Cullen Jennings <fluffy@iii.ca>
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To: Justin Uberti <juberti=40google.com@dmarc.ietf.org>, "v3@ietf.org" <v3@ietf.org>
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On the stateless or not =E2=80=A6 I=E2=80=99ve tried implementing this a =
few different ways and I think what we want is a sort of mostly =
stateless with frequent sate updates for the parts that are not =
stateless. Let me explain =E2=80=A6.

First keep in mind there is nothing stateless about the QUIC connection. =
Lots of state for crypto. It also seems like it would be a good thing to =
allow state for compression. So we are already out of a fully stateless =
design before we even start.=20

So when we say RIPT can map to RTP, we really mean RTP + RTCP as that is =
needed. In the RTCP we need things like the NTP timestamp. I don=E2=80=99t=
 think we want to send the full NTP timestamp in ever RIPT packet which =
probably rules out a fully stateless design. In WebRTC we took the =
approach of putting some data in RTP header extensions and sending them =
not ever packet but relatively frequently like every second or every =
IFrame.  I think we should do the same thing here - we should have some =
stuff that is sent frequently enough that if you loose all state until =
it is resent, it is  not a huge deal.=20

This would allow us to do things like send the info to map to NTP time =
once a second not in every packet. When you look at what is in a basic =
RTP packet, I think it is easy to have that 1:1 mapping of all that data =
to something in the RIPT packet but when you start looking at the RTCP =
data or some of the RTP header extensions, I think that needs to all map =
to stuff in RIPT but probably not something that has to be every RIPT =
packet.=20

Hope that makes sense. That said, I think this still is an extremely =
mostly 1:1 and simple 20 lines of code to map between RTP/RTCP and RIPT =
but I think we will end up with a bit of state that if frequently =
refreshed and is no worse than loosing an IFrame if you loose that =
state.=20
=20

> On Mar 27, 2020, at 2:22 PM, Justin Uberti =
<juberti=3D40google.com@dmarc.ietf.org> wrote:
>=20
> It's hard to imagine that we can avoid this if we are in fact going to =
use H3 as a substrate.=20
>=20
> I do think it would be useful to ensure the RIPT->RTP gateway can be =
stateless and transparent (i.e., you can transform a RIPT frame into a =
RTP packet with reasonable fidelity), so that said gateway is all you =
need in order for existing services to participate in the RIPT =
ecosystem.
>=20
> On Fri, Mar 27, 2020 at 9:32 AM Cullen Jennings <fluffy@iii.ca =
<mailto:fluffy@iii.ca>> wrote:
>=20
>=20
>> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund =
<magnus.westerlund=3D40ericsson.com@dmarc.ietf.org =
<mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org>> wrote:
>>=20
>> The next media transport issue that was barely discussed in the BOF =
is the
>> legacy (RTP) interoperate question. This is going to come back in the =
scope
>> discussion and do needs at least a direction.=20
>=20
> In WebRTC, we decided to be able to gateway SIP to WebRTC without =
processing media. That turned out to be a major limitation to how WebRTC =
worked. That is good and we have WebRTC but for this, I am storngly of =
the view we should not have that design constraint.=20
>=20
> I think it is fine to require a media gateway to go from RIPT to RTP. =
Sure that should be pretty easy - one of the major use cases for this =
work will likely involve building SBCs that talk RIPT on one side and =
SIP on the other. Pretty sure that a bunch of other people are thinking =
about the same.=20
>=20
> That sound right to you are or are you arguing for something else ?
>=20
>=20
> --=20
> V3 mailing list
> V3@ietf.org <mailto:V3@ietf.org>
> https://www.ietf.org/mailman/listinfo/v3 =
<https://www.ietf.org/mailman/listinfo/v3>
> --=20
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3


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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""><div =
class=3D""><br class=3D""></div>On the stateless or not =E2=80=A6 I=E2=80=99=
ve tried implementing this a few different ways and I think what we want =
is a sort of mostly stateless with frequent sate updates for the parts =
that are not stateless. Let me explain =E2=80=A6.<div class=3D""><br =
class=3D""></div><div class=3D"">First keep in mind there is nothing =
stateless about the QUIC connection. Lots of state for crypto. It also =
seems like it would be a good thing to allow state for compression. So =
we are already out of a fully stateless design before we even =
start.&nbsp;</div><div class=3D""><br class=3D""></div><div class=3D"">So =
when we say RIPT can map to RTP, we really mean RTP + RTCP as that is =
needed. In the RTCP we need things like the NTP timestamp. I don=E2=80=99t=
 think we want to send the full NTP timestamp in ever RIPT packet which =
probably rules out a fully stateless design. In WebRTC we took the =
approach of putting some data in RTP header extensions and sending them =
not ever packet but relatively frequently like every second or every =
IFrame. &nbsp;I think we should do the same thing here - we should have =
some stuff that is sent frequently enough that if you loose all state =
until it is resent, it is &nbsp;not a huge deal.&nbsp;</div><div =
class=3D""><br class=3D""></div><div class=3D"">This would allow us to =
do things like send the info to map to NTP time once a second not in =
every packet. When you look at what is in a basic RTP packet, I think it =
is easy to have that 1:1 mapping of all that data to something in the =
RIPT packet but when you start looking at the RTCP data or some of the =
RTP header extensions, I think that needs to all map to stuff in RIPT =
but probably not something that has to be every RIPT =
packet.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">Hope that makes sense. That said, I think this still is an =
extremely mostly 1:1 and simple 20 lines of code to map between RTP/RTCP =
and RIPT but I think we will end up with a bit of state that if =
frequently refreshed and is no worse than loosing an IFrame if you loose =
that state.&nbsp;</div><div class=3D"">&nbsp;<br class=3D""><div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">On Mar =
27, 2020, at 2:22 PM, Justin Uberti &lt;<a =
href=3D"mailto:juberti=3D40google.com@dmarc.ietf.org" =
class=3D"">juberti=3D40google.com@dmarc.ietf.org</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D"">It's hard to imagine that we can avoid this if we are in fact =
going to use H3 as a substrate.&nbsp;<div class=3D""><br =
class=3D""></div><div class=3D"">I do think it would be useful to ensure =
the RIPT-&gt;RTP gateway can be stateless and transparent (i.e., you can =
transform a RIPT frame into a RTP packet with reasonable fidelity), so =
that said gateway is all you need in order for existing services to =
participate&nbsp;in the RIPT ecosystem.</div></div><br class=3D""><div =
class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Fri, Mar =
27, 2020 at 9:32 AM Cullen Jennings &lt;<a href=3D"mailto:fluffy@iii.ca" =
class=3D"">fluffy@iii.ca</a>&gt; wrote:<br class=3D""></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;" class=3D""><br class=3D""><div class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">On Mar =
27, 2020, at 8:23 AM, Magnus Westerlund &lt;<a =
href=3D"mailto:magnus.westerlund=3D40ericsson.com@dmarc.ietf.org" =
target=3D"_blank" =
class=3D"">magnus.westerlund=3D40ericsson.com@dmarc.ietf.org</a>&gt; =
wrote:</div><br class=3D""><div class=3D""><span =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none;float:none;display:inline" class=3D"">The next media =
transport issue that was barely discussed in the BOF is the</span><br =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none" class=3D""><span =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none;float:none;display:inline" class=3D"">legacy (RTP) =
interoperate question. This is going to come back in the scope</span><br =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none" class=3D""><span =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varia=
nt-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;t=
ext-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;tex=
t-decoration:none;float:none;display:inline" class=3D"">discussion and =
do needs at least a direction.<span =
class=3D"">&nbsp;</span></span></div></blockquote></div><br =
class=3D""><div class=3D"">In WebRTC, we decided to be able to gateway =
SIP to WebRTC without processing media. That turned out to be a major =
limitation to how WebRTC worked. That is good and we have WebRTC but for =
this, I am storngly of the view we should not have that design =
constraint.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">I think it is fine to require a media gateway to go from RIPT =
to RTP. Sure that should be pretty easy - one of the major use cases for =
this work will likely involve building SBCs that talk RIPT on one side =
and SIP on the other. Pretty sure that a bunch of other people are =
thinking about the same.&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D"">That sound right to you are or are you =
arguing for something else ?</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div></div>-- <br =
class=3D"">
V3 mailing list<br class=3D"">
<a href=3D"mailto:V3@ietf.org" target=3D"_blank" =
class=3D"">V3@ietf.org</a><br class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" =
target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/v3</a><br class=3D"">
</blockquote></div>
-- <br class=3D"">V3 mailing list<br class=3D""><a =
href=3D"mailto:V3@ietf.org" class=3D"">V3@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/v3<br =
class=3D""></div></blockquote></div><br class=3D""></div></body></html>=

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From nobody Mon Mar 30 07:15:32 2020
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From: Cullen Jennings <fluffy@iii.ca>
In-Reply-To: <3c2943c636663908da232674719abbf6c7ea7918.camel@ericsson.com>
Date: Mon, 30 Mar 2020 08:15:13 -0600
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [V3] Centralized conference media
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I=E2=80=99m thinking of this as 100% client / server.  Sure you can =
build a conferences on top of that but the RIPT layer would not even =
need to have a clue about the overall conference media topology. The =
server can do things like pause / resume media FIR etc but it all just =
two parties involved - the client and the server. Assuming a server that =
maps media is one of the things that make this so much easier than an =
RTP multicast conferences.=20


> On Mar 27, 2020, at 4:35 PM, Magnus Westerlund =
<magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
>=20
> Hi,
>=20
> So yes, I think there are the aspect of the topology as well as how =
one handle
> the media related controll messages, i.e. the equivalent of RTCP. For =
example
> things like FIR, PAUSE/Resume.=20
>=20
> So, if one with RTCP terminating middleboxes means that the ID spaces =
are soley
> locally that could be a choice. I will note that in some cases one =
still need to
> have the complete set of media sources from all end-points in a =
conference be
> identified at a specific point to point leg. Even if only a subset is =
currently
> transmitted over this leg. There are both benefits and downsides I =
think of
> using a local namespace and having each node translate these.
>=20
> I will note that one may still need to have end-to-end identies on =
media ADUs if
> one are going to have any end-to-end security.=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
>=20
>=20
> On Fri, 2020-03-27 at 09:28 -0700, Justin Uberti wrote:
>> I think Magnus is trying to establish the conference topology, which =
has
>> ramifications on how we want to handle things like RTCP.
>>=20
>> One strawman could be to focus on RTCP-terminating SFUs, i.e., =
cementing the
>> notion that RIPT media flows are indeed point-to-point (i..e., =
client-to-SFU,
>> as opposed to client-to-client1, client2, client3).
>>=20
>> On Fri, Mar 27, 2020 at 9:11 AM Cullen Jennings <fluffy@iii.ca> =
wrote:
>>>=20
>>>> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
>>>> magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
>>>>=20
>>>> I am also very worried that if the model for the media transport is =
only
>>>> point
>>>> to point calls then we get something that will become obsolete or =
at least
>>>> need
>>>> hacks immediately. This thing need to support centralized =
conferences with
>>>> cascaded or meshed central media nodes. So I think it is cruical =
that one
>>>> define
>>>> the usage model with switched central nodes and have that in mind =
so that
>>>> one
>>>> can actually support the applications needed. I think by defining a =
strict
>>>> model
>>>> one can actually avoid a lot of feature creep in the media =
transport.=20
>>>=20
>>> I think it does make sense to support some meta data on a media flow =
for
>>> centralized conferencing support. What semantic level information do =
you
>>> think is needed in the media ?
>>>=20
>>>=20
>>>=20
>>> --=20
>>> V3 mailing list
>>> V3@ietf.org
>>> https://www.ietf.org/mailman/listinfo/v3
> --=20
> Cheers
>=20
> Magnus Westerlund=20
>=20
>=20
> ----------------------------------------------------------------------
> Networks, Ericsson Research
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Torshamnsgatan 23           | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
> --=20
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3


From nobody Mon Mar 30 07:37:41 2020
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From: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
Date: Mon, 30 Mar 2020 09:37:07 -0500
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To: Magnus Westerlund <magnus.westerlund=40ericsson.com@dmarc.ietf.org>
Cc: "fluffy@iii.ca" <fluffy@iii.ca>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] High level comments on the RIPT charter and scope
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For what it's worth,

On Mon, Mar 30, 2020 at 8:44 AM Magnus Westerlund <magnus.westerlund=3D
40ericsson.com@dmarc.ietf.org> wrote:

> On Fri, 2020-03-27 at 10:19 -0600, Cullen Jennings wrote:
> >
> >
> > > On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
> > > magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
> > >
> > > The third big part is the media transport. I think we need to strongl=
y
> > > consider
> > > if this can be broken out in to it seperate WG.
> >
> > I think the WG we are talking about here is the WG to do media of H3. S=
o
> we
> > can talk about other things being in or out of it but that is the core
> thing.
> >
> > I would like to point out that one of the design properties that has be=
en
> > making this easier is that media is self describing. What I mean by tha=
t
> is
> > that if a device receives media that is one of the supported formats an=
d
> > within in the limits of that device supports, the media has all the
> > information the device needs to decode it. No out of band information
> needed.
> > That is key to making this easier to deal with HA and scalability on
> cloud
> > side. I think that is a good property for modern designs. We have been
> going
> > that way for a long time with things like opus and it works out well.
>
> So, let me ask a question here. How important is it be capable to transmi=
tt
> individual media ADU or even RTP payloads in a single HTTP request that
> ripples
> through the system independent compared to have a system that can in a
> 0-rtt
> fashion (re-)establish an point to point by initiating an HTTP request?
>
> Treating each media ADU as its own request potentially being routed in a
> non-
> sticky way will induce significant jitter would it not? The drafts talks
> about
> sticky routing, and I did notice some comments about that assumption
> already.
> Thus, from my perspective it appears that the high-performance target
> would be
> to establish an point to point quic based transport with the relevant
> server
> node that handles media forwarding. If that fails have a mechanism that c=
an
> rapidly re-establish that connection with what ever node that is now the
> appointed one for the given call/conference.
>
> Then I do understand the desire to have a fallback, and maybe that is
> plain HTTP
> requests, but let that be the fallback and not the primary mechanism.
> Becasue I
> think that would unnecessary limit the potential of this system.
>
> >
> > > here are several aspects here.
> > > First of the split in expertise between the media transport and the
> > > signalling
> > > folks.
> >
> > I=E2=80=99m not sure what you mean by signalling but the the singling h=
ere was
> mostly
> > start media , stop media. It is not call routing, forking etc that is
> done by
> > SIP.
>
> Yes, but even session establishment and control is still signalling from =
my
> perspective.
>

I've certainly been thinking of "everything except media payload" as
signaling in my own comments. So, close to Magnus here, IIUC.

If we don't have a common understanding of this (and it sounds like we
might not), that would be a good thing to talk about during the charter
development.

Best,

Spencer


> Cheers
>
> Magnus Westerlund
>
>
> ----------------------------------------------------------------------
> Networks, Ericsson Research
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Torshamnsgatan 23           | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>

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<div dir=3D"ltr"><div dir=3D"ltr">For what it&#39;s worth,=C2=A0</div><br><=
div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Mar=
 30, 2020 at 8:44 AM Magnus Westerlund &lt;magnus.westerlund=3D<a href=3D"m=
ailto:40ericsson.com@dmarc.ietf.org">40ericsson.com@dmarc.ietf.org</a>&gt; =
wrote:<br></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">On Fri, 2=
020-03-27 at 10:19 -0600, Cullen Jennings wrote:<br>
&gt; <br>
&gt; <br>
&gt; &gt; On Mar 27, 2020, at 8:23 AM, Magnus Westerlund &lt;<br>
&gt; &gt; magnus.westerlund=3D<a href=3D"mailto:40ericsson.com@dmarc.ietf.o=
rg" target=3D"_blank">40ericsson.com@dmarc.ietf.org</a>&gt; wrote:<br>
&gt; &gt; <br>
&gt; &gt; The third big part is the media transport. I think we need to str=
ongly<br>
&gt; &gt; consider<br>
&gt; &gt; if this can be broken out in to it seperate WG.<br>
&gt; <br>
&gt; I think the WG we are talking about here is the WG to do media of H3. =
So we<br>
&gt; can talk about other things being in or out of it but that is the core=
 thing. <br>
&gt; <br>
&gt; I would like to point out that one of the design properties that has b=
een<br>
&gt; making this easier is that media is self describing. What I mean by th=
at is<br>
&gt; that if a device receives media that is one of the supported formats a=
nd<br>
&gt; within in the limits of that device supports, the media has all the<br=
>
&gt; information the device needs to decode it. No out of band information =
needed.<br>
&gt; That is key to making this easier to deal with HA and scalability on c=
loud<br>
&gt; side. I think that is a good property for modern designs. We have been=
 going<br>
&gt; that way for a long time with things like opus and it works out well. =
<br>
<br>
So, let me ask a question here. How important is it be capable to transmitt=
<br>
individual media ADU or even RTP payloads in a single HTTP request that rip=
ples<br>
through the system independent compared to have a system that can in a 0-rt=
t<br>
fashion (re-)establish an point to point by initiating an HTTP request? <br=
>
<br>
Treating each media ADU as its own request potentially being routed in a no=
n-<br>
sticky way will induce significant jitter would it not? The drafts talks ab=
out<br>
sticky routing, and I did notice some comments about that assumption alread=
y.<br>
Thus, from my perspective it appears that the high-performance target would=
 be<br>
to establish an point to point quic based transport with the relevant serve=
r<br>
node that handles media forwarding. If that fails have a mechanism that can=
<br>
rapidly re-establish that connection with what ever node that is now the<br=
>
appointed one for the given call/conference. <br>
<br>
Then I do understand the desire to have a fallback, and maybe that is plain=
 HTTP<br>
requests, but let that be the fallback and not the primary mechanism. Becas=
ue I<br>
think that would unnecessary limit the potential of this system. <br>
<br>
&gt; <br>
&gt; &gt; here are several aspects here.<br>
&gt; &gt; First of the split in expertise between the media transport and t=
he<br>
&gt; &gt; signalling<br>
&gt; &gt; folks. <br>
&gt; <br>
&gt; I=E2=80=99m not sure what you mean by signalling but the the singling =
here was mostly<br>
&gt; start media , stop media. It is not call routing, forking etc that is =
done by<br>
&gt; SIP. <br>
<br>
Yes, but even session establishment and control is still signalling from my=
<br>
perspective. <br></blockquote><div><br></div><div>I&#39;ve certainly been t=
hinking of &quot;everything except media payload&quot; as signaling in my o=
wn comments. So, close to Magnus here, IIUC.=C2=A0</div><div><br></div><div=
>If we don&#39;t have a common understanding of this (and it sounds like we=
 might not), that would be a good thing to talk about during the charter de=
velopment.=C2=A0</div><div><br></div><div>Best,</div><div><br></div><div>Sp=
encer</div><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">Cheers<br>
<br>
Magnus Westerlund <br>
<br>
<br>
----------------------------------------------------------------------<br>
Networks, Ericsson Research<br>
----------------------------------------------------------------------<br>
Ericsson AB=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =
Phone=C2=A0 +46 10 7148287<br>
Torshamnsgatan 23=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| Mobile +46 73 0=
949079<br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
<br>
-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div></div>

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Date: Mon, 30 Mar 2020 09:56:34 -0700
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On Mon, Mar 30, 2020 at 6:55 AM Cullen Jennings <fluffy@iii.ca> wrote:

>
> On the stateless or not =E2=80=A6 I=E2=80=99ve tried implementing this a =
few different
> ways and I think what we want is a sort of mostly stateless with frequent
> sate updates for the parts that are not stateless. Let me explain =E2=80=
=A6.
>
> First keep in mind there is nothing stateless about the QUIC connection.
> Lots of state for crypto. It also seems like it would be a good thing to
> allow state for compression. So we are already out of a fully stateless
> design before we even start.
>

Sure, there is lots of state for crypto. But this will typically be handled
by a frontend server at the edge of the cloud infra, so we should focus on
the RTP-specific parts.

>
> So when we say RIPT can map to RTP, we really mean RTP + RTCP as that is
> needed. In the RTCP we need things like the NTP timestamp. I don=E2=80=99=
t think we
> want to send the full NTP timestamp in ever RIPT packet which probably
> rules out a fully stateless design. In WebRTC we took the approach of
> putting some data in RTP header extensions and sending them not ever pack=
et
> but relatively frequently like every second or every IFrame.  I think we
> should do the same thing here - we should have some stuff that is sent
> frequently enough that if you loose all state until it is resent, it is
>  not a huge deal.
>
> This would allow us to do things like send the info to map to NTP time
> once a second not in every packet. When you look at what is in a basic RT=
P
> packet, I think it is easy to have that 1:1 mapping of all that data to
> something in the RIPT packet but when you start looking at the RTCP data =
or
> some of the RTP header extensions, I think that needs to all map to stuff
> in RIPT but probably not something that has to be every RIPT packet.
>
> Hope that makes sense. That said, I think this still is an extremely
> mostly 1:1 and simple 20 lines of code to map between RTP/RTCP and RIPT b=
ut
> I think we will end up with a bit of state that if frequently refreshed a=
nd
> is no worse than loosing an IFrame if you loose that state.
>

I can get on board with this. As you say, there already is some state with
RTP/RTCP in terms of the timestamp mapping. The main thing is that this
needs to be loss-tolerant (which makes compression somewhat harder) and
self-healing, e.g., on a gateway failover.

But yeah, it would be nice to not have to send the ~same 12 bytes + 0xBEDE
in every packet. (And this is why I think we can reach parity or better
with RTP efficiency.)

>
>
> On Mar 27, 2020, at 2:22 PM, Justin Uberti <
> juberti=3D40google.com@dmarc.ietf.org> wrote:
>
> It's hard to imagine that we can avoid this if we are in fact going to us=
e
> H3 as a substrate.
>
> I do think it would be useful to ensure the RIPT->RTP gateway can be
> stateless and transparent (i.e., you can transform a RIPT frame into a RT=
P
> packet with reasonable fidelity), so that said gateway is all you need in
> order for existing services to participate in the RIPT ecosystem.
>
> On Fri, Mar 27, 2020 at 9:32 AM Cullen Jennings <fluffy@iii.ca> wrote:
>
>>
>>
>> On Mar 27, 2020, at 8:23 AM, Magnus Westerlund <
>> magnus.westerlund=3D40ericsson.com@dmarc.ietf.org> wrote:
>>
>> The next media transport issue that was barely discussed in the BOF is t=
he
>> legacy (RTP) interoperate question. This is going to come back in the
>> scope
>> discussion and do needs at least a direction.
>>
>>
>> In WebRTC, we decided to be able to gateway SIP to WebRTC without
>> processing media. That turned out to be a major limitation to how WebRTC
>> worked. That is good and we have WebRTC but for this, I am storngly of t=
he
>> view we should not have that design constraint.
>>
>> I think it is fine to require a media gateway to go from RIPT to RTP.
>> Sure that should be pretty easy - one of the major use cases for this wo=
rk
>> will likely involve building SBCs that talk RIPT on one side and SIP on =
the
>> other. Pretty sure that a bunch of other people are thinking about the
>> same.
>>
>> That sound right to you are or are you arguing for something else ?
>>
>>
>> --
>> V3 mailing list
>> V3@ietf.org
>> https://www.ietf.org/mailman/listinfo/v3
>>
> --
> V3 mailing list
> V3@ietf.org
> https://www.ietf.org/mailman/listinfo/v3
>
>
>

--000000000000195a2e05a215561a
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<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, Mar 30, 2020 at 6:55 AM Culle=
n Jennings &lt;<a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</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 style=3D"=
overflow-wrap: break-word;"><div><br></div>On the stateless or not =E2=80=
=A6 I=E2=80=99ve tried implementing this a few different ways and I think w=
hat we want is a sort of mostly stateless with frequent sate updates for th=
e parts that are not stateless. Let me explain =E2=80=A6.<div><br></div><di=
v>First keep in mind there is nothing stateless about the QUIC connection. =
Lots of state for crypto. It also seems like it would be a good thing to al=
low state for compression. So we are already out of a fully stateless desig=
n before we even start.=C2=A0</div></div></blockquote><div><br></div><div>S=
ure, there is lots of state for crypto. But this will typically be handled =
by a frontend server at the edge of the cloud infra, so we should focus on =
the RTP-specific parts.=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><br></div><div>So=
 when we say RIPT can map to RTP, we really mean RTP + RTCP as that is need=
ed. In the RTCP we need things like the NTP timestamp. I don=E2=80=99t thin=
k we want to send the full NTP timestamp in ever RIPT packet which probably=
 rules out a fully stateless design. In WebRTC we took the approach of putt=
ing some data in RTP header extensions and sending them not ever packet but=
 relatively frequently like every second or every IFrame.=C2=A0 I think we =
should do the same thing here - we should have some stuff that is sent freq=
uently enough that if you loose all state until it is resent, it is =C2=A0n=
ot a huge deal.=C2=A0</div><div><br></div><div>This would allow us to do th=
ings like send the info to map to NTP time once a second not in every packe=
t. When you look at what is in a basic RTP packet, I think it is easy to ha=
ve that 1:1 mapping of all that data to something in the RIPT packet but wh=
en you start looking at the RTCP data or some of the RTP header extensions,=
 I think that needs to all map to stuff in RIPT but probably not something =
that has to be every RIPT packet.=C2=A0</div><div><br></div><div>Hope that =
makes sense. That said, I think this still is an extremely mostly 1:1 and s=
imple 20 lines of code to map between RTP/RTCP and RIPT but I think we will=
 end up with a bit of state that if frequently refreshed and is no worse th=
an loosing an IFrame if you loose that state.=C2=A0</div></div></blockquote=
><div><br></div><div>I can get on board with this. As you say, there alread=
y is some state with RTP/RTCP in terms of the timestamp mapping. The main t=
hing is that this needs to be loss-tolerant (which makes compression somewh=
at harder) and self-healing, e.g., on a gateway failover.</div><div><br></d=
iv><div>But yeah, it would be nice to not have to send the ~same 12 bytes=
=C2=A0+ 0xBEDE in every packet. (And this is why I think we can reach parit=
y or better with RTP efficiency.)</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 style=3D"overflow-wrap: break-word;"><div>=C2=A0<br><div=
><br><blockquote type=3D"cite"><div>On Mar 27, 2020, at 2:22 PM, Justin Ube=
rti &lt;<a href=3D"mailto:juberti=3D40google.com@dmarc.ietf.org" target=3D"=
_blank">juberti=3D40google.com@dmarc.ietf.org</a>&gt; wrote:</div><br><div>=
<div dir=3D"ltr">It&#39;s hard to imagine that we can avoid this if we are =
in fact going to use H3 as a substrate.=C2=A0<div><br></div><div>I do think=
 it would be useful to ensure the RIPT-&gt;RTP gateway can be stateless and=
 transparent (i.e., you can transform a RIPT frame into a RTP packet with r=
easonable fidelity), so that said gateway is all you need in order for exis=
ting services to participate=C2=A0in the RIPT ecosystem.</div></div><br><di=
v class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Fri, Mar 2=
7, 2020 at 9:32 AM Cullen Jennings &lt;<a href=3D"mailto:fluffy@iii.ca" tar=
get=3D"_blank">fluffy@iii.ca</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"><div><br><div><br><blockquote type=3D"cite"><di=
v>On Mar 27, 2020, at 8:23 AM, Magnus Westerlund &lt;<a href=3D"mailto:magn=
us.westerlund=3D40ericsson.com@dmarc.ietf.org" target=3D"_blank">magnus.wes=
terlund=3D40ericsson.com@dmarc.ietf.org</a>&gt; wrote:</div><br><div><span =
style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-varian=
t-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;tex=
t-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-d=
ecoration:none;float:none;display:inline">The next media transport issue th=
at was barely discussed in the BOF is the</span><br style=3D"font-family:He=
lvetica;font-size:14px;font-style:normal;font-variant-caps:normal;font-weig=
ht:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-trans=
form:none;white-space:normal;word-spacing:0px;text-decoration:none"><span s=
tyle=3D"font-family:Helvetica;font-size:14px;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-de=
coration:none;float:none;display:inline">legacy (RTP) interoperate question=
. This is going to come back in the scope</span><br style=3D"font-family:He=
lvetica;font-size:14px;font-style:normal;font-variant-caps:normal;font-weig=
ht:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-trans=
form:none;white-space:normal;word-spacing:0px;text-decoration:none"><span s=
tyle=3D"font-family:Helvetica;font-size:14px;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-de=
coration:none;float:none;display:inline">discussion and do needs at least a=
 direction.<span>=C2=A0</span></span></div></blockquote></div><br><div>In W=
ebRTC, we decided to be able to gateway SIP to WebRTC without processing me=
dia. That turned out to be a major limitation to how WebRTC worked. That is=
 good and we have WebRTC but for this, I am storngly of the view we should =
not have that design constraint.=C2=A0</div><div><br></div><div>I think it =
is fine to require a media gateway to go from RIPT to RTP. Sure that should=
 be pretty easy - one of the major use cases for this work will likely invo=
lve building SBCs that talk RIPT on one side and SIP on the other. Pretty s=
ure that a bunch of other people are thinking about the same.=C2=A0</div><d=
iv><br></div><div>That sound right to you are or are you arguing for someth=
ing else ?</div><div><br></div><div><br></div></div>-- <br>
V3 mailing list<br>
<a href=3D"mailto:V3@ietf.org" target=3D"_blank">V3@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/v3" rel=3D"noreferrer" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br>
</blockquote></div>
-- <br>V3 mailing list<br><a href=3D"mailto:V3@ietf.org" target=3D"_blank">=
V3@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/v3" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/v3</a><br></div></bloc=
kquote></div><br></div></div></blockquote></div></div>

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From: "Asveren, Tolga" <tasveren@rbbn.com>
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A basic question about how RIPT would make things better compared to RTP in=
 terms of making use of Public Cloud Load-balancers:

With RTP: One can use L4 LB distributing traffic to a Redundancy Group. The=
 pain point is that LB is slow in detecting failures of Redundancy Group me=
mbers.
With RIPT media: One can use HTTP LB distributing traffic a Redundancy Grou=
p. I would think failure detection characteristics will be similar to L4 LB=
 and delayed failure detection issue would continue to be an issue.

In a nutshell, what would one gain by using a HTTP LB instead of L4 LB?

Thanks,
Tolga


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<html><head><meta http-equiv=3DContent-Type content=3D"text/html; charset=
=3Dus-ascii"><meta name=3DGenerator content=3D"Microsoft Word 15 (filtered =
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<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3D"#0563C1=
" vlink=3D"#954F72"><div class=3DWordSection1><p class=3DMsoNormal>A basic =
question about how RIPT would make things better compared to RTP in terms o=
f making use of Public Cloud Load-balancers:<o:p></o:p></p><p class=3DMsoNo=
rmal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>With RTP: One can use L4 LB =
distributing traffic to a Redundancy Group. The pain point is that LB is sl=
ow in detecting failures of Redundancy Group members.<o:p></o:p></p><p clas=
s=3DMsoNormal>With RIPT media: One can use HTTP LB distributing traffic a R=
edundancy Group. I would think failure detection characteristics will be si=
milar to L4 LB and delayed failure detection issue would continue to be an =
issue.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DM=
soNormal>In a nutshell, what would one gain by using a HTTP LB instead of L=
4 LB?<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMs=
oNormal>Thanks,<o:p></o:p></p><p class=3DMsoNormal>Tolga<o:p></o:p></p></di=
v><br><br><span style=3D"font-family:Arial; Font-size:8.0pt"> <hr> Notice: =
This e-mail together with any attachments may contain information of Ribbon=
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From: "Asveren, Tolga" <tasveren@rbbn.com>
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To be able to perform RIPT/RTP interworking, I think it is unavoidable that=
 one has to address all the existing difficulties with RTP in Public Cloud.=
 So, this translates to that any potential benefits of RIPT would be mainly=
 for cases not requiring interworking, or?

Thanks,
Tolga


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" vlink=3D"#954F72"><div class=3DWordSection1><p class=3DMsoNormal>To be ab=
le to perform RIPT/RTP interworking, I think it is unavoidable that one has=
 to address all the existing difficulties with RTP in Public Cloud. So, thi=
s translates to that any potential benefits of RIPT would be mainly for cas=
es not requiring interworking, or?<o:p></o:p></p><p class=3DMsoNormal><o:p>=
&nbsp;</o:p></p><p class=3DMsoNormal>Thanks,<o:p></o:p></p><p class=3DMsoNo=
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From: Justin Uberti <juberti@google.com>
Date: Mon, 30 Mar 2020 17:08:41 -0700
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To: "Asveren, Tolga" <tasveren@rbbn.com>
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Subject: Re: [V3] RIPT/RTP interworking
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Interworking with the RTP world would be addressed via RIPT/RTP gateways,
as discussed on the "Gateway new to old" thread.

On Mon, Mar 30, 2020 at 5:06 PM Asveren, Tolga <tasveren@rbbn.com> wrote:

> To be able to perform RIPT/RTP interworking, I think it is unavoidable
> that one has to address all the existing difficulties with RTP in Public
> Cloud. So, this translates to that any potential benefits of RIPT would be
> mainly for cases not requiring interworking, or?
>
>
>
> Thanks,
>
> Tolga
>
>
> ------------------------------
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<div dir=3D"ltr">Interworking with the RTP world would be addressed via RIP=
T/RTP gateways, as discussed on the &quot;Gateway new to old&quot; thread.<=
/div><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">O=
n Mon, Mar 30, 2020 at 5:06 PM Asveren, Tolga &lt;<a href=3D"mailto:tasvere=
n@rbbn.com" target=3D"_blank">tasveren@rbbn.com</a>&gt; wrote:<br></div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-lef=
t:1px solid rgb(204,204,204);padding-left:1ex"><div lang=3D"EN-US"><div><p =
class=3D"MsoNormal">To be able to perform RIPT/RTP interworking, I think it=
 is unavoidable that one has to address all the existing difficulties with =
RTP in Public Cloud. So, this translates to that any potential benefits of =
RIPT would be mainly for cases not requiring interworking, or?<u></u><u></u=
></p><p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=3D"MsoNormal">=
Thanks,<u></u><u></u></p><p class=3D"MsoNormal">Tolga<u></u><u></u></p></di=
v><br><br><span style=3D"font-family:Arial;font-size:8pt"> <hr> Notice: Thi=
s e-mail together with any attachments may contain information of Ribbon Co=
mmunications Inc. that is confidential and/or proprietary for the sole use =
of the intended recipient.  Any review, disclosure, reliance or distributio=
n by others or forwarding without express permission is strictly prohibited=
.  If you are not the intended recipient, please notify the sender immediat=
ely and then delete all copies, including any attachments.<hr> </span></div=
>
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To: Justin Uberti <juberti@google.com>
CC: "v3@ietf.org" <v3@ietf.org>
Date: Mon, 30 Mar 2020 20:21:07 -0400
Thread-Topic: [V3] RIPT/RTP interworking
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Subject: Re: [V3] RIPT/RTP interworking
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "Asveren, Tolga" <tasveren@rbbn.com>, "v3@ietf.org" <v3@ietf.org>
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Subject: Re: [V3] LB / Failure Detection
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--_000_DAA98BE896CBEmzanatyciscocom_
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> In a nutshell, what would one gain by using a HTTP LB instead of L4 LB?

There are many reasons why HTTP apps use HTTP LB instead of (or in addition=
 to) L4 LB, such as flexibility, programmability, feature rich ecosystem, e=
tc. The HTTP list may be a better place for such a question and discussion =
though, since this is not specific to RIPT, but applies to all HTTP apps.

Thanks,
Mo

On 3/30/20, 7:19 PM, "V3 on behalf of Asveren, Tolga" <v3-bounces@ietf.org<=
mailto:v3-bounces@ietf.org> on behalf of tasveren@rbbn.com<mailto:tasveren@=
rbbn.com>> wrote:

A basic question about how RIPT would make things better compared to RTP in=
 terms of making use of Public Cloud Load-balancers:

With RTP: One can use L4 LB distributing traffic to a Redundancy Group. The=
 pain point is that LB is slow in detecting failures of Redundancy Group me=
mbers.
With RIPT media: One can use HTTP LB distributing traffic a Redundancy Grou=
p. I would think failure detection characteristics will be similar to L4 LB=
 and delayed failure detection issue would continue to be an issue.

In a nutshell, what would one gain by using a HTTP LB instead of L4 LB?

Thanks,
Tolga


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<div>
<p class=3D"MsoNormal">&gt; In a nutshell, what would one gain by using a H=
TTP LB instead of L4 LB?</p>
</div>
<div><br>
</div>
<div>There are many reasons why HTTP apps use HTTP LB instead of (or in add=
ition to) L4 LB, such as flexibility, programmability, feature rich ecosyst=
em, etc. The HTTP list may be a better place for such a question and discus=
sion though, since this is not specific
 to RIPT, but applies to all HTTP apps.</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Mo</div>
<div><br>
</div>
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<div>On 3/30/20, 7:19 PM, &quot;V3 on behalf of Asveren, Tolga&quot; &lt;<a=
 href=3D"mailto:v3-bounces@ietf.org">v3-bounces@ietf.org</a> on behalf of
<a href=3D"mailto:tasveren@rbbn.com">tasveren@rbbn.com</a>&gt; wrote:</div>
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<p class=3D"MsoNormal">A basic question about how RIPT would make things be=
tter compared to RTP in terms of making use of Public Cloud Load-balancers:=
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">With RTP: One can use L4 LB distributing traffic to =
a Redundancy Group. The pain point is that LB is slow in detecting failures=
 of Redundancy Group members.<o:p></o:p></p>
<p class=3D"MsoNormal">With RIPT media: One can use HTTP LB distributing tr=
affic a Redundancy Group. I would think failure detection characteristics w=
ill be similar to L4 LB and delayed failure detection issue would continue =
to be an issue.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In a nutshell, what would one gain by using a HTTP L=
B instead of L4 LB?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
<p class=3D"MsoNormal">Tolga<o:p></o:p></p>
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