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From: Tim Panton <thp@westhawk.co.uk>
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Subject: [rtcweb] ICE exposes 'real' local IP to javascript
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Firstly- sorry for cross posting - I=E2=80=99m not sure which side of =
the line this falls.
Secondly - if this is covered, please let me know, I don=E2=80=99t =
recall it cropping up...

I=E2=80=99ve been reading worried blogs that WEBRTC in browsers =
=E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=80=99 ip addresses to =
the javascript.
The principle worriers are VPN users e.g =
https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096 =
<https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096>
The concern is that this can be done without user notification =
(DataChannel request) and might be used to=20
identify or finger-print users. Clearly the most vulnerable are Tor =
users who are on a real routeable IP address
or directly on a carrier grade nat (eg android phones etc) where the IP =
may reveal the identity or location of the user.

It seems to me that this concern will be increased in the case of ipv6 =
deployments (MNOs).

Do we need to specify a config option on the browser =E2=80=98I=E2=80=99m =
using a VPN don=E2=80=99t expose my local IP=E2=80=99=20

Again, sorry if I missed this being hashed to death already.=20

T

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




--Apple-Mail=_6F972E43-2EA1-41BC-9BC9-6062EA970ABF
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
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charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><div class=3D"">Firstly- sorry for cross posting - I=E2=80=99m =
not sure which side of the line this falls.</div><div class=3D"">Secondly =
- if this is covered, please let me know, I don=E2=80=99t recall it =
cropping up...</div><div class=3D""><br class=3D""></div><div =
class=3D"">I=E2=80=99ve been reading worried blogs that WEBRTC in =
browsers =E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=80=99 ip =
addresses to the javascript.</div><div class=3D"">The principle worriers =
are VPN users e.g <a =
href=3D"https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=3D=
13096#p13096" =
class=3D"">https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=
=3D13096#p13096</a></div><div class=3D"">The concern is that this can be =
done without user notification (DataChannel request) and might be used =
to&nbsp;</div><div class=3D"">identify or finger-print users. Clearly =
the most vulnerable are Tor users who are on a real routeable IP =
address</div><div class=3D"">or directly on a carrier grade nat (eg =
android phones etc) where the IP may reveal the identity or location of =
the user.</div><div class=3D""><br class=3D""></div><div class=3D"">It =
seems to me that this concern will be increased in the case of ipv6 =
deployments (MNOs).</div><div class=3D""><br class=3D""></div><div =
class=3D"">Do we need to specify a config option on the browser =E2=80=98I=
=E2=80=99m using a VPN don=E2=80=99t expose my local =
IP=E2=80=99&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">Again, sorry if I missed this being hashed to death =
already.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">T</div><br class=3D""><div apple-content-edited=3D"true" =
class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
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-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div class=3D"">Tim Panton - =
Web/VoIP consultant and implementor</div><div class=3D""><a =
href=3D"http://www.westhawk.co.uk" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></body></html>=

--Apple-Mail=_6F972E43-2EA1-41BC-9BC9-6062EA970ABF--


From nobody Mon Feb  2 09:05:13 2015
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From: Tim Panton <thp@westhawk.co.uk>
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Subject: [rtcweb] ICE exposes 'real' local IP to javascript
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--Apple-Mail=_8F081823-C453-4A5C-BFA6-2CE530841D1A
Content-Transfer-Encoding: quoted-printable
Content-Type: text/plain;
	charset=utf-8

Firstly- sorry for cross posting (to w3c webrtc) - I=E2=80=99m not sure =
which side of the line this falls.
Secondly - if this is covered, please let me know, I don=E2=80=99t =
recall it cropping up...

I=E2=80=99ve been reading worried blogs that WEBRTC in browsers =
=E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=80=99 ip addresses to =
the javascript.
The principle worriers are VPN users e.g =
https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096 =
<https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096>
The concern is that this can be done without user notification =
(DataChannel request) and might be used to=20
identify or finger-print users. Clearly the most vulnerable are Tor =
users who are on a real routeable IP address
or directly on a carrier grade nat (eg android phones etc) where the IP =
may reveal the identity or location of the user.

It seems to me that this concern will be increased in the case of ipv6 =
deployments (MNOs).

Do we need to specify a config option on the browser =E2=80=98I=E2=80=99m =
using a VPN don=E2=80=99t expose my local IP=E2=80=99=20

Again, sorry if I missed this being hashed to death already.=20

T

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




--Apple-Mail=_8F081823-C453-4A5C-BFA6-2CE530841D1A
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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; -webkit-line-break: after-white-space;" =
class=3D""><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8" class=3D""><meta http-equiv=3D"Content-Type" =
content=3D"text/html charset=3Dutf-8" class=3D""><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;" class=3D""><div class=3D"">Firstly- sorry for cross =
posting (to w3c webrtc) - I=E2=80=99m not sure which side of the line =
this falls.</div><div class=3D"">Secondly - if this is covered, please =
let me know, I don=E2=80=99t recall it cropping up...</div><div =
class=3D""><br class=3D""></div><div class=3D"">I=E2=80=99ve been =
reading worried blogs that WEBRTC in browsers =E2=80=98leaks=E2=80=99 =
the local =E2=80=98real=E2=80=99 ip addresses to the =
javascript.</div><div class=3D"">The principle worriers are VPN users =
e.g <a =
href=3D"https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=3D=
13096#p13096" =
class=3D"">https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=
=3D13096#p13096</a></div><div class=3D"">The concern is that this can be =
done without user notification (DataChannel request) and might be used =
to&nbsp;</div><div class=3D"">identify or finger-print users. Clearly =
the most vulnerable are Tor users who are on a real routeable IP =
address</div><div class=3D"">or directly on a carrier grade nat (eg =
android phones etc) where the IP may reveal the identity or location of =
the user.</div><div class=3D""><br class=3D""></div><div class=3D"">It =
seems to me that this concern will be increased in the case of ipv6 =
deployments (MNOs).</div><div class=3D""><br class=3D""></div><div =
class=3D"">Do we need to specify a config option on the browser =E2=80=98I=
=E2=80=99m using a VPN don=E2=80=99t expose my local =
IP=E2=80=99&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">Again, sorry if I missed this being hashed to death =
already.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D"">T</div><br class=3D""><div apple-content-edited=3D"true" =
class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
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white-space: normal; widows: 2; word-spacing: 0px; border-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-stroke-width: =
0px;"><div class=3D"">Tim Panton - Web/VoIP consultant and =
implementor</div><div class=3D""><a href=3D"http://www.westhawk.co.uk/" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></div></body></html>=

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Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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Standards-wise: You might want to have a look at
http://tools.ietf.org/html/draft-schwartz-rtcweb-return-04#section-5.3 (a
draft which I'm hoping will be adopted by the rtcweb group).

Reality-wise:
Tor is not a VPN.  It acts as a SOCKS5 proxy.  Tor doesn't support UDP, and
none of the major browsers support SOCKS5-UDP anyway, so it's not much use
for WebRTC.  Tor Browser Bundle, IMHO the only responsible way to use Tor,
has disabled WebRTC from the beginning, precisely to avoid revealing the
user's IP address.

VPN users who want to be safe should set permissions such that the browser
can only access the VPN, not the physical network.  (I don't personally
know how to do this, especially on all different operating systems!)

On Mon, Feb 2, 2015 at 9:16 AM, Tim Panton <thp@westhawk.co.uk> wrote:

> Firstly- sorry for cross posting - I=E2=80=99m not sure which side of the=
 line
> this falls.
> Secondly - if this is covered, please let me know, I don=E2=80=99t recall=
 it
> cropping up...
>
> I=E2=80=99ve been reading worried blogs that WEBRTC in browsers =E2=80=98=
leaks=E2=80=99 the local
> =E2=80=98real=E2=80=99 ip addresses to the javascript.
> The principle worriers are VPN users e.g
> https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096
> The concern is that this can be done without user notification
> (DataChannel request) and might be used to
> identify or finger-print users. Clearly the most vulnerable are Tor users
> who are on a real routeable IP address
> or directly on a carrier grade nat (eg android phones etc) where the IP
> may reveal the identity or location of the user.
>
> It seems to me that this concern will be increased in the case of ipv6
> deployments (MNOs).
>
> Do we need to specify a config option on the browser =E2=80=98I=E2=80=99m=
 using a VPN
> don=E2=80=99t expose my local IP=E2=80=99
>
> Again, sorry if I missed this being hashed to death already.
>
> T
>
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
>
>
>
>

--089e01634ccade239c050e1c2ea0
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<div dir=3D"ltr">Standards-wise: You might want to have a look at=C2=A0<a h=
ref=3D"http://tools.ietf.org/html/draft-schwartz-rtcweb-return-04#section-5=
.3">http://tools.ietf.org/html/draft-schwartz-rtcweb-return-04#section-5.3<=
/a> (a draft which I&#39;m hoping will be adopted by the rtcweb group).<div=
><br></div><div>Reality-wise:</div><div>Tor is not a VPN.=C2=A0 It acts as =
a SOCKS5 proxy.=C2=A0 Tor doesn&#39;t support UDP, and none of the major br=
owsers support SOCKS5-UDP anyway, so it&#39;s not much use for WebRTC.=C2=
=A0 Tor Browser Bundle, IMHO the only responsible way to use Tor, has disab=
led WebRTC from the beginning, precisely to avoid revealing the user&#39;s =
IP address.</div><div><br></div><div>VPN users who want to be safe should s=
et permissions such that the browser can only access the VPN, not the physi=
cal network. =C2=A0(I don&#39;t personally know how to do this, especially =
on all different operating systems!)</div></div><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote">On Mon, Feb 2, 2015 at 9:16 AM, Tim Panton <=
span dir=3D"ltr">&lt;<a href=3D"mailto:thp@westhawk.co.uk" target=3D"_blank=
">thp@westhawk.co.uk</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
"><div style=3D"word-wrap:break-word"><div>Firstly- sorry for cross posting=
 - I=E2=80=99m not sure which side of the line this falls.</div><div>Second=
ly - if this is covered, please let me know, I don=E2=80=99t recall it crop=
ping up...</div><div><br></div><div>I=E2=80=99ve been reading worried blogs=
 that WEBRTC in browsers =E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=
=80=99 ip addresses to the javascript.</div><div>The principle worriers are=
 VPN users e.g <a href=3D"https://cryptostorm.org/viewtopic.php?f=3D50&amp;=
t=3D2867&amp;p=3D13096#p13096" target=3D"_blank">https://cryptostorm.org/vi=
ewtopic.php?f=3D50&amp;t=3D2867&amp;p=3D13096#p13096</a></div><div>The conc=
ern is that this can be done without user notification (DataChannel request=
) and might be used to=C2=A0</div><div>identify or finger-print users. Clea=
rly the most vulnerable are Tor users who are on a real routeable IP addres=
s</div><div>or directly on a carrier grade nat (eg android phones etc) wher=
e the IP may reveal the identity or location of the user.</div><div><br></d=
iv><div>It seems to me that this concern will be increased in the case of i=
pv6 deployments (MNOs).</div><div><br></div><div>Do we need to specify a co=
nfig option on the browser =E2=80=98I=E2=80=99m using a VPN don=E2=80=99t e=
xpose my local IP=E2=80=99=C2=A0</div><div><br></div><div>Again, sorry if I=
 missed this being hashed to death already.=C2=A0</div><div><br></div><div>=
T</div><br><div>
<span style=3D"border-collapse:separate;color:rgb(0,0,0);font-family:Helvet=
ica;font-size:12px;font-style:normal;font-variant:normal;font-weight:normal=
;letter-spacing:normal;line-height:normal;text-align:auto;text-indent:0px;t=
ext-transform:none;white-space:normal;word-spacing:0px"><div>Tim Panton - W=
eb/VoIP consultant and implementor</div><div><a href=3D"http://www.westhawk=
.co.uk" target=3D"_blank">www.westhawk.co.uk</a></div><div><br></div></span=
><br>

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

--089e01634ccade239c050e1c2ea0--


From nobody Mon Feb  2 12:00:37 2015
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From: <richard.vandet@kpn.com>
To: <bemasc@google.com>
Thread-Topic: [rtcweb] ICE exposes 'real' local IP to javascript
Thread-Index: AQHQPwp9s87XqwGewUumvP+AnHU1QpzdYnqAgABkkww=
Date: Mon, 2 Feb 2015 19:58:09 +0000
Message-ID: <715F96D7-0BCA-4C91-BDEC-9DF28FD9A484@kpn.com>
References: <5B986D58-AB56-4976-8F61-4E80110916A2@westhawk.co.uk>, <CAHbrMsAydrfvHc22tOP2pk0nEiN4hrTwmFEfUFLq7fuWyXhgfA@mail.gmail.com>
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Cc: rtcweb@ietf.org, thp@westhawk.co.uk, public-webrtc@w3.org
Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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--_000_715F96D70BCA4C91BDEC9DF28FD9A484kpncom_
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Isn't that done by whitelisting. You as a user use a application wich uses =
webrtc.

M.vr.gr<http://M.vr.gr>.,

Richard van Det
06 2054 7291

Op 2 feb. 2015 om 18:05 heeft Benjamin Schwartz <bemasc@google.com<mailto:b=
emasc@google.com>> het volgende geschreven:

Standards-wise: You might want to have a look at http://tools.ietf.org/html=
/draft-schwartz-rtcweb-return-04#section-5.3 (a draft which I'm hoping will=
 be adopted by the rtcweb group).

Reality-wise:
Tor is not a VPN.  It acts as a SOCKS5 proxy.  Tor doesn't support UDP, and=
 none of the major browsers support SOCKS5-UDP anyway, so it's not much use=
 for WebRTC.  Tor Browser Bundle, IMHO the only responsible way to use Tor,=
 has disabled WebRTC from the beginning, precisely to avoid revealing the u=
ser's IP address.

VPN users who want to be safe should set permissions such that the browser =
can only access the VPN, not the physical network.  (I don't personally kno=
w how to do this, especially on all different operating systems!)

On Mon, Feb 2, 2015 at 9:16 AM, Tim Panton <thp@westhawk.co.uk<mailto:thp@w=
esthawk.co.uk>> wrote:
Firstly- sorry for cross posting - I=92m not sure which side of the line th=
is falls.
Secondly - if this is covered, please let me know, I don=92t recall it crop=
ping up...

I=92ve been reading worried blogs that WEBRTC in browsers =91leaks=92 the l=
ocal =91real=92 ip addresses to the javascript.
The principle worriers are VPN users e.g https://cryptostorm.org/viewtopic.=
php?f=3D50&t=3D2867&p=3D13096#p13096
The concern is that this can be done without user notification (DataChannel=
 request) and might be used to
identify or finger-print users. Clearly the most vulnerable are Tor users w=
ho are on a real routeable IP address
or directly on a carrier grade nat (eg android phones etc) where the IP may=
 reveal the identity or location of the user.

It seems to me that this concern will be increased in the case of ipv6 depl=
oyments (MNOs).

Do we need to specify a config option on the browser =91I=92m using a VPN d=
on=92t expose my local IP=92

Again, sorry if I missed this being hashed to death already.

T

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk<http://www.westhawk.co.uk>




_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

--_000_715F96D70BCA4C91BDEC9DF28FD9A484kpncom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body dir=3D"auto">
<div>Isn't that done by whitelisting. You as a user use a application wich =
uses webrtc.<br>
<br>
<a href=3D"http://M.vr.gr">M.vr.gr</a>.,
<div><br>
</div>
<div>Richard van Det</div>
<div>06 2054 7291</div>
</div>
<div><br>
Op 2 feb. 2015 om 18:05 heeft Benjamin Schwartz &lt;<a href=3D"mailto:bemas=
c@google.com">bemasc@google.com</a>&gt; het volgende geschreven:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">Standards-wise: You might want to have a look at&nbsp;<a h=
ref=3D"http://tools.ietf.org/html/draft-schwartz-rtcweb-return-04#section-5=
.3">http://tools.ietf.org/html/draft-schwartz-rtcweb-return-04#section-5.3<=
/a> (a draft which I'm hoping will be adopted
 by the rtcweb group).
<div><br>
</div>
<div>Reality-wise:</div>
<div>Tor is not a VPN.&nbsp; It acts as a SOCKS5 proxy.&nbsp; Tor doesn't s=
upport UDP, and none of the major browsers support SOCKS5-UDP anyway, so it=
's not much use for WebRTC.&nbsp; Tor Browser Bundle, IMHO the only respons=
ible way to use Tor, has disabled WebRTC from the
 beginning, precisely to avoid revealing the user's IP address.</div>
<div><br>
</div>
<div>VPN users who want to be safe should set permissions such that the bro=
wser can only access the VPN, not the physical network. &nbsp;(I don't pers=
onally know how to do this, especially on all different operating systems!)=
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Mon, Feb 2, 2015 at 9:16 AM, Tim Panton <span=
 dir=3D"ltr">
&lt;<a href=3D"mailto:thp@westhawk.co.uk" target=3D"_blank">thp@westhawk.co=
.uk</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div>Firstly- sorry for cross posting - I=92m not sure which side of the li=
ne this falls.</div>
<div>Secondly - if this is covered, please let me know, I don=92t recall it=
 cropping up...</div>
<div><br>
</div>
<div>I=92ve been reading worried blogs that WEBRTC in browsers =91leaks=92 =
the local =91real=92 ip addresses to the javascript.</div>
<div>The principle worriers are VPN users e.g <a href=3D"https://cryptostor=
m.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=3D13096#p13096" target=3D"_bl=
ank">
https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=3D13096#p13=
096</a></div>
<div>The concern is that this can be done without user notification (DataCh=
annel request) and might be used to&nbsp;</div>
<div>identify or finger-print users. Clearly the most vulnerable are Tor us=
ers who are on a real routeable IP address</div>
<div>or directly on a carrier grade nat (eg android phones etc) where the I=
P may reveal the identity or location of the user.</div>
<div><br>
</div>
<div>It seems to me that this concern will be increased in the case of ipv6=
 deployments (MNOs).</div>
<div><br>
</div>
<div>Do we need to specify a config option on the browser =91I=92m using a =
VPN don=92t expose my local IP=92&nbsp;</div>
<div><br>
</div>
<div>Again, sorry if I missed this being hashed to death already.&nbsp;</di=
v>
<div><br>
</div>
<div>T</div>
<br>
<div><span style=3D"border-collapse:separate;color:rgb(0,0,0);font-family:H=
elvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:n=
ormal;letter-spacing:normal;line-height:normal;text-align:auto;text-indent:=
0px;text-transform:none;white-space:normal;word-spacing:0px">
<div>Tim Panton - Web/VoIP consultant and implementor</div>
<div><a href=3D"http://www.westhawk.co.uk" target=3D"_blank">www.westhawk.c=
o.uk</a></div>
<div><br>
</div>
</span><br>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_715F96D70BCA4C91BDEC9DF28FD9A484kpncom_--


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To: Tim Panton <thp@westhawk.co.uk>, public-webrtc <public-webrtc@w3.org>
Thread-Topic: ICE exposes 'real' local IP to javascript
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Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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From nobody Tue Feb  3 06:03:56 2015
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From: Bjoern Hoehrmann <derhoermi@gmx.net>
To: Tim Panton <thp@westhawk.co.uk>
Date: Tue, 03 Feb 2015 15:03:43 +0100
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Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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* Tim Panton wrote:
>I’ve been reading worried blogs that WEBRTC in browsers ‘leaks’ the local ‘real’ ip addresses to the javascript.
>The principle worriers are VPN users e.g https://cryptostorm.org/viewtopic.php?f=50&t=2867&p=13096#p13096 <https://cryptostorm.org/viewtopic.php?f=50&t=2867&p=13096#p13096>
>The concern is that this can be done without user notification (DataChannel request) and might be used to 
>identify or finger-print users. Clearly the most vulnerable are Tor users who are on a real routeable IP address
>or directly on a carrier grade nat (eg android phones etc) where the IP may reveal the identity or location of the user.

If this actually allows random web sites to know that I am using the
addresses 192.168.12.96, 192.168.3.222, and 192.168.200.7 (Ethernet,
Wifi, virtual machine) on this computer then that is pretty much like
broadcasting a device GUID and should concern everybody. I note that
some of the recent news coverage of this indicated that some major web
sites and services already use this to identify and track users.
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
D-10243 Berlin · PGP Pub. KeyID: 0xA4357E78 · http://www.bjoernsworld.de
 Available for hire in Berlin (early 2015)  · http://www.websitedev.de/ 


From nobody Tue Feb  3 07:29:10 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Data Channel Closure: Is sctp-reset mandatory?
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Hi,

When I am about to close the data channel, if I also want to tear down the =
SCTP-over-DTLS association, do I first need to send sctp-reset (in order to=
 close my data channel(s)), before sending SHUTDOWN? Or, can I simply send =
SHUTDOWN?

Regards,

Christer

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	font-family:"Calibri",sans-serif;
	color:windowtext;}
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	{mso-style-type:export-only;
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	mso-fareast-language:EN-US;}
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	margin:72.0pt 72.0pt 72.0pt 72.0pt;}
div.WordSection1
	{page:WordSection1;}
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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
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<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
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<body lang=3D"EN-GB" link=3D"#0563C1" vlink=3D"#954F72">
<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">When I am about to close the data channel, if I also=
 want to tear down the SCTP-over-DTLS association, do I first need to send =
sctp-reset (in order to close my data channel(s)), before sending SHUTDOWN?=
 Or, can I simply send SHUTDOWN?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D68E833ESESSMB209erics_--


From nobody Tue Feb  3 08:42:14 2015
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Date: Tue, 03 Feb 2015 17:42:01 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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Some thoughts....

1) Datachannel is a red herring. There are many ways to do a valid 
CreateOffer with m-lines, which is all that is required:

- Datachannel
- OfferToReceiveVideo  / Audio
- Generate a MediaStreamTrack from WebAudio

and so on.

2) Speaking with my WebRTC hat on: IP addresses have to be surfaced at 
the API as long as the other side needs to try to send packets to these 
interfaces. We can't obfuscate them or encrypt them because they have to 
be communicated to the other party, through channels that aren't in the 
WebRTC spec.

3) Speaking with my (imaginary) implementors hat: One can imagine a 
(browser-wide) configuration setting for which addresses to allow access 
to, possibly with a whitelist of apps / pages / sites allowed more 
access than others. Normal people will never configure this (and if they 
tried, they would get it wrong), so the defaults need to be "safe enough 
for most", but sysadmins and the people with special reasons to care 
about security might.

4) Again wearing my WebRTC spec shepherding hat: It seems that the spec 
should make it clear:
a) that IP addresses will be exposed (in SDP and in oncandidate callbacks)
b) why IP addresses are being exposed
but not really anything more than that.

5) Wearing my forum-shuffling hat, it seems that the "why" part belongs 
more in the IETF than in the WebRTC side of things - so I'd favour 
continuing this thread on rtcweb@ietf only....

Harald




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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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--001a1147d2d0c302f5050e3238c8
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The thing I was wondering about was, should there be a confirmation dialog
when browser tries to setup any type of peer-to-peer connection? We get a
confirmation dialog when microphone or camera access is requested. I think
setting up a peer-to-peer connection is something that should be controlled
by the user on the per web site basis in the similar manner.

_____________
Roman Shpount

On Tue, Feb 3, 2015 at 11:42 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> Some thoughts....
>
> 1) Datachannel is a red herring. There are many ways to do a valid
> CreateOffer with m-lines, which is all that is required:
>
> - Datachannel
> - OfferToReceiveVideo  / Audio
> - Generate a MediaStreamTrack from WebAudio
>
> and so on.
>
> 2) Speaking with my WebRTC hat on: IP addresses have to be surfaced at the
> API as long as the other side needs to try to send packets to these
> interfaces. We can't obfuscate them or encrypt them because they have to be
> communicated to the other party, through channels that aren't in the WebRTC
> spec.
>
> 3) Speaking with my (imaginary) implementors hat: One can imagine a
> (browser-wide) configuration setting for which addresses to allow access
> to, possibly with a whitelist of apps / pages / sites allowed more access
> than others. Normal people will never configure this (and if they tried,
> they would get it wrong), so the defaults need to be "safe enough for
> most", but sysadmins and the people with special reasons to care about
> security might.
>
> 4) Again wearing my WebRTC spec shepherding hat: It seems that the spec
> should make it clear:
> a) that IP addresses will be exposed (in SDP and in oncandidate callbacks)
> b) why IP addresses are being exposed
> but not really anything more than that.
>
> 5) Wearing my forum-shuffling hat, it seems that the "why" part belongs
> more in the IETF than in the WebRTC side of things - so I'd favour
> continuing this thread on rtcweb@ietf only....
>
> Harald
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a1147d2d0c302f5050e3238c8
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">The thing I was wondering about was, should there be a con=
firmation dialog when browser tries to setup any type of peer-to-peer conne=
ction? We get a confirmation dialog when microphone or camera access is req=
uested. I think setting up a peer-to-peer connection is something that shou=
ld be controlled by the user on the per web site basis in the similar manne=
r.</div><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gma=
il_signature">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Tue, Feb 3, 2015 at 11:42 AM, Harald Alve=
strand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=
=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">Some thoughts....<br>
<br>
1) Datachannel is a red herring. There are many ways to do a valid CreateOf=
fer with m-lines, which is all that is required:<br>
<br>
- Datachannel<br>
- OfferToReceiveVideo=C2=A0 / Audio<br>
- Generate a MediaStreamTrack from WebAudio<br>
<br>
and so on.<br>
<br>
2) Speaking with my WebRTC hat on: IP addresses have to be surfaced at the =
API as long as the other side needs to try to send packets to these interfa=
ces. We can&#39;t obfuscate them or encrypt them because they have to be co=
mmunicated to the other party, through channels that aren&#39;t in the WebR=
TC spec.<br>
<br>
3) Speaking with my (imaginary) implementors hat: One can imagine a (browse=
r-wide) configuration setting for which addresses to allow access to, possi=
bly with a whitelist of apps / pages / sites allowed more access than other=
s. Normal people will never configure this (and if they tried, they would g=
et it wrong), so the defaults need to be &quot;safe enough for most&quot;, =
but sysadmins and the people with special reasons to care about security mi=
ght.<br>
<br>
4) Again wearing my WebRTC spec shepherding hat: It seems that the spec sho=
uld make it clear:<br>
a) that IP addresses will be exposed (in SDP and in oncandidate callbacks)<=
br>
b) why IP addresses are being exposed<br>
but not really anything more than that.<br>
<br>
5) Wearing my forum-shuffling hat, it seems that the &quot;why&quot; part b=
elongs more in the IETF than in the WebRTC side of things - so I&#39;d favo=
ur continuing this thread on rtcweb@ietf only....<span class=3D"HOEnZb"><fo=
nt color=3D"#888888"><br>
<br>
Harald</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--001a1147d2d0c302f5050e3238c8--


From nobody Tue Feb  3 09:28:55 2015
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
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> On 03 Feb 2015, at 16:28, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi,
> =20
> When I am about to close the data channel, if I also want to tear down =
the SCTP-over-DTLS association, do I first need to send sctp-reset (in =
order to close my data channel(s)), before sending SHUTDOWN? Or, can I =
simply send SHUTDOWN?
You can simply shutdown. See the last paragraph in
=
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
Best regards
Michael
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Feb  3 09:35:13 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
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Date: Tue, 3 Feb 2015 17:35:02 +0000
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Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
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Thanks!

-----Original Message-----
From: Michael Tuexen [mailto:Michael.Tuexen@lurchi.franken.de]=20
Sent: 03 February 2015 19:29
To: Christer Holmberg
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?

> On 03 Feb 2015, at 16:28, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:
>=20
> Hi,
> =20
> When I am about to close the data channel, if I also want to tear down th=
e SCTP-over-DTLS association, do I first need to send sctp-reset (in order =
to close my data channel(s)), before sending SHUTDOWN? Or, can I simply sen=
d SHUTDOWN?
You can simply shutdown. See the last paragraph in
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
Best regards
Michael
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Feb  3 10:36:26 2015
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
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Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
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Michael,

thanks for answering the case of the termination of the SCTP association in=
 outgoing direction.
What about the incoming direction?
How should I react in case of an incoming SCTP RE-CONFIG chunk (with a SCTP=
 stream reset)?
a) may I immediately send a SCTP SHUTDOWN chunk?
or
b) firstly replying the RE-CONFIG procedure and subsequently initiating the=
 SCTP association shutdwon with a SCTP SHUTDOWN chunk?

The distinction between incoming and outgoing direction isn't made in https=
://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2 in my =
understanding.

Thanks,
Albrecht

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmber=
g
Sent: Dienstag, 3. Februar 2015 18:35
To: Michael Tuexen
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?

Thanks!

-----Original Message-----
From: Michael Tuexen [mailto:Michael.Tuexen@lurchi.franken.de]=20
Sent: 03 February 2015 19:29
To: Christer Holmberg
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?

> On 03 Feb 2015, at 16:28, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:
>=20
> Hi,
> =20
> When I am about to close the data channel, if I also want to tear down th=
e SCTP-over-DTLS association, do I first need to send sctp-reset (in order =
to close my data channel(s)), before sending SHUTDOWN? Or, can I simply sen=
d SHUTDOWN?
You can simply shutdown. See the last paragraph in
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
Best regards
Michael
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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


From nobody Tue Feb  3 12:15:32 2015
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Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
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> On 03 Feb 2015, at 19:36, Schwarz, Albrecht (Albrecht) =
<albrecht.schwarz@ALCATEL-LUCENT.COM> wrote:
>=20
> Michael,
>=20
> thanks for answering the case of the termination of the SCTP =
association in outgoing direction.
> What about the incoming direction?
> How should I react in case of an incoming SCTP RE-CONFIG chunk (with a =
SCTP stream reset)?
> a) may I immediately send a SCTP SHUTDOWN chunk?
> or
> b) firstly replying the RE-CONFIG procedure and subsequently =
initiating the SCTP association shutdwon with a SCTP SHUTDOWN chunk?
>=20
> The distinction between incoming and outgoing direction isn't made in =
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2 =
in my understanding.
No it isn't. If you want to turn down the association, but can do that =
at any time. This
is at least my understanding when I put the text into the document.

Best regards
Michael
>=20
> Thanks,
> Albrecht
>=20
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer =
Holmberg
> Sent: Dienstag, 3. Februar 2015 18:35
> To: Michael Tuexen
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
>=20
> Thanks!
>=20
> -----Original Message-----
> From: Michael Tuexen [mailto:Michael.Tuexen@lurchi.franken.de]=20
> Sent: 03 February 2015 19:29
> To: Christer Holmberg
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Data Channel Closure: Is sctp-reset mandatory?
>=20
>> On 03 Feb 2015, at 16:28, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>=20
>> Hi,
>>=20
>> When I am about to close the data channel, if I also want to tear =
down the SCTP-over-DTLS association, do I first need to send sctp-reset =
(in order to close my data channel(s)), before sending SHUTDOWN? Or, can =
I simply send SHUTDOWN?
> You can simply shutdown. See the last paragraph in
> =
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
> Best regards
> Michael
>>=20
>> Regards,
>>=20
>> Christer
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From nobody Tue Feb  3 13:03:25 2015
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From: Martin Thomson <martin.thomson@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] DTLS requirements
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The current security architecture mandates *both* DTLS 1.0 and 1.2.
At the last meeting we agreed MUST 1.0, SHOULD 1.2 [1].

Also, I think that the only elliptic curve feature that is risky is
ECDSA, not ECDHE.  I'm going to propose that we move to ECDHE.

    https://github.com/rtcweb-wg/security-arch/pull/19

In addition to that change, I would also like to propose that we
*require* a forward-secrecy-capable cipher suite.

    https://github.com/rtcweb-wg/security-arch/pull/20

[1] https://tools.ietf.org/wg/rtcweb/minutes?item=minutes-91-rtcweb.html


From nobody Wed Feb  4 02:40:39 2015
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Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
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Is it too late to add some explicit text indicating TCP rather than 
having to reply on another draft? It's easily missed by people reading 
the draft and given the real estate given in the data channel draft its 
possible to come to another conclusion.

Christian

On 28/01/2015 5:15 AM, Michael Tuexen wrote:
>> On 27 Jan 2015, at 18:44, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>
>> Hi,
>>   
>>> Well, if you are doing ICE-TCP, I would expect you to run >DataChannels over that. ISTM that that suggests that the marker >should be TCP/DTLS/SCTP.
>>   
>> I am not arguing against that. I just wonder why the data channel draft doesn’t say anything about TCP, while explicitly talking about UDP.
> That is correct. However,
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-09
> says in the Abstract:
>     Using the
>     encapsulation method described in this document, SCTP is unaware of
>     the protocols being used below DTLS;
> So it is clear that you can use other protocols like ICE/TCP. However, this
> is not explicitly discussed and using TCP has the known drawback of running
> a CC (in SCTP) over a CC (in TCP).
>
> I guess the reason why TCP is not mentioned explicitly is that noone brought
> the issue up. It was always meant that you run over ICE no matter what ICE is running over...
>
> Best regards
> Michael
>>   
>> Regards,
>>   
>> Christer
>>   
>>   
>>   
>> On Tue, Jan 27, 2015 at 4:18 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>> Hi,
>>   
>> Related to this, when I read the data channel draft, it explicitly talks about UDP. TCP is not even mentioned.
>>   
>> So, does that mean that TCP/DTLS/SCTP is not “officially” a part of the data channel spec?
>>   
>> Regards,
>>   
>> Christer
>>   
>> From: Makaraju, Maridi Raju (Raju) [mailto:Raju.Makaraju@alcatel-lucent.com]
>> Sent: 24. tammikuuta 2015 15:35
>> To: Christer Holmberg; Justin Uberti
>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>   
>> I am ok to use UDP/DTLS/SCTP and TCP/DTLS/SCTP.
>> I agree that use of RFC #s is not the best option.
>> However, I find some unease in the discussions about “TCP/DTLS” part, which seems to suggest why not use “TCP/TLS”?  We understand that it can’t be called that way because of the RFC 4571 “shim layer” in between DTLS and TCP layers. Unfortunately, unlike TPKT, RFC 4571 did not give a name to the protocol, which would have been easy to use than the RFC directly.
>>   
>> BR
>> Raju
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> Sent: Friday, January 23, 2015 1:22 AM
>> To: Justin Uberti; Makaraju, Maridi Raju (Raju)
>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>   
>> Hi,
>>   
>> I agree that we should not use RFC numbers in proto values.
>>   
>> Also keep in mind that UDP/DTLS/SCTP does not mean “ICE based”. ICE is optional for UDP/DTLS/SCTP (that fact that we mandate ICE for RTCWEB is a separate issue).
>>   
>> Regards,
>>   
>> Christer
>>   
>> From: Justin Uberti [mailto:juberti@google.com]
>> Sent: 23. tammikuuta 2015 5:57
>> To: Makaraju, Maridi Raju (Raju)
>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>   
>> No, I don't think including RFC4571 is reasonable. That ship has already sailed.
>>   
>> On Thu, Jan 22, 2015 at 3:35 PM, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@alcatel-lucent.com> wrote:
>> I was also suggesting the following identifying string to make it unambiguous up to L4 protocol.
>> I don’t hear any objections to it explicitly. Or did I misinterpret the response?
>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>   
>> BR
>> Raju
>>   
>> From: Justin Uberti [mailto:juberti@google.com]
>> Sent: Thursday, January 22, 2015 4:45 PM
>> To: Makaraju, Maridi Raju (Raju)
>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>   
>> The protocol is the same. We are just deciding what the identifying string should be.
>>   
>> Given the amount of confusion in this space, I think it makes sense to be as explicit as possible, and use DTLS instead of TLS for the name.
>>   
>> On Thu, Jan 22, 2015 at 4:01 AM, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@alcatel-lucent.com> wrote:
>>  From protocol stacks layering point of view, isn’t the following more accurately represent them?
>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>   
>> Per my understanding, independent of underlying transport (UDP; or “TCP over RFC4471” framing mimicking datagram transport) the DTLS protocol is same.
>>   
>> BR
>> Raju
>>   
>>   
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>   
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Feb  4 03:07:34 2015
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Date: Wed, 4 Feb 2015 12:07:24 +0100
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References: <CAJrXDUEBBQmaixOJ+oOsUefw-YwyOYQnm8CBn15VpbjSL5xhtw@mail.gmail.com> <786615F3A85DF44AA2A76164A71FE1AC4B360F87@FR711WXCHMBA01.zeu.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B1D675C09@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17828E6A482E@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAOJ7v-1tODn-fqSvGAf1ToY0htkA_q=zBx48tLQKNo2-YC5azg@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17828E6A5D45@US70UWXCHMBA02.zam.alcatel-lucent.com> <CAOJ7v-3YEeZ+AhcEnap6fK5B03wYnP0d6fQ5ZkLWM5q9GnArTQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D678F81@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17828E6A88E6@US70UWXCHMBA02.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B1D681465@ESESSMB209.ericsson.se> <CABcZeBMQJbBVEdhWjJzY-N=-0wGAa6mn3fZ8Bb4qTm6WLVHnig@ mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D682175@ESESSMB209.ericsson.se> <C6952D3E-7BDA-4CE2-8C5A-D6C23CBEE60B@lurchi.franken.de> <54D1F710.4090504@nteczone.com>
To: Christian Groves <Christian.Groves@NTECZONE.COM>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
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> On 04 Feb 2015, at 11:40, Christian Groves =
<Christian.Groves@NTECZONE.COM> wrote:
>=20
> Is it too late to add some explicit text indicating TCP rather than =
having to reply on another draft? It's easily missed by people reading =
the draft and given the real estate given in the data channel draft its =
possible to come to another conclusion.
The document is in the RFC Editor queue... I would think that it is =
clear that ICE can run over several
protocols, not only UDP. So it might have been better not show nothing =
below ICE. But it think ICE/UDP
will be the stack used in a lot of cases.

We could still try to make sure the stack is clear in
https://tools.ietf.org/html/draft-ietf-rtcweb-transports-07

That has the advantage that it also deals with the media channels.

Best regards
Michael
>=20
> Christian
>=20
> On 28/01/2015 5:15 AM, Michael Tuexen wrote:
>>> On 27 Jan 2015, at 18:44, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>>=20
>>> Hi,
>>> =20
>>>> Well, if you are doing ICE-TCP, I would expect you to run =
>DataChannels over that. ISTM that that suggests that the marker >should =
be TCP/DTLS/SCTP.
>>>  I am not arguing against that. I just wonder why the data channel =
draft doesn=E2=80=99t say anything about TCP, while explicitly talking =
about UDP.
>> That is correct. However,
>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-09
>> says in the Abstract:
>>    Using the
>>    encapsulation method described in this document, SCTP is unaware =
of
>>    the protocols being used below DTLS;
>> So it is clear that you can use other protocols like ICE/TCP. =
However, this
>> is not explicitly discussed and using TCP has the known drawback of =
running
>> a CC (in SCTP) over a CC (in TCP).
>>=20
>> I guess the reason why TCP is not mentioned explicitly is that noone =
brought
>> the issue up. It was always meant that you run over ICE no matter =
what ICE is running over...
>>=20
>> Best regards
>> Michael
>>>  Regards,
>>>  Christer
>>>      On Tue, Jan 27, 2015 at 4:18 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>> Hi,
>>>  Related to this, when I read the data channel draft, it explicitly =
talks about UDP. TCP is not even mentioned.
>>>  So, does that mean that TCP/DTLS/SCTP is not =E2=80=9Cofficially=E2=80=
=9D a part of the data channel spec?
>>>  Regards,
>>>  Christer
>>>  From: Makaraju, Maridi Raju (Raju) =
[mailto:Raju.Makaraju@alcatel-lucent.com]
>>> Sent: 24. tammikuuta 2015 15:35
>>> To: Christer Holmberg; Justin Uberti
>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; =
<rtcweb@ietf.org>
>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>  I am ok to use UDP/DTLS/SCTP and TCP/DTLS/SCTP.
>>> I agree that use of RFC #s is not the best option.
>>> However, I find some unease in the discussions about =E2=80=9CTCP/DTLS=
=E2=80=9D part, which seems to suggest why not use =E2=80=9CTCP/TLS=E2=80=9D=
?  We understand that it can=E2=80=99t be called that way because of the =
RFC 4571 =E2=80=9Cshim layer=E2=80=9D in between DTLS and TCP layers. =
Unfortunately, unlike TPKT, RFC 4571 did not give a name to the =
protocol, which would have been easy to use than the RFC directly.
>>>  BR
>>> Raju
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: Friday, January 23, 2015 1:22 AM
>>> To: Justin Uberti; Makaraju, Maridi Raju (Raju)
>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; =
<rtcweb@ietf.org>
>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>  Hi,
>>>  I agree that we should not use RFC numbers in proto values.
>>>  Also keep in mind that UDP/DTLS/SCTP does not mean =E2=80=9CICE =
based=E2=80=9D. ICE is optional for UDP/DTLS/SCTP (that fact that we =
mandate ICE for RTCWEB is a separate issue).
>>>  Regards,
>>>  Christer
>>>  From: Justin Uberti [mailto:juberti@google.com]
>>> Sent: 23. tammikuuta 2015 5:57
>>> To: Makaraju, Maridi Raju (Raju)
>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas =
Nandakumar; <rtcweb@ietf.org>
>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>  No, I don't think including RFC4571 is reasonable. That ship has =
already sailed.
>>>  On Thu, Jan 22, 2015 at 3:35 PM, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:
>>> I was also suggesting the following identifying string to make it =
unambiguous up to L4 protocol.
>>> I don=E2=80=99t hear any objections to it explicitly. Or did I =
misinterpret the response?
>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>  BR
>>> Raju
>>>  From: Justin Uberti [mailto:juberti@google.com]
>>> Sent: Thursday, January 22, 2015 4:45 PM
>>> To: Makaraju, Maridi Raju (Raju)
>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas =
Nandakumar; <rtcweb@ietf.org>
>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>  The protocol is the same. We are just deciding what the identifying =
string should be.
>>>  Given the amount of confusion in this space, I think it makes sense =
to be as explicit as possible, and use DTLS instead of TLS for the name.
>>>  On Thu, Jan 22, 2015 at 4:01 AM, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:
>>> =46rom protocol stacks layering point of view, isn=E2=80=99t the =
following more accurately represent them?
>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>  Per my understanding, independent of underlying transport (UDP; or =
=E2=80=9CTCP over RFC4471=E2=80=9D framing mimicking datagram transport) =
the DTLS protocol is same.
>>>  BR
>>> Raju
>>>   =20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>>  _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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To: =?utf-8?Q?G=C3=B6ran_Eriksson_AP?= <goran.ap.eriksson@ericsson.com>
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Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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> On 2 Feb 2015, at 21:13, G=C3=B6ran Eriksson AP =
<goran.ap.eriksson@ericsson.com> wrote:
>=20
> =20
> =20
> From: Tim Panton [mailto:thp@westhawk.co.uk =
<mailto:thp@westhawk.co.uk>]=20
> Sent: den 2 februari 2015 15:17
> To: public-webrtc
> Cc: rtcweb@ietf.org <mailto:rtcweb@ietf.org> >> rtcweb@ietf.org =
<mailto:rtcweb@ietf.org>
> Subject: ICE exposes 'real' local IP to javascript
> =20
> Firstly- sorry for cross posting - I=E2=80=99m not sure which side of =
the line this falls.
> Secondly - if this is covered, please let me know, I don=E2=80=99t =
recall it cropping up...
> =20
> I=E2=80=99ve been reading worried blogs that WEBRTC in browsers =
=E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=80=99 ip addresses to =
the javascript.
> The principle worriers are VPN users e.g =
https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096 =
<https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096>
> The concern is that this can be done without user notification =
(DataChannel request) and might be used to=20
> identify or finger-print users. Clearly the most vulnerable are Tor =
users who are on a real routeable IP address
> or directly on a carrier grade nat (eg android phones etc) where the =
IP may reveal the identity or location of the user.
> =20
> It seems to me that this concern will be increased in the case of ipv6 =
deployments (MNOs).
> =20
> Do we need to specify a config option on the browser =E2=80=98I=E2=80=99=
m using a VPN don=E2=80=99t expose my local IP=E2=80=99=20
> =20
> Again, sorry if I missed this being hashed to death already.
> [GAPE:] There are different =E2=80=9Cchallenges=E2=80=9D as I see it; =
a) one to =E2=80=98hide=E2=80=99 the information from the involved web =
sites and peers and b) another from a web site owner perspective, how to =
safeguard users privacy and security. =E2=80=98a=E2=80=99 has been =
discussed and partly addressed, e.g.  in [1] and [2] .

I think these VPN users have hit the =E2=80=98partly=E2=80=99 aspect. =
5.4 in [1] says:=20
"Note that these requirements are NOT intended to protect the user=E2=80=99=
s
 IP address from a malicious site. In general, the site will learn at
least a user's server reflexive address from any HTTP transaction.=E2=80=9D=
=20
The VPN folks are worried that their privacy may be compromised by =
webRTC exposing their local IP without their permission.
An advert might call a dataChannel createOffer() for the purposes of =
gaining the local IPV6 address and using it to fingerprint the user in
ways that the reflexive address couldn=E2=80=99t. - So they don=E2=80=99t =
want to trust =E2=80=98evil_banners.com=E2=80=99 with their local =
addresses and before webRTC they
didn=E2=80=99t have to. I=E2=80=99m not clear that this is a big enough =
issue to warrant a spec change, but it is an issue they seem to care =
about.

T.


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 2 Feb 2015, at 21:13, G=C3=B6ran Eriksson AP &lt;<a =
href=3D"mailto:goran.ap.eriksson@ericsson.com" =
class=3D"">goran.ap.eriksson@ericsson.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div =
class=3D"WordSection1" style=3D"page: WordSection1; font-family: =
LucidaSansUnicode; font-size: 12px; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;"><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D""><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);" class=3D"">&nbsp;</span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><span style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125);" =
class=3D"">&nbsp;</span></div><div style=3D"border-style: none none none =
solid; border-left-color: blue; border-left-width: 1.5pt; padding: 0cm =
0cm 0cm 4pt;" class=3D""><div class=3D""><div style=3D"border-style: =
solid none none; border-top-color: rgb(181, 196, 223); border-top-width: =
1pt; padding: 3pt 0cm 0cm;" class=3D""><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D""><b class=3D""><span style=3D"font-size: 10pt; font-family: =
Tahoma, sans-serif;" class=3D"">From:</span></b><span style=3D"font-size: =
10pt; font-family: Tahoma, sans-serif;" class=3D""><span =
class=3D"Apple-converted-space">&nbsp;</span>Tim Panton [<a =
href=3D"mailto:thp@westhawk.co.uk" style=3D"color: purple; =
text-decoration: underline;" =
class=3D"">mailto:thp@westhawk.co.uk</a>]<span =
class=3D"Apple-converted-space">&nbsp;</span><br class=3D""><b =
class=3D"">Sent:</b><span class=3D"Apple-converted-space">&nbsp;</span>den=
 2 februari 2015 15:17<br class=3D""><b class=3D"">To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>public-webrtc<br =
class=3D""><b class=3D"">Cc:</b><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: purple; text-decoration: =
underline;" class=3D"">rtcweb@ietf.org</a><span =
class=3D"Apple-converted-space">&nbsp;</span>&gt;&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: purple; text-decoration: =
underline;" class=3D"">rtcweb@ietf.org</a><br class=3D""><b =
class=3D"">Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>ICE exposes 'real' local IP =
to javascript<o:p class=3D""></o:p></span></div></div></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><o:p class=3D"">&nbsp;</o:p></div><div =
class=3D""><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;" class=3D"">Firstly- sorry for =
cross posting - I=E2=80=99m not sure which side of the line this =
falls.<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">Secondly - if this is covered, please let =
me know, I don=E2=80=99t recall it cropping up...<o:p =
class=3D""></o:p></div></div><div class=3D""><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D""><o:p class=3D"">&nbsp;</o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">I=E2=80=99ve been reading worried blogs =
that WEBRTC in browsers =E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=
=80=99 ip addresses to the javascript.<o:p =
class=3D""></o:p></div></div><div class=3D""><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D"">The principle worriers are VPN users e.g<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=3D=
13096#p13096" style=3D"color: purple; text-decoration: underline;" =
class=3D"">https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=
=3D13096#p13096</a><o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">The concern is that this can be done =
without user notification (DataChannel request) and might be used =
to&nbsp;<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">identify or finger-print users. Clearly =
the most vulnerable are Tor users who are on a real routeable IP =
address<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">or directly on a carrier grade nat (eg =
android phones etc) where the IP may reveal the identity or location of =
the user.<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><o:p =
class=3D"">&nbsp;</o:p></div></div><div class=3D""><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;" class=3D"">It seems to me that this concern will be increased in =
the case of ipv6 deployments (MNOs).<o:p class=3D""></o:p></div></div><div=
 class=3D""><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;" class=3D""><o:p =
class=3D"">&nbsp;</o:p></div></div><div class=3D""><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;" class=3D"">Do we need to specify a config option on the browser =
=E2=80=98I=E2=80=99m using a VPN don=E2=80=99t expose my local =
IP=E2=80=99&nbsp;<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><o:p =
class=3D"">&nbsp;</o:p></div></div><div class=3D""><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;" class=3D"">Again, sorry if I missed this being hashed to death =
already.<span style=3D"color: rgb(31, 73, 125);" class=3D""><o:p =
class=3D""></o:p></span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif;" class=3D""><b =
class=3D""><i class=3D""><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125);" class=3D"">[GAPE:] There =
are different =E2=80=9Cchallenges=E2=80=9D as I see it; a) one to =
=E2=80=98hide=E2=80=99 the information from the involved web sites and =
peers and b) another from a web site owner perspective, how to safeguard =
users privacy and security. =E2=80=98a=E2=80=99 has been discussed and =
partly addressed, e.g. &nbsp;in [1] and [2] =
.</span></i></b></div></div></div></div></div></blockquote><div><br =
class=3D""></div><div>I think these VPN users have hit the =E2=80=98partly=
=E2=80=99 aspect. 5.4 in [1] says:&nbsp;</div><div>"<span =
style=3D"white-space: pre-wrap;" class=3D"">Note that these requirements =
are NOT intended to protect the user=E2=80=99s</span></div><div><span =
style=3D"white-space: pre-wrap;" class=3D""> IP address from a malicious =
site.&nbsp;</span><span style=3D"white-space: pre-wrap;" class=3D"">In =
general, the site will learn at</span></div><div><span =
style=3D"white-space: pre-wrap;" class=3D"">least a user's server =
reflexive address from any HTTP transaction.=E2=80=9D =
</span></div><div><span style=3D"white-space: pre-wrap;" class=3D"">The =
VPN folks are worried that their privacy may be compromised by webRTC =
exposing their local IP without their permission.</span></div><div><span =
style=3D"white-space: pre-wrap;" class=3D"">An advert might call a =
dataChannel createOffer() for the purposes of gaining the local IPV6 =
address and using it to fingerprint the user in</span></div><div><span =
style=3D"white-space: pre-wrap;" class=3D"">ways that the reflexive =
address couldn=E2=80=99t. - So they don=E2=80=99t want to trust =
=E2=80=98evil_banners.com=E2=80=99 with their local addresses and before =
webRTC they</span></div><div><span style=3D"white-space: pre-wrap;" =
class=3D"">didn=E2=80=99t have to. I=E2=80=99m not clear that this is a =
big enough issue to warrant a spec change, but it is an issue they seem =
to care about.</span></div><div><span style=3D"white-space: pre-wrap;" =
class=3D""><br class=3D""></span></div><div class=3D"">T.</div><blockquote=
 type=3D"cite" class=3D""><div class=3D""><div class=3D"WordSection1" =
style=3D"page: WordSection1; font-family: LucidaSansUnicode; font-size: =
12px; font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;"><div =
style=3D"border-style: none none none solid; border-left-color: blue; =
border-left-width: 1.5pt; padding: 0cm 0cm 0cm 4pt;" class=3D""><div =
class=3D""><div class=3D""><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif;" class=3D""><span =
style=3D"font-size: 9pt; font-family: Helvetica, sans-serif;" =
class=3D""></span></div></div></div></div></div></div></blockquote></div><=
br class=3D""></body></html>=

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Subject: Re: [rtcweb] ICE exposes 'real' local IP to javascript
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On Feb 2, 2015, at 02:16 PM, Tim Panton new <thp@westhawk.co.uk> wrote:=20=

>=20
>> On 2 Feb 2015, at 21:13, G=C3=B6ran Eriksson AP =
<goran.ap.eriksson@ericsson.com <mailto:goran.ap.eriksson@ericsson.com>> =
wrote:
>>=20
>> =20
>> =20
>> From: Tim Panton [mailto:thp@westhawk.co.uk =
<mailto:thp@westhawk.co.uk>]=20
>> Sent: den 2 februari 2015 15:17
>> To: public-webrtc
>> Cc: rtcweb@ietf.org <mailto:rtcweb@ietf.org> >> rtcweb@ietf.org =
<mailto:rtcweb@ietf.org>
>> Subject: ICE exposes 'real' local IP to javascript
>> =20
>> Firstly- sorry for cross posting - I=E2=80=99m not sure which side of =
the line this falls.
>> Secondly - if this is covered, please let me know, I don=E2=80=99t =
recall it cropping up...
>> =20
>> I=E2=80=99ve been reading worried blogs that WEBRTC in browsers =
=E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=80=99 ip addresses to =
the javascript.
>> The principle worriers are VPN users e.g =
https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096 =
<https://cryptostorm.org/viewtopic.php?f=3D50&t=3D2867&p=3D13096#p13096>
>> The concern is that this can be done without user notification =
(DataChannel request) and might be used to=20
>> identify or finger-print users. Clearly the most vulnerable are Tor =
users who are on a real routeable IP address
>> or directly on a carrier grade nat (eg android phones etc) where the =
IP may reveal the identity or location of the user.
>> =20
>> It seems to me that this concern will be increased in the case of =
ipv6 deployments (MNOs).
>> =20
>> Do we need to specify a config option on the browser =E2=80=98I=E2=80=99=
m using a VPN don=E2=80=99t expose my local IP=E2=80=99=20
>> =20
>> Again, sorry if I missed this being hashed to death already.
>> [GAPE:] There are different =E2=80=9Cchallenges=E2=80=9D as I see it; =
a) one to =E2=80=98hide=E2=80=99 the information from the involved web =
sites and peers and b) another from a web site owner perspective, how to =
safeguard users privacy and security. =E2=80=98a=E2=80=99 has been =
discussed and partly addressed, e.g.  in [1] and [2] .
>=20
> I think these VPN users have hit the =E2=80=98partly=E2=80=99 aspect. =
5.4 in [1] says:=20
> "Note that these requirements are NOT intended to protect the user=E2=80=
=99s
>  IP address from a malicious site. In general, the site will learn at
> least a user's server reflexive address from any HTTP transaction.=E2=80=
=9D=20
> The VPN folks are worried that their privacy may be compromised by =
webRTC exposing their local IP without their permission.
> An advert might call a dataChannel createOffer() for the purposes of =
gaining the local IPV6 address and using it to fingerprint the user in
> ways that the reflexive address couldn=E2=80=99t. - So they don=E2=80=99=
t want to trust =E2=80=98evil_banners.com=E2=80=99 with their local =
addresses and before webRTC they
> didn=E2=80=99t have to. I=E2=80=99m not clear that this is a big =
enough issue to warrant a spec change, but it is an issue they seem to =
care about.

Witholding the local IP address means traffic always has to go through a =
TURN server, which harms the user experience.  Consider for example two =
users in the same office with their traffic tromboning up their little =
DSL access link to a TURN server on the Internet.  We don't want that.  =
But, likewise, those users may value their IPv6 privacy addresses and =
their IPv4 NAPT to obscure "Alice" from "Bob" at that same company, and =
disclosing their local IP address does leak information.  This worry has =
long been a problem on enterprise networks implementing ICE, to such a =
degree that stripping a=3Dcandidate lines containing the internal =
addresses is desirable for privacy (as well as to force ICE to complete =
faster, as a firewall or NAT would prevent ICE from working with those =
addresses in many cases, anyway).

-d






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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; -webkit-line-break: after-white-space;" =
class=3D""><div><div class=3D"">
<br class=3D"">On Feb 2, 2015, at 02:16 PM, Tim Panton new &lt;<a =
href=3D"mailto:thp@westhawk.co.uk" class=3D"">thp@westhawk.co.uk</a>&gt; =
wrote:  <br class=3D""></div><blockquote type=3D"cite" class=3D""><div =
class=3D"">
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dutf-8" =
class=3D""><div style=3D"word-wrap: break-word; -webkit-nbsp-mode: =
space; -webkit-line-break: after-white-space;" class=3D""><br =
class=3D""><div class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 2 Feb 2015, at 21:13, G=C3=B6ran Eriksson AP &lt;<a =
href=3D"mailto:goran.ap.eriksson@ericsson.com" =
class=3D"">goran.ap.eriksson@ericsson.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div =
class=3D"WordSection1" style=3D"page: WordSection1; font-family: =
LucidaSansUnicode; font-size: 12px; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;"><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D""><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);" class=3D"">&nbsp;</span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><span style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125);" =
class=3D"">&nbsp;</span></div><div style=3D"border-style: none none none =
solid; border-left-color: blue; border-left-width: 1.5pt; padding: 0cm =
0cm 0cm 4pt;" class=3D""><div class=3D""><div style=3D"border-style: =
solid none none; border-top-color: rgb(181, 196, 223); border-top-width: =
1pt; padding: 3pt 0cm 0cm;" class=3D""><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D""><b class=3D""><span style=3D"font-size: 10pt; font-family: =
Tahoma, sans-serif;" class=3D"">From:</span></b><span style=3D"font-size: =
10pt; font-family: Tahoma, sans-serif;" class=3D""><span =
class=3D"Apple-converted-space">&nbsp;</span>Tim Panton [<a =
href=3D"mailto:thp@westhawk.co.uk" style=3D"color: purple; =
text-decoration: underline;" =
class=3D"">mailto:thp@westhawk.co.uk</a>]<span =
class=3D"Apple-converted-space">&nbsp;</span><br class=3D""><b =
class=3D"">Sent:</b><span class=3D"Apple-converted-space">&nbsp;</span>den=
 2 februari 2015 15:17<br class=3D""><b class=3D"">To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>public-webrtc<br =
class=3D""><b class=3D"">Cc:</b><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: purple; text-decoration: =
underline;" class=3D"">rtcweb@ietf.org</a><span =
class=3D"Apple-converted-space">&nbsp;</span>&gt;&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: purple; text-decoration: =
underline;" class=3D"">rtcweb@ietf.org</a><br class=3D""><b =
class=3D"">Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>ICE exposes 'real' local IP =
to javascript<o:p class=3D""></o:p></span></div></div></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><o:p class=3D"">&nbsp;</o:p></div><div =
class=3D""><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;" class=3D"">Firstly- sorry for =
cross posting - I=E2=80=99m not sure which side of the line this =
falls.<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">Secondly - if this is covered, please let =
me know, I don=E2=80=99t recall it cropping up...<o:p =
class=3D""></o:p></div></div><div class=3D""><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D""><o:p class=3D"">&nbsp;</o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">I=E2=80=99ve been reading worried blogs =
that WEBRTC in browsers =E2=80=98leaks=E2=80=99 the local =E2=80=98real=E2=
=80=99 ip addresses to the javascript.<o:p =
class=3D""></o:p></div></div><div class=3D""><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;" =
class=3D"">The principle worriers are VPN users e.g<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=3D=
13096#p13096" style=3D"color: purple; text-decoration: underline;" =
class=3D"">https://cryptostorm.org/viewtopic.php?f=3D50&amp;t=3D2867&amp;p=
=3D13096#p13096</a><o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">The concern is that this can be done =
without user notification (DataChannel request) and might be used =
to&nbsp;<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">identify or finger-print users. Clearly =
the most vulnerable are Tor users who are on a real routeable IP =
address<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D"">or directly on a carrier grade nat (eg =
android phones etc) where the IP may reveal the identity or location of =
the user.<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><o:p =
class=3D"">&nbsp;</o:p></div></div><div class=3D""><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;" class=3D"">It seems to me that this concern will be increased in =
the case of ipv6 deployments (MNOs).<o:p class=3D""></o:p></div></div><div=
 class=3D""><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;" class=3D""><o:p =
class=3D"">&nbsp;</o:p></div></div><div class=3D""><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;" class=3D"">Do we need to specify a config option on the browser =
=E2=80=98I=E2=80=99m using a VPN don=E2=80=99t expose my local =
IP=E2=80=99&nbsp;<o:p class=3D""></o:p></div></div><div class=3D""><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;" class=3D""><o:p =
class=3D"">&nbsp;</o:p></div></div><div class=3D""><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;" class=3D"">Again, sorry if I missed this being hashed to death =
already.<span style=3D"color: rgb(31, 73, 125);" class=3D""><o:p =
class=3D""></o:p></span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif;" class=3D""><b =
class=3D""><i class=3D""><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125);" class=3D"">[GAPE:] There =
are different =E2=80=9Cchallenges=E2=80=9D as I see it; a) one to =
=E2=80=98hide=E2=80=99 the information from the involved web sites and =
peers and b) another from a web site owner perspective, how to safeguard =
users privacy and security. =E2=80=98a=E2=80=99 has been discussed and =
partly addressed, e.g. &nbsp;in [1] and [2] =
.</span></i></b></div></div></div></div></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">I think these VPN users =
have hit the =E2=80=98partly=E2=80=99 aspect. 5.4 in [1] =
says:&nbsp;</div><div class=3D"">"<span style=3D"white-space: pre-wrap;" =
class=3D"">Note that these requirements are NOT intended to protect the =
user=E2=80=99s</span></div><div class=3D""><span style=3D"white-space: =
pre-wrap;" class=3D""> IP address from a malicious =
site.&nbsp;</span><span style=3D"white-space: pre-wrap;" class=3D"">In =
general, the site will learn at</span></div><div class=3D""><span =
style=3D"white-space: pre-wrap;" class=3D"">least a user's server =
reflexive address from any HTTP transaction.=E2=80=9D </span></div><div =
class=3D""><span style=3D"white-space: pre-wrap;" class=3D"">The VPN =
folks are worried that their privacy may be compromised by webRTC =
exposing their local IP without their permission.</span></div><div =
class=3D""><span style=3D"white-space: pre-wrap;" class=3D"">An advert =
might call a dataChannel createOffer() for the purposes of gaining the =
local IPV6 address and using it to fingerprint the user =
in</span></div><div class=3D""><span style=3D"white-space: pre-wrap;" =
class=3D"">ways that the reflexive address couldn=E2=80=99t. - So they =
don=E2=80=99t want to trust =E2=80=98evil_banners.com=E2=80=99 with =
their local addresses and before webRTC they</span></div><div =
class=3D""><span style=3D"white-space: pre-wrap;" class=3D"">didn=E2=80=99=
t have to. I=E2=80=99m not clear that this is a big enough issue to =
warrant a spec change, but it is an issue they seem to care =
about.</span></div></div></div></div></blockquote><br =
class=3D""></div><div>Witholding the local IP address means traffic =
always has to go through a TURN server, which harms the user experience. =
&nbsp;Consider for example two users in the same office with their =
traffic tromboning up their little DSL access link to a TURN server on =
the Internet. &nbsp;We don't want that. &nbsp;But, likewise, those users =
may value their IPv6 privacy addresses and their IPv4 NAPT to obscure =
"Alice" from "Bob" at that same company, and disclosing their local IP =
address does leak information. &nbsp;This worry has long been a problem =
on enterprise networks implementing ICE, to such a degree that stripping =
a=3Dcandidate lines containing the internal addresses is desirable for =
privacy (as well as to force ICE to complete faster, as a firewall or =
NAT would prevent ICE from working with those addresses in many cases, =
anyway).</div><div><br class=3D""></div><div>-d</div><div><br =
class=3D""></div><div><br class=3D""></div><div><br class=3D""></div><br =
class=3D""><div class=3D""><br class=3D""></div></body></html>=

--Apple-Mail=_23BE5529-278A-4809-999E-EEDE50E0BD44--


From nobody Wed Feb  4 16:02:28 2015
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To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
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Hello Michael,

Yes I think UDP will be the common case and I agree ICE can run over 
several protocols however Figure 1 (and associated 
text)/[http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13] 
only shows ICE/UDP. The draft really gives the impression that whilst 
ICE is used, ICE/UDP is the variant the draft has chosen.

Could we ask the RFC editor to insert something in the introduction like 
"Note: This document focusses on SCTP over DTLS over ICE/UDP. However 
SCTP over DTLS over ICE/TCP is also supported."?

I think it's much better to try to get something in the data channel 
draft than updating the transports draft. Although it still wouldn't 
hurt to add something there either.

Regards, Christian

On 4/02/2015 10:07 PM, Michael Tuexen wrote:
>> On 04 Feb 2015, at 11:40, Christian Groves <Christian.Groves@NTECZONE.COM> wrote:
>>
>> Is it too late to add some explicit text indicating TCP rather than having to reply on another draft? It's easily missed by people reading the draft and given the real estate given in the data channel draft its possible to come to another conclusion.
> The document is in the RFC Editor queue... I would think that it is clear that ICE can run over several
> protocols, not only UDP. So it might have been better not show nothing below ICE. But it think ICE/UDP
> will be the stack used in a lot of cases.
>
> We could still try to make sure the stack is clear in
> https://tools.ietf.org/html/draft-ietf-rtcweb-transports-07
>
> That has the advantage that it also deals with the media channels.
>
> Best regards
> Michael
>> Christian
>>
>> On 28/01/2015 5:15 AM, Michael Tuexen wrote:
>>>> On 27 Jan 2015, at 18:44, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>>>
>>>> Hi,
>>>>   
>>>>> Well, if you are doing ICE-TCP, I would expect you to run >DataChannels over that. ISTM that that suggests that the marker >should be TCP/DTLS/SCTP.
>>>>   I am not arguing against that. I just wonder why the data channel draft doesn’t say anything about TCP, while explicitly talking about UDP.
>>> That is correct. However,
>>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-09
>>> says in the Abstract:
>>>     Using the
>>>     encapsulation method described in this document, SCTP is unaware of
>>>     the protocols being used below DTLS;
>>> So it is clear that you can use other protocols like ICE/TCP. However, this
>>> is not explicitly discussed and using TCP has the known drawback of running
>>> a CC (in SCTP) over a CC (in TCP).
>>>
>>> I guess the reason why TCP is not mentioned explicitly is that noone brought
>>> the issue up. It was always meant that you run over ICE no matter what ICE is running over...
>>>
>>> Best regards
>>> Michael
>>>>   Regards,
>>>>   Christer
>>>>       On Tue, Jan 27, 2015 at 4:18 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>>> Hi,
>>>>   Related to this, when I read the data channel draft, it explicitly talks about UDP. TCP is not even mentioned.
>>>>   So, does that mean that TCP/DTLS/SCTP is not “officially” a part of the data channel spec?
>>>>   Regards,
>>>>   Christer
>>>>   From: Makaraju, Maridi Raju (Raju) [mailto:Raju.Makaraju@alcatel-lucent.com]
>>>> Sent: 24. tammikuuta 2015 15:35
>>>> To: Christer Holmberg; Justin Uberti
>>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>   I am ok to use UDP/DTLS/SCTP and TCP/DTLS/SCTP.
>>>> I agree that use of RFC #s is not the best option.
>>>> However, I find some unease in the discussions about “TCP/DTLS” part, which seems to suggest why not use “TCP/TLS”?  We understand that it can’t be called that way because of the RFC 4571 “shim layer” in between DTLS and TCP layers. Unfortunately, unlike TPKT, RFC 4571 did not give a name to the protocol, which would have been easy to use than the RFC directly.
>>>>   BR
>>>> Raju
>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>> Sent: Friday, January 23, 2015 1:22 AM
>>>> To: Justin Uberti; Makaraju, Maridi Raju (Raju)
>>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>   Hi,
>>>>   I agree that we should not use RFC numbers in proto values.
>>>>   Also keep in mind that UDP/DTLS/SCTP does not mean “ICE based”. ICE is optional for UDP/DTLS/SCTP (that fact that we mandate ICE for RTCWEB is a separate issue).
>>>>   Regards,
>>>>   Christer
>>>>   From: Justin Uberti [mailto:juberti@google.com]
>>>> Sent: 23. tammikuuta 2015 5:57
>>>> To: Makaraju, Maridi Raju (Raju)
>>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>   No, I don't think including RFC4571 is reasonable. That ship has already sailed.
>>>>   On Thu, Jan 22, 2015 at 3:35 PM, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>> I was also suggesting the following identifying string to make it unambiguous up to L4 protocol.
>>>> I don’t hear any objections to it explicitly. Or did I misinterpret the response?
>>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>>   BR
>>>> Raju
>>>>   From: Justin Uberti [mailto:juberti@google.com]
>>>> Sent: Thursday, January 22, 2015 4:45 PM
>>>> To: Makaraju, Maridi Raju (Raju)
>>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>   The protocol is the same. We are just deciding what the identifying string should be.
>>>>   Given the amount of confusion in this space, I think it makes sense to be as explicit as possible, and use DTLS instead of TLS for the name.
>>>>   On Thu, Jan 22, 2015 at 4:01 AM, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>>  From protocol stacks layering point of view, isn’t the following more accurately represent them?
>>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>>   Per my understanding, independent of underlying transport (UDP; or “TCP over RFC4471” framing mimicking datagram transport) the DTLS protocol is same.
>>>>   BR
>>>> Raju
>>>>     
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>   _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>


From nobody Wed Feb  4 16:29:57 2015
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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To: Christian Groves <Christian.Groves@NTECZONE.COM>
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Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
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> On 05 Feb 2015, at 01:02, Christian Groves =
<Christian.Groves@NTECZONE.COM> wrote:
>=20
> Hello Michael,
>=20
> Yes I think UDP will be the common case and I agree ICE can run over =
several protocols however Figure 1 (and associated =
text)/[http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13] =
only shows ICE/UDP. The draft really gives the impression that whilst =
ICE is used, ICE/UDP is the variant the draft has chosen.
>=20
> Could we ask the RFC editor to insert something in the introduction =
like "Note: This document focusses on SCTP over DTLS over ICE/UDP. =
However SCTP over DTLS over ICE/TCP is also supported."?
I don't know. I don't think this is appropriate for an AUTH48 comment. =
But the WG chairs might now
the process better...

Best regards
Michael
>=20
> I think it's much better to try to get something in the data channel =
draft than updating the transports draft. Although it still wouldn't =
hurt to add something there either.
>=20
> Regards, Christian
>=20
> On 4/02/2015 10:07 PM, Michael Tuexen wrote:
>>> On 04 Feb 2015, at 11:40, Christian Groves =
<Christian.Groves@NTECZONE.COM> wrote:
>>>=20
>>> Is it too late to add some explicit text indicating TCP rather than =
having to reply on another draft? It's easily missed by people reading =
the draft and given the real estate given in the data channel draft its =
possible to come to another conclusion.
>> The document is in the RFC Editor queue... I would think that it is =
clear that ICE can run over several
>> protocols, not only UDP. So it might have been better not show =
nothing below ICE. But it think ICE/UDP
>> will be the stack used in a lot of cases.
>>=20
>> We could still try to make sure the stack is clear in
>> https://tools.ietf.org/html/draft-ietf-rtcweb-transports-07
>>=20
>> That has the advantage that it also deals with the media channels.
>>=20
>> Best regards
>> Michael
>>> Christian
>>>=20
>>> On 28/01/2015 5:15 AM, Michael Tuexen wrote:
>>>>> On 27 Jan 2015, at 18:44, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>>>>=20
>>>>> Hi,
>>>>> =20
>>>>>> Well, if you are doing ICE-TCP, I would expect you to run =
>DataChannels over that. ISTM that that suggests that the marker >should =
be TCP/DTLS/SCTP.
>>>>>  I am not arguing against that. I just wonder why the data channel =
draft doesn=E2=80=99t say anything about TCP, while explicitly talking =
about UDP.
>>>> That is correct. However,
>>>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-09
>>>> says in the Abstract:
>>>>    Using the
>>>>    encapsulation method described in this document, SCTP is unaware =
of
>>>>    the protocols being used below DTLS;
>>>> So it is clear that you can use other protocols like ICE/TCP. =
However, this
>>>> is not explicitly discussed and using TCP has the known drawback of =
running
>>>> a CC (in SCTP) over a CC (in TCP).
>>>>=20
>>>> I guess the reason why TCP is not mentioned explicitly is that =
noone brought
>>>> the issue up. It was always meant that you run over ICE no matter =
what ICE is running over...
>>>>=20
>>>> Best regards
>>>> Michael
>>>>>  Regards,
>>>>>  Christer
>>>>>      On Tue, Jan 27, 2015 at 4:18 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>>>> Hi,
>>>>>  Related to this, when I read the data channel draft, it =
explicitly talks about UDP. TCP is not even mentioned.
>>>>>  So, does that mean that TCP/DTLS/SCTP is not =E2=80=9Cofficially=E2=
=80=9D a part of the data channel spec?
>>>>>  Regards,
>>>>>  Christer
>>>>>  From: Makaraju, Maridi Raju (Raju) =
[mailto:Raju.Makaraju@alcatel-lucent.com]
>>>>> Sent: 24. tammikuuta 2015 15:35
>>>>> To: Christer Holmberg; Justin Uberti
>>>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; =
<rtcweb@ietf.org>
>>>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>>>  I am ok to use UDP/DTLS/SCTP and TCP/DTLS/SCTP.
>>>>> I agree that use of RFC #s is not the best option.
>>>>> However, I find some unease in the discussions about =
=E2=80=9CTCP/DTLS=E2=80=9D part, which seems to suggest why not use =
=E2=80=9CTCP/TLS=E2=80=9D?  We understand that it can=E2=80=99t be =
called that way because of the RFC 4571 =E2=80=9Cshim layer=E2=80=9D in =
between DTLS and TCP layers. Unfortunately, unlike TPKT, RFC 4571 did =
not give a name to the protocol, which would have been easy to use than =
the RFC directly.
>>>>>  BR
>>>>> Raju
>>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>>> Sent: Friday, January 23, 2015 1:22 AM
>>>>> To: Justin Uberti; Makaraju, Maridi Raju (Raju)
>>>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; =
<rtcweb@ietf.org>
>>>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>>>  Hi,
>>>>>  I agree that we should not use RFC numbers in proto values.
>>>>>  Also keep in mind that UDP/DTLS/SCTP does not mean =E2=80=9CICE =
based=E2=80=9D. ICE is optional for UDP/DTLS/SCTP (that fact that we =
mandate ICE for RTCWEB is a separate issue).
>>>>>  Regards,
>>>>>  Christer
>>>>>  From: Justin Uberti [mailto:juberti@google.com]
>>>>> Sent: 23. tammikuuta 2015 5:57
>>>>> To: Makaraju, Maridi Raju (Raju)
>>>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas =
Nandakumar; <rtcweb@ietf.org>
>>>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>>>  No, I don't think including RFC4571 is reasonable. That ship has =
already sailed.
>>>>>  On Thu, Jan 22, 2015 at 3:35 PM, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:
>>>>> I was also suggesting the following identifying string to make it =
unambiguous up to L4 protocol.
>>>>> I don=E2=80=99t hear any objections to it explicitly. Or did I =
misinterpret the response?
>>>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>>>  BR
>>>>> Raju
>>>>>  From: Justin Uberti [mailto:juberti@google.com]
>>>>> Sent: Thursday, January 22, 2015 4:45 PM
>>>>> To: Makaraju, Maridi Raju (Raju)
>>>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas =
Nandakumar; <rtcweb@ietf.org>
>>>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel =
m-lines, right?
>>>>>  The protocol is the same. We are just deciding what the =
identifying string should be.
>>>>>  Given the amount of confusion in this space, I think it makes =
sense to be as explicit as possible, and use DTLS instead of TLS for the =
name.
>>>>>  On Thu, Jan 22, 2015 at 4:01 AM, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:
>>>>> =46rom protocol stacks layering point of view, isn=E2=80=99t the =
following more accurately represent them?
>>>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>>>  Per my understanding, independent of underlying transport (UDP; =
or =E2=80=9CTCP over RFC4471=E2=80=9D framing mimicking datagram =
transport) the DTLS protocol is same.
>>>>>  BR
>>>>> Raju
>>>>>    _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>=20
>>>>>  _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20
>=20


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Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
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Den 04. feb. 2015 12:07, skrev Michael Tuexen:
>> On 04 Feb 2015, at 11:40, Christian Groves <Christian.Groves@NTECZONE.COM> wrote:
>>
>> Is it too late to add some explicit text indicating TCP rather than having to reply on another draft? It's easily missed by people reading the draft and given the real estate given in the data channel draft its possible to come to another conclusion.
> The document is in the RFC Editor queue... I would think that it is clear that ICE can run over several
> protocols, not only UDP. So it might have been better not show nothing below ICE. But it think ICE/UDP
> will be the stack used in a lot of cases.
> 
> We could still try to make sure the stack is clear in
> https://tools.ietf.org/html/draft-ietf-rtcweb-transports-07
> 
> That has the advantage that it also deals with the media channels.

Completely BTW: The -transports draft is now on github:

git@github.com:alvestrand/webrtc-specs.git

Feel free to send me a pull request if you have specific language
suggestions.


> 
> Best regards
> Michael
>>
>> Christian
>>
>> On 28/01/2015 5:15 AM, Michael Tuexen wrote:
>>>> On 27 Jan 2015, at 18:44, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>>>
>>>> Hi,
>>>>  
>>>>> Well, if you are doing ICE-TCP, I would expect you to run >DataChannels over that. ISTM that that suggests that the marker >should be TCP/DTLS/SCTP.
>>>>  I am not arguing against that. I just wonder why the data channel draft doesn’t say anything about TCP, while explicitly talking about UDP.
>>> That is correct. However,
>>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-09
>>> says in the Abstract:
>>>    Using the
>>>    encapsulation method described in this document, SCTP is unaware of
>>>    the protocols being used below DTLS;
>>> So it is clear that you can use other protocols like ICE/TCP. However, this
>>> is not explicitly discussed and using TCP has the known drawback of running
>>> a CC (in SCTP) over a CC (in TCP).
>>>
>>> I guess the reason why TCP is not mentioned explicitly is that noone brought
>>> the issue up. It was always meant that you run over ICE no matter what ICE is running over...
>>>
>>> Best regards
>>> Michael
>>>>  Regards,
>>>>  Christer
>>>>      On Tue, Jan 27, 2015 at 4:18 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>>> Hi,
>>>>  Related to this, when I read the data channel draft, it explicitly talks about UDP. TCP is not even mentioned.
>>>>  So, does that mean that TCP/DTLS/SCTP is not “officially” a part of the data channel spec?
>>>>  Regards,
>>>>  Christer
>>>>  From: Makaraju, Maridi Raju (Raju) [mailto:Raju.Makaraju@alcatel-lucent.com]
>>>> Sent: 24. tammikuuta 2015 15:35
>>>> To: Christer Holmberg; Justin Uberti
>>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>  I am ok to use UDP/DTLS/SCTP and TCP/DTLS/SCTP.
>>>> I agree that use of RFC #s is not the best option.
>>>> However, I find some unease in the discussions about “TCP/DTLS” part, which seems to suggest why not use “TCP/TLS”?  We understand that it can’t be called that way because of the RFC 4571 “shim layer” in between DTLS and TCP layers. Unfortunately, unlike TPKT, RFC 4571 did not give a name to the protocol, which would have been easy to use than the RFC directly.
>>>>  BR
>>>> Raju
>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>> Sent: Friday, January 23, 2015 1:22 AM
>>>> To: Justin Uberti; Makaraju, Maridi Raju (Raju)
>>>> Cc: Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: RE: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>  Hi,
>>>>  I agree that we should not use RFC numbers in proto values.
>>>>  Also keep in mind that UDP/DTLS/SCTP does not mean “ICE based”. ICE is optional for UDP/DTLS/SCTP (that fact that we mandate ICE for RTCWEB is a separate issue).
>>>>  Regards,
>>>>  Christer
>>>>  From: Justin Uberti [mailto:juberti@google.com]
>>>> Sent: 23. tammikuuta 2015 5:57
>>>> To: Makaraju, Maridi Raju (Raju)
>>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>  No, I don't think including RFC4571 is reasonable. That ship has already sailed.
>>>>  On Thu, Jan 22, 2015 at 3:35 PM, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>> I was also suggesting the following identifying string to make it unambiguous up to L4 protocol.
>>>> I don’t hear any objections to it explicitly. Or did I misinterpret the response?
>>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>>  BR
>>>> Raju
>>>>  From: Justin Uberti [mailto:juberti@google.com]
>>>> Sent: Thursday, January 22, 2015 4:45 PM
>>>> To: Makaraju, Maridi Raju (Raju)
>>>> Cc: Christer Holmberg; Schwarz, Albrecht (Albrecht); Suhas Nandakumar; <rtcweb@ietf.org>
>>>> Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
>>>>  The protocol is the same. We are just deciding what the identifying string should be.
>>>>  Given the amount of confusion in this space, I think it makes sense to be as explicit as possible, and use DTLS instead of TLS for the name.
>>>>  On Thu, Jan 22, 2015 at 4:01 AM, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@alcatel-lucent.com> wrote:
>>>> From protocol stacks layering point of view, isn’t the following more accurately represent them?
>>>> For TCP ICE based data channel transport: TCP/RFC4571/DTLS/SCTP
>>>> For UDP ICE based data channel transport: UDP/DTLS/SCTP
>>>>  Per my understanding, independent of underlying transport (UDP; or “TCP over RFC4471” framing mimicking datagram transport) the DTLS protocol is same.
>>>>  BR
>>>> Raju
>>>>    
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>  _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
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Den 05. feb. 2015 12:39, skrev Harald Alvestrand:
> Den 04. feb. 2015 12:07, skrev Michael Tuexen:
>>> On 04 Feb 2015, at 11:40, Christian Groves <Christian.Groves@NTECZONE.COM> wrote:
>>>
>>> Is it too late to add some explicit text indicating TCP rather than having to reply on another draft? It's easily missed by people reading the draft and given the real estate given in the data channel draft its possible to come to another conclusion.
>> The document is in the RFC Editor queue... I would think that it is clear that ICE can run over several
>> protocols, not only UDP. So it might have been better not show nothing below ICE. But it think ICE/UDP
>> will be the stack used in a lot of cases.
>>
>> We could still try to make sure the stack is clear in
>> https://tools.ietf.org/html/draft-ietf-rtcweb-transports-07
>>
>> That has the advantage that it also deals with the media channels.
> 
> Completely BTW: The -transports draft is now on github:
> 
> git@github.com:alvestrand/webrtc-specs.git

Self-correcting: The master master is at this URL:

git@github.com:rtcweb-wg/rtcweb-transport.git

There is no issue tracker, and I can't enable one - seems the site
admins have to do that.

> 
> Feel free to send me a pull request if you have specific language
> suggestions.

This still applies.


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From: Ted Hardie <ted.ietf@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] It's "UDP/DTLS/SCTP" for the data channel m-lines, right?
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--047d7b41444c62a1de050e5cd34c
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On Thu, Feb 5, 2015 at 3:46 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>
> Self-correcting: The master master is at this URL:
>
> git@github.com:rtcweb-wg/rtcweb-transport.git
>
> There is no issue tracker, and I can't enable one - seems the site
> admins have to do that.
>

=E2=80=8BI believe the issue tracker should be working now.

regards,

Ted=E2=80=8B

--047d7b41444c62a1de050e5cd34c
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Feb 5, 2015 at 3:46 AM,=
 Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestran=
d.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><div =
class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=3D""><br>
</span>Self-correcting: The master master is at this URL:<br>
<br>
git@github.com:rtcweb-wg/rtcweb-transport.git<br>
<br>
There is no issue tracker, and I can&#39;t enable one - seems the site<br>
admins have to do that.<br>
<span class=3D""></span></blockquote><div><br><div class=3D"gmail_default" =
style=3D"font-family:tahoma,sans-serif;font-size:small;display:inline">=E2=
=80=8BI believe the issue tracker should be working now.<br><br>regards,<br=
><br>Ted=E2=80=8B</div>=C2=A0</div></div></div></div>

--047d7b41444c62a1de050e5cd34c--


From nobody Thu Feb  5 15:40:05 2015
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To: Barry Dingle <btdingle@gmail.com>
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Subject: Re: [rtcweb] Consistent WebRTC Descriptions in WebRTC RFC Abstracts
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Hi Barry,


On Tue, Jan 13, 2015 at 12:15:27PM +1100, Barry Dingle wrote:
> The recently proposed Standards *WebRTC Data Channel RFC* and *WebRTC Data
> Channel Establishment Protocol RFC* have a consistent single sentence
> describing what WebRTC is in their Abstracts. They both say:
> 
> The WebRTC framework specifies protocol support for direct interactive
> rich communication using audio,
> video, and data between two peers' web-browsers.
> 
> I notice that several other WebRTC proposed RFCs have single sentence
> WebRTC descriptions in their Abstract *but they are differen*t e.g. WebRTC
> Overview RFC; Media Transport + Use of RTP; Security; Security Architecture
> 
> All the others have *NO WebRTC description*.
> 
> *Do we need a consistent single sentence in the Abstract of all WebRTC
> RFCs?*

We did not see any further discusison on this topic.  The RFC Editor
does not have a requirement here.  We suggest discussing this with
your WG group and chairs. 

Thank you,
RFC Editor/sg


> 
> If so, what should the wording be?
> 
> Should the wording be in the Abstract of ALL WebRTC RFCs?
> 
> 
> Barry Dingle
> 
> Fellow of University of Melbourne, Australia
> 
> On Fri, Jan 9, 2015 at 1:38 AM, The IESG <iesg-secretary@ietf.org> wrote:
> 
> > The IESG has approved the following document:
> > - 'WebRTC Data Channels'
> >   (draft-ietf-rtcweb-data-channel-13.txt) as Proposed Standard
> >
> > This document is the product of the Real-Time Communication in
> > WEB-browsers Working Group.
> >
> > The IESG contact persons are Richard Barnes and Alissa Cooper.
> >
> > A URL of this Internet Draft is:
> > http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/
> >
> >
> >
> >
> > Technical Summary:
> >
> > This document specifies the non??media data transport aspects of the WebRTC
> > framework. It provides an architectural overview of how the Stream Control
> > Transmission
> > Protocol (SCTP) is used in the WebRTC context as a generic transport
> > service.
> >
> > Working Group Summary:
> >
> > There was early discussion of the stacking order, but there has been no
> > significant
> > controversy since that was fixed. There have been a number of discussion
> > on how to manage
> > particular aspects of the larger context (e.g. WebRTC??level congestion
> > control, since SCTP
> > manages congestion control at the association level) and this has played a
> > part in those, but
> > not in any way that mde it the focus of controversy.
> >
> > Document Quality:
> >
> > There are implmentations of previous versions of this document, and we
> > expect updates to
> > them to the final version. Vendor support seems solid. This document did
> > not require
> > expert review of the types noted.
> >
> > Personnel:
> >
> > The document shepherd is Ted Hardie; the responsible Area Director is
> > Richard Barnes.
> >
> >
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >


From nobody Thu Feb  5 17:02:06 2015
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From: Justin Uberti <juberti@google.com>
Date: Thu, 5 Feb 2015 17:01:36 -0800
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I just resubmitted the current draft as a -00 WG draft.

Once that is accepted, I will work on a -01 with the feedback from Honolulu=
.

On Fri, Jan 30, 2015 at 10:21 AM, Varun Singh <vsingh.ietf@gmail.com> wrote=
:

> I support this too.  The corresponding RTP payload draft was recently
> adopted as well.
> On 30-Jan-2015 6:10 pm, "Magnus Westerlund" <
> magnus.westerlund@ericsson.com> wrote:
>
>> Hi,
>>
>> I support this. Considering that the deadline is passed. Is this now
>> adopted?
>>
>> /Magnus
>>
>> On 2015-01-08 19:57, Cullen Jennings (fluffy) wrote:
>> >
>> > At the last meeting there was strong support for adopting
>> > draft-uberti-rtcweb-fec as a WG draft. Unless we hear significant
>> > objections to this, the chairs plan to more forward with adopting
>> > this. If you have objection to this, please reply to this thread
>> > before Jan 15.
>> >
>> > Thanks,
>> >
>> > Cullen, Sean, & Ted
>> >
>> >
>> >
>> >
>> > _______________________________________________ rtcweb mailing list
>> > rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>> >
>>
>>
>> --
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">I just resubmitted the current draft as a -00 WG draft.<di=
v><br></div><div>Once that is accepted, I will work on a -01 with the feedb=
ack from Honolulu.</div></div><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote">On Fri, Jan 30, 2015 at 10:21 AM, Varun Singh <span dir=3D"ltr=
">&lt;<a href=3D"mailto:vsingh.ietf@gmail.com" target=3D"_blank">vsingh.iet=
f@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir=
=3D"ltr">I support this too.=C2=A0 The corresponding RTP payload draft was =
recently adopted as well. </p><div class=3D"HOEnZb"><div class=3D"h5">
<div class=3D"gmail_quote">On 30-Jan-2015 6:10 pm, &quot;Magnus Westerlund&=
quot; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blan=
k">magnus.westerlund@ericsson.com</a>&gt; wrote:<br type=3D"attribution"><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex">Hi,<br>
<br>
I support this. Considering that the deadline is passed. Is this now<br>
adopted?<br>
<br>
/Magnus<br>
<br>
On 2015-01-08 19:57, Cullen Jennings (fluffy) wrote:<br>
&gt;<br>
&gt; At the last meeting there was strong support for adopting<br>
&gt; draft-uberti-rtcweb-fec as a WG draft. Unless we hear significant<br>
&gt; objections to this, the chairs plan to more forward with adopting<br>
&gt; this. If you have objection to this, please reply to this thread<br>
&gt; before Jan 15.<br>
&gt;<br>
&gt; Thanks,<br>
&gt;<br>
&gt; Cullen, Sean, &amp; Ted<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________ rtcweb mailing list<br=
>
&gt; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a> <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
<br>
<br>
--<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<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 <a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287" tar=
get=3D"_blank">+46 10 7148287</a><br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+467309490=
79" target=3D"_blank">+46 73 0949079</a><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>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--bcaec5158d918d94a9050e60f753--


From nobody Fri Feb  6 08:56:27 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Forward Error Correction Requirements
        Author          : Justin Uberti
	Filename        : draft-ietf-rtcweb-fec-00.txt
	Pages           : 7
	Date            : 2015-02-05

Abstract:
   This document makes recommendations for how Forward Error Correction
   (FEC) should be used by WebRTC applications.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Fri Feb  6 23:49:16 2015
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Subject: [rtcweb] Use of redundancy and RFC 2119 language in draft-ietf-rtcweb-fec-00
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I have two comments to draft-ietf-rtcweb-fec-00

1. Use of redundancy.
Section 3.2 "Redundant encoding" informs about RFC 2198 .
The last sentence says "This approach is also only applicable to audio 
content."
RFC 2198 redundancy is successfully used in SIP and RTP for real-time 
text content (RFC 4103), because of its robustness, very low cost in 
bandwidth for low bandwidth media, and no risk for excessive delays.
Chapter 3 seems to be a general overview of available methods without 
specific scope to WebRTC.
Therefore, I suggest to either replace the sentence

"This approach is also only applicable to audio content."
with
"This approach is also not applicable to video content."
or delete it.

2. Use of SHOULD and MUST in abstract and introduction
There are a couple of places in the document where the use of "must" and "should" may be confusing and should be avoided.

The last sentence in the introduction says:
"This document describes what FEC mechanisms should be used by WebRTC client implementations."
There is a similar sentence in the abstract.
There is also a "must" in the first sentence of the introduction, even if it is not thought to be normative.
In the document body, there are varying normative levels on the different methods.
Therefore I suggest to avoid the must and should in the abstract and introduction.

2.1 The last sentence of the introduction can be changed to:
"This document provides information and requirements on the use of FEC mechanisms by WebRTC client implementations."

2.2 The abstract could be changed from:
"This document makes recommendations for how Forward Error Correction (FEC) should be used by WebRTC applications."
to
"This document provides information and requirements on the use of Forward Error Correction (FEC) mechanisms by WebRTC applications."

2.3 The first sentence in the introduction, starting:
"In situations where packet loss is high, or media quality must be perfect,"
could be changed to:
"In situations where packet loss is high, or high media quality is essential,"
to avoid the use of "must" that apparently is not intended to be normative.

Regards

Gunnar Hellstrm






internet-drafts@ietf.org skrev den 2015-02-06 17:56:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>
>          Title           : WebRTC Forward Error Correction Requirements
>          Author          : Justin Uberti
> 	Filename        : draft-ietf-rtcweb-fec-00.txt
> 	Pages           : 7
> 	Date            : 2015-02-05
>
> Abstract:
>     This document makes recommendations for how Forward Error Correction
>     (FEC) should be used by WebRTC applications.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt

-- 
-----------------------------------------
Gunnar Hellstrm
Omnitor
gunnar.hellstrom@omnitor.se
+46 708 204 288


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From: Justin Uberti <juberti@google.com>
Date: Sat, 7 Feb 2015 10:24:22 -0800
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Subject: Re: [rtcweb] Use of redundancy and RFC 2119 language in draft-ietf-rtcweb-fec-00
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Thanks for the quick review and comments. I'll incorporate this into the
-01 version I am working on.

On Fri, Feb 6, 2015 at 11:49 PM, Gunnar Hellstr=C3=B6m <
gunnar.hellstrom@omnitor.se> wrote:

> I have two comments to draft-ietf-rtcweb-fec-00
>
> 1. Use of redundancy.
> Section 3.2 "Redundant encoding" informs about RFC 2198 .
> The last sentence says "This approach is also only applicable to audio
> content."
> RFC 2198 redundancy is successfully used in SIP and RTP for real-time tex=
t
> content (RFC 4103), because of its robustness, very low cost in bandwidth
> for low bandwidth media, and no risk for excessive delays.
> Chapter 3 seems to be a general overview of available methods without
> specific scope to WebRTC.
> Therefore, I suggest to either replace the sentence
>
> "This approach is also only applicable to audio content."
> with
> "This approach is also not applicable to video content."
> or delete it.
>
> 2. Use of SHOULD and MUST in abstract and introduction
> There are a couple of places in the document where the use of "must" and
> "should" may be confusing and should be avoided.
>
> The last sentence in the introduction says:
> "This document describes what FEC mechanisms should be used by WebRTC
> client implementations."
> There is a similar sentence in the abstract.
> There is also a "must" in the first sentence of the introduction, even if
> it is not thought to be normative.
> In the document body, there are varying normative levels on the different
> methods.
> Therefore I suggest to avoid the must and should in the abstract and
> introduction.
>
> 2.1 The last sentence of the introduction can be changed to:
> "This document provides information and requirements on the use of FEC
> mechanisms by WebRTC client implementations."
>
> 2.2 The abstract could be changed from:
> "This document makes recommendations for how Forward Error Correction
> (FEC) should be used by WebRTC applications."
> to
> "This document provides information and requirements on the use of Forwar=
d
> Error Correction (FEC) mechanisms by WebRTC applications."
>
> 2.3 The first sentence in the introduction, starting:
> "In situations where packet loss is high, or media quality must be
> perfect,"
> could be changed to:
> "In situations where packet loss is high, or high media quality is
> essential,"
> to avoid the use of "must" that apparently is not intended to be normativ=
e.
>
> Regards
>
> Gunnar Hellstr=C3=B6m
>
>
>
>
>
>
> internet-drafts@ietf.org skrev den 2015-02-06 17:56:
>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>   This draft is a work item of the Real-Time Communication in
>> WEB-browsers Working Group of the IETF.
>>
>>          Title           : WebRTC Forward Error Correction Requirements
>>          Author          : Justin Uberti
>>         Filename        : draft-ietf-rtcweb-fec-00.txt
>>         Pages           : 7
>>         Date            : 2015-02-05
>>
>> Abstract:
>>     This document makes recommendations for how Forward Error Correction
>>     (FEC) should be used by WebRTC applications.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00
>>
>>
>> Please note that it may take a couple of minutes from the time of
>> submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> _______________________________________________
>> I-D-Announce mailing list
>> I-D-Announce@ietf.org
>> https://www.ietf.org/mailman/listinfo/i-d-announce
>> Internet-Draft directories: http://www.ietf.org/shadow.html
>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>
>
> --
> -----------------------------------------
> Gunnar Hellstr=C3=B6m
> Omnitor
> gunnar.hellstrom@omnitor.se
> +46 708 204 288
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Thanks for the quick review and comments. I&#39;ll incorpo=
rate this into the -01 version I am working on.</div><div class=3D"gmail_ex=
tra"><br><div class=3D"gmail_quote">On Fri, Feb 6, 2015 at 11:49 PM, Gunnar=
 Hellstr=C3=B6m <span dir=3D"ltr">&lt;<a href=3D"mailto:gunnar.hellstrom@om=
nitor.se" target=3D"_blank">gunnar.hellstrom@omnitor.se</a>&gt;</span> wrot=
e:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex">I have two comments to draft-ietf-rtcw=
eb-fec-00<br>
<br>
1. Use of redundancy.<br>
Section 3.2 &quot;Redundant encoding&quot; informs about RFC 2198 .<br>
The last sentence says &quot;This approach is also only applicable to audio=
 content.&quot;<br>
RFC 2198 redundancy is successfully used in SIP and RTP for real-time text =
content (RFC 4103), because of its robustness, very low cost in bandwidth f=
or low bandwidth media, and no risk for excessive delays.<br>
Chapter 3 seems to be a general overview of available methods without speci=
fic scope to WebRTC.<br>
Therefore, I suggest to either replace the sentence<br>
<br>
&quot;This approach is also only applicable to audio content.&quot;<br>
with<br>
&quot;This approach is also not applicable to video content.&quot;<br>
or delete it.<br>
<br>
2. Use of SHOULD and MUST in abstract and introduction<br>
There are a couple of places in the document where the use of &quot;must&qu=
ot; and &quot;should&quot; may be confusing and should be avoided.<br>
<br>
The last sentence in the introduction says:<br>
&quot;This document describes what FEC mechanisms should be used by WebRTC =
client implementations.&quot;<br>
There is a similar sentence in the abstract.<br>
There is also a &quot;must&quot; in the first sentence of the introduction,=
 even if it is not thought to be normative.<br>
In the document body, there are varying normative levels on the different m=
ethods.<br>
Therefore I suggest to avoid the must and should in the abstract and introd=
uction.<br>
<br>
2.1 The last sentence of the introduction can be changed to:<br>
&quot;This document provides information and requirements on the use of FEC=
 mechanisms by WebRTC client implementations.&quot;<br>
<br>
2.2 The abstract could be changed from:<br>
&quot;This document makes recommendations for how Forward Error Correction =
(FEC) should be used by WebRTC applications.&quot;<br>
to<br>
&quot;This document provides information and requirements on the use of For=
ward Error Correction (FEC) mechanisms by WebRTC applications.&quot;<br>
<br>
2.3 The first sentence in the introduction, starting:<br>
&quot;In situations where packet loss is high, or media quality must be per=
fect,&quot;<br>
could be changed to:<br>
&quot;In situations where packet loss is high, or high media quality is ess=
ential,&quot;<br>
to avoid the use of &quot;must&quot; that apparently is not intended to be =
normative.<br>
<br>
Regards<br>
<br>
Gunnar Hellstr=C3=B6m<br>
<br>
<br>
<br>
<br>
<br>
<br>
<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">internet-draf=
ts@ietf.org</a> skrev den 2015-02-06 17:56:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0 This draft is a work item of the Real-Time Communication in WEB-brow=
sers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0: WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
: Justin Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-02-05<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0 This document makes recommendations for how Forward Error Cor=
rection<br>
=C2=A0 =C2=A0 (FEC) should be used by WebRTC applications.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/<u></u>doc/draft-ietf-rtcweb-fec/<=
/a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00" target=3D"_=
blank">http://tools.ietf.org/html/<u></u>draft-ietf-rtcweb-fec-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-<u></u>drafts/</a><br>
<br>
______________________________<u></u>_________________<br>
I-D-Announce mailing list<br>
<a href=3D"mailto:I-D-Announce@ietf.org" target=3D"_blank">I-D-Announce@iet=
f.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce" target=3D"_b=
lank">https://www.ietf.org/mailman/<u></u>listinfo/i-d-announce</a><br>
Internet-Draft directories: <a href=3D"http://www.ietf.org/shadow.html" tar=
get=3D"_blank">http://www.ietf.org/shadow.<u></u>html</a><br>
or <a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" target=3D"_blank">=
ftp://ftp.ietf.org/ietf/<u></u>1shadow-sites.txt</a><span class=3D"HOEnZb">=
<font color=3D"#888888"><br>
</font></span></blockquote><span class=3D"HOEnZb"><font color=3D"#888888">
<br>
-- <br>
------------------------------<u></u>-----------<br>
Gunnar Hellstr=C3=B6m<br>
Omnitor<br>
<a href=3D"mailto:gunnar.hellstrom@omnitor.se" target=3D"_blank">gunnar.hel=
lstrom@omnitor.se</a><br>
<a href=3D"tel:%2B46%20708%20204%20288" value=3D"+46708204288" target=3D"_b=
lank">+46 708 204 288</a><br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>

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From: Barry Dingle <btdingle@gmail.com>
Date: Sun, 8 Feb 2015 12:44:34 +1100
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The new "WebRTC Forward Error Correction Requirements" draft refers solely
to 'media' FEC requirements.

It does not refer to non-media (or DataChannel) uses of FEC.

If this draft is intended to concentrate on the media FEC requirements,
then the Title should be changed to reflect that, namely:

"WebRTC *Media* Forward Error Correction Requirements",

If it intended to also cover DataChannel uses of FEC then the draft should
have a new section after Section 5 with a Heading of "FEC for DataChannel
Content".

/Barry Dingle



On Sat, Feb 7, 2015 at 3:56 AM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : WebRTC Forward Error Correction Requirements
>         Author          : Justin Uberti
>         Filename        : draft-ietf-rtcweb-fec-00.txt
>         Pages           : 7
>         Date            : 2015-02-05
>
> Abstract:
>    This document makes recommendations for how Forward Error Correction
>    (FEC) should be used by WebRTC applications.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><div><div>The new &quot;WebRTC Forward Error Correction Re=
quirements&quot; draft refers solely to &#39;media&#39; FEC requirements. <=
br><br></div>It does not refer to non-media (or DataChannel) uses of FEC. <=
br><br></div>If this draft is intended to concentrate on the media FEC requ=
irements, then the Title should be changed to reflect that, namely:<br><br>=
<div><div><div><div><div class=3D"gmail_extra">&quot;WebRTC <b>Media</b> Fo=
rward Error Correction Requirements&quot;,<br><br></div><div class=3D"gmail=
_extra">If it intended to also cover DataChannel uses of FEC then the draft=
 should have a new section after Section 5 with a Heading of &quot;FEC for =
DataChannel Content&quot;. <br>=C2=A0<br></div><div class=3D"gmail_extra"><=
div><div class=3D"gmail_signature"><div dir=3D"ltr">/Barry Dingle<br><br><b=
r></div></div></div>
<br><div class=3D"gmail_quote">On Sat, Feb 7, 2015 at 3:56 AM,  <span dir=
=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">=
internet-drafts@ietf.org</a>&gt;</span> wrote:<br><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"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-02-05<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document makes recommendations for how Forward Error Corr=
ection<br>
=C2=A0 =C2=A0(FEC) should be used by WebRTC applications.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div></div></div></div></div>

--089e01494026e4e118050e89ccc1--


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From: Justin Uberti <juberti@google.com>
Date: Sat, 7 Feb 2015 22:17:49 -0800
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-fec-00.txt
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--001a113ec9282637e4050e8d9e63
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Interesting point. I don't think the WG has really discussed data FEC in
detail yet, so your title suggestion makes sense to me.

On Sat, Feb 7, 2015 at 5:44 PM, Barry Dingle <btdingle@gmail.com> wrote:

> The new "WebRTC Forward Error Correction Requirements" draft refers solely
> to 'media' FEC requirements.
>
> It does not refer to non-media (or DataChannel) uses of FEC.
>
> If this draft is intended to concentrate on the media FEC requirements,
> then the Title should be changed to reflect that, namely:
>
> "WebRTC *Media* Forward Error Correction Requirements",
>
> If it intended to also cover DataChannel uses of FEC then the draft should
> have a new section after Section 5 with a Heading of "FEC for DataChannel
> Content".
>
> /Barry Dingle
>
>
>
> On Sat, Feb 7, 2015 at 3:56 AM, <internet-drafts@ietf.org> wrote:
>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>  This draft is a work item of the Real-Time Communication in WEB-browsers
>> Working Group of the IETF.
>>
>>         Title           : WebRTC Forward Error Correction Requirements
>>         Author          : Justin Uberti
>>         Filename        : draft-ietf-rtcweb-fec-00.txt
>>         Pages           : 7
>>         Date            : 2015-02-05
>>
>> Abstract:
>>    This document makes recommendations for how Forward Error Correction
>>    (FEC) should be used by WebRTC applications.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00
>>
>>
>> Please note that it may take a couple of minutes from the time of
>> submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a113ec9282637e4050e8d9e63
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<div dir=3D"ltr">Interesting point. I don&#39;t think the WG has really dis=
cussed data FEC in detail yet, so your title suggestion makes sense to me.<=
/div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sat, Feb =
7, 2015 at 5:44 PM, Barry Dingle <span dir=3D"ltr">&lt;<a href=3D"mailto:bt=
dingle@gmail.com" target=3D"_blank">btdingle@gmail.com</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div>The new &quo=
t;WebRTC Forward Error Correction Requirements&quot; draft refers solely to=
 &#39;media&#39; FEC requirements. <br><br></div>It does not refer to non-m=
edia (or DataChannel) uses of FEC. <br><br></div>If this draft is intended =
to concentrate on the media FEC requirements, then the Title should be chan=
ged to reflect that, namely:<br><br><div><div><div><div><div class=3D"gmail=
_extra">&quot;WebRTC <b>Media</b> Forward Error Correction Requirements&quo=
t;,<br><br></div><div class=3D"gmail_extra">If it intended to also cover Da=
taChannel uses of FEC then the draft should have a new section after Sectio=
n 5 with a Heading of &quot;FEC for DataChannel Content&quot;. <br><span cl=
ass=3D"HOEnZb"><font color=3D"#888888">=C2=A0<br></font></span></div><div c=
lass=3D"gmail_extra"><span class=3D"HOEnZb"><font color=3D"#888888"><div><d=
iv><div dir=3D"ltr">/Barry Dingle<br><br><br></div></div></div></font></spa=
n><div><div class=3D"h5">
<br><div class=3D"gmail_quote">On Sat, Feb 7, 2015 at 3:56 AM,  <span dir=
=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">=
internet-drafts@ietf.org</a>&gt;</span> wrote:<br><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"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-02-05<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document makes recommendations for how Forward Error Corr=
ection<br>
=C2=A0 =C2=A0(FEC) should be used by WebRTC applications.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div></div></div></div></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a113ec9282637e4050e8d9e63--


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Justin Uberti wrote:
> Interesting point. I don't think the WG has really discussed data FEC in
> detail yet, so your title suggestion makes sense to me.

Wouldn't we just use SCTP retransmissions? Something more sophisticated 
than that seems like it would require application knowledge, so it 
should probably be handled at the application layer.


From nobody Sun Feb  8 05:11:09 2015
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Justin Uberti wrote:

> Interesting point. I don't think the WG has really discussed data FEC in
>> detail yet, so your title suggestion makes sense to me.
>>
>
> Wouldn't we just use SCTP retransmissions? Something more sophisticated
> than that seems like it would require application knowledge, so it should
> probably be handled at the application layer.
>
> For non-realtime data - Yes.

For realtime data applications - such as realtime text - with low delay
requirements from text entry a Sender to display at Receiver, you often
need FEC particularly on wireless connections.

/Barry Dingle

--001a11333d46b9be17050e936267
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">Justin Uberti wrote:<br>
<div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex"><span class=3D""><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Interesting point. I don&#39;t think the WG has really discussed data FEC i=
n<br>
detail yet, so your title suggestion makes sense to me.<br>
</blockquote>
<br></span>
Wouldn&#39;t we just use SCTP retransmissions? Something more sophisticated=
 than that seems like it would require application knowledge, so it should =
probably be handled at the application layer.<div class=3D"HOEnZb"><div cla=
ss=3D"h5"><br>
</div></div></blockquote></div>For non-realtime data - Yes. <br><br></div><=
div class=3D"gmail_extra">For realtime data applications - such as realtime=
 text - with low delay requirements from text entry a Sender to display at =
Receiver, you often need FEC particularly on wireless connections. <br><br>=
</div><div class=3D"gmail_extra">/Barry Dingle<br></div></div>

--001a11333d46b9be17050e936267--


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Barry Dingle <btdingle@gmail.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-fec-00.txt
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--_000_1F48239078314A728B5D2A11B8A01CA6ciscocom_
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Media FEC is necessary within WebRTC internals because apps have no access =
to the RTP media. Apps obviously have full access to data channel messages,=
 as well as reliability settings, so app layer FEC is always possible. It m=
ay be useful to extend SCTP reliability settings with FEC options, but that=
 would need new tsvwg work before any rtcweb work. I would not support norm=
ative dependencies on such work in this draft, although I would support the=
 new work.

Mo

On Feb 8, 2015, at 8:11 AM, "Barry Dingle" <btdingle@gmail.com<mailto:btdin=
gle@gmail.com>> wrote:

Justin Uberti wrote:
Interesting point. I don't think the WG has really discussed data FEC in
detail yet, so your title suggestion makes sense to me.

Wouldn't we just use SCTP retransmissions? Something more sophisticated tha=
n that seems like it would require application knowledge, so it should prob=
ably be handled at the application layer.

For non-realtime data - Yes.

For realtime data applications - such as realtime text - with low delay req=
uirements from text entry a Sender to display at Receiver, you often need F=
EC particularly on wireless connections.

/Barry Dingle
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

--_000_1F48239078314A728B5D2A11B8A01CA6ciscocom_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>Media FEC is necessary within WebRTC internals because apps have no ac=
cess to the RTP media. Apps obviously have full access to data channel mess=
ages, as well as reliability settings, so app layer FEC is always possible.=
 It may be useful to extend SCTP
 reliability settings with FEC options, but that would need new tsvwg work =
before any rtcweb work. I would not support normative dependencies on such =
work in this draft, although I would support the new work.&nbsp;</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
On Feb 8, 2015, at 8:11 AM, &quot;Barry Dingle&quot; &lt;<a href=3D"mailto:=
btdingle@gmail.com">btdingle@gmail.com</a>&gt; wrote:<br>
<br>
</div>
<div>
<div dir=3D"ltr">Justin Uberti wrote:<br>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<span class=3D"">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Interesting point. I don't think the WG has really discussed data FEC in<br=
>
detail yet, so your title suggestion makes sense to me.<br>
</blockquote>
<br>
</span>Wouldn't we just use SCTP retransmissions? Something more sophistica=
ted than that seems like it would require application knowledge, so it shou=
ld probably be handled at the application layer.
<div class=3D"HOEnZb">
<div class=3D"h5"><br>
</div>
</div>
</blockquote>
</div>
For non-realtime data - Yes. <br>
<br>
</div>
<div class=3D"gmail_extra">For realtime data applications - such as realtim=
e text - with low delay requirements from text entry a Sender to display at=
 Receiver, you often need FEC particularly on wireless connections.
<br>
<br>
</div>
<div class=3D"gmail_extra">/Barry Dingle<br>
</div>
</div>
</div>
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</body>
</html>

--_000_1F48239078314A728B5D2A11B8A01CA6ciscocom_--


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        Authors         : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-22.txt
	Pages           : 45
	Date            : 2015-02-09

Abstract:
   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc.  between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-22


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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This version changes the reference to draft-uberti-rtcweb-fec to a =
reference to draft-ietf-rtcweb-fec.=20

Colin



On 9 Feb 2015, at 11:19, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>        Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
>        Authors         : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-22.txt
> 	Pages           : 45
> 	Date            : 2015-02-09
>=20
> Abstract:
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc.  between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-22
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-22
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
https://csperkins.org/





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I just wondering what kind of FEC is recommended (or will be supported) for
G.711. The draft does not recommend neither redundant encoding nor a
separate FEC stream.

Jozsef

Jozsef Vass
SDK Engineering
[image: twilio] <http://www.twilio.com/?utm_source=email_signature>
MOBILE(415) 216-6603EMAILjvass@twilio.com
SIGNAL: The modern communications conference
<http://signal.twilio.com/?utm_source=employee_signature&utm_medium=email&utm_content=signal>
May 19/20 2015, Fort Mason Festival Pavilion, SF, CA
<http://signal.twilio.com/?utm_source=employee_signature&utm_medium=email&utm_content=signal>
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On Fri, Feb 6, 2015 at 8:56 AM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : WebRTC Forward Error Correction Requirements
>         Author          : Justin Uberti
>         Filename        : draft-ietf-rtcweb-fec-00.txt
>         Pages           : 7
>         Date            : 2015-02-05
>
> Abstract:
>    This document makes recommendations for how Forward Error Correction
>    (FEC) should be used by WebRTC applications.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a1140e638271f51050ed2f615
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<div dir=3D"ltr">I just wondering what kind of FEC is recommended (or will =
be supported) for G.711. The draft does not recommend neither redundant enc=
oding nor a separate FEC stream.<div><br></div><div>Jozsef</div></div><div =
class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_signature"=
><div dir=3D"ltr"><div style=3D"color:rgb(45,44,42);font-family:&#39;Helvet=
ica Neue&#39;,Helvetica,Arial,sans-serif;font-size:1em;line-height:20px;pad=
ding-bottom:2px">Jozsef Vass</div><div style=3D"font-family:&#39;Helvetica =
Neue&#39;,Helvetica,Arial,sans-serif;font-size:0.95em;padding-bottom:15px;c=
olor:rgb(178,173,162)">SDK Engineering</div><div style=3D"color:rgb(45,44,4=
2);font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;font-siz=
e:14px;line-height:20px;padding-bottom:20px"><a href=3D"http://www.twilio.c=
om/?utm_source=3Demail_signature" style=3D"color:red;text-decoration:none;f=
ont-size:32px;border:none;outline:none;background:0px 0px" target=3D"_blank=
"><img src=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Compan=
y-Signature/img/twilio-logo.png" width=3D"140" height=3D"42" alt=3D"twilio"=
 style=3D"border:0px;vertical-align:middle"></a></div><div style=3D"color:r=
gb(45,44,42);font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-seri=
f;font-size:14px;line-height:20px;padding-bottom:20px"><table style=3D"bord=
er-spacing:0px;border-collapse:collapse;background-color:transparent"><tbod=
y><tr><td width=3D"60" style=3D"padding:4px 0px 5px;border:none;color:rgb(1=
78,173,162);font-size:11px;line-height:12px;text-transform:uppercase;vertic=
al-align:middle">MOBILE</td><td style=3D"padding:0px;font-size:12px"><a hre=
f=3D"tel:(415)+216-6603" style=3D"color:rgb(45,44,42)!important;text-decora=
tion:none!important;font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sa=
ns-serif;background:0px 0px" target=3D"_blank">(415) 216-6603</a></td></tr>=
<tr><td width=3D"60" style=3D"padding:4px 0px 5px;border:none;color:rgb(178=
,173,162);font-size:11px;line-height:12px;text-transform:uppercase;vertical=
-align:middle">EMAIL</td><td style=3D"padding:0px;font-size:12px"><a href=
=3D"mailto:jvass@twilio.com" style=3D"color:rgb(45,44,42)!important;text-de=
coration:none!important;font-family:&#39;Helvetica Neue&#39;,Helvetica,Aria=
l,sans-serif;background:0px 0px" target=3D"_blank">jvass@twilio.com</a></td=
></tr></tbody></table></div><div style=3D"font-family:&#39;Helvetica Neue&#=
39;,Helvetica,Arial,sans-serif;color:rgb(178,173,162);font-size:12px;paddin=
g-bottom:4px"><a href=3D"http://signal.twilio.com/?utm_source=3Demployee_si=
gnature&amp;utm_medium=3Demail&amp;utm_content=3Dsignal" style=3D"color:rgb=
(178,173,162);text-decoration:none!important;background:0px 0px" target=3D"=
_blank"><span style=3D"color:rgb(235,100,97);font-weight:900">S</span><span=
 style=3D"color:rgb(255,64,109);font-weight:900">I</span><span style=3D"col=
or:rgb(214,67,117);font-weight:900">G</span><span style=3D"color:rgb(122,99=
,126);font-weight:900">N</span><span style=3D"color:rgb(66,99,113);font-wei=
ght:900">A</span><span style=3D"color:rgb(43,154,137);font-weight:900">L</s=
pan>: The modern communications conference</a></div><div style=3D"font-fami=
ly:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;color:rgb(178,173,16=
2);font-size:12px;padding-bottom:20px"><a href=3D"http://signal.twilio.com/=
?utm_source=3Demployee_signature&amp;utm_medium=3Demail&amp;utm_content=3Ds=
ignal" style=3D"color:rgb(178,173,162);text-decoration:none!important;backg=
round:0px 0px" target=3D"_blank">May 19/20 2015, Fort Mason Festival Pavili=
on, SF, CA</a></div><div style=3D"color:rgb(45,44,42);font-family:&#39;Helv=
etica Neue&#39;,Helvetica,Arial,sans-serif;font-size:14px;line-height:20px;=
padding-bottom:20px"><table style=3D"border-spacing:0px;border-collapse:col=
lapse;background-color:transparent"><tbody><tr><td style=3D"padding:12px 16=
px 12px 12px;border-style:solid;border-color:rgb(216,214,208);border-width:=
2px 0px"><a href=3D"https://www.twilio.com/" style=3D"color:rgb(66,139,202)=
;text-decoration:none;background:0px 0px" target=3D"_blank"><img src=3D"htt=
ps://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signature/img/i=
con-twilio.png" width=3D"20" style=3D"border:0px;vertical-align:middle"></a=
></td><td style=3D"padding:12px 16px;border-style:solid;border-color:rgb(21=
6,214,208);border-width:2px 0px"><a href=3D"https://www.facebook.com/TeamTw=
ilio" style=3D"color:rgb(66,139,202);text-decoration:none;background:0px 0p=
x" target=3D"_blank"><img src=3D"https://s3.amazonaws.com/ahoy-assets.twili=
o.com/Emails/Company-Signature/img/icon-facebook.png" width=3D"20" style=3D=
"border:0px;vertical-align:middle"></a></td><td style=3D"padding:12px 16px;=
border-style:solid;border-color:rgb(216,214,208);border-width:2px 0px"><a h=
ref=3D"https://plus.google.com/u/0/+twilio/posts" style=3D"color:rgb(66,139=
,202);text-decoration:none;background:0px 0px" target=3D"_blank"><img src=
=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signatur=
e/img/icon-google.png" width=3D"20" style=3D"border:0px;vertical-align:midd=
le"></a></td><td style=3D"padding:12px 16px;border-style:solid;border-color=
:rgb(216,214,208);border-width:2px 0px"><a href=3D"https://twitter.com/twil=
io" style=3D"color:rgb(66,139,202);text-decoration:none;background:0px 0px"=
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com/Emails/Company-Signature/img/icon-twitter.png" width=3D"20" style=3D"bo=
rder:0px;vertical-align:middle"></a></td><td style=3D"padding:12px 16px;bor=
der-style:solid;border-color:rgb(216,214,208);border-width:2px 0px"><a href=
=3D"https://www.linkedin.com/company/twilio-inc." style=3D"color:rgb(66,139=
,202);text-decoration:none;background:0px 0px" target=3D"_blank"><img src=
=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signatur=
e/img/icon-linkedin.png" width=3D"20" style=3D"border:0px;vertical-align:mi=
ddle"></a></td><td style=3D"padding:12px 12px 12px 16px;border-style:solid;=
border-color:rgb(216,214,208);border-width:2px 0px"><a href=3D"http://www.g=
lassdoor.com/Overview/Working-at-Twilio-EI_IE410790.11,17.htm" style=3D"col=
or:rgb(66,139,202);text-decoration:none;background:0px 0px" target=3D"_blan=
k"><img src=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Compa=
ny-Signature/img/icon-glassdoor.png" width=3D"20" style=3D"border:0px;verti=
cal-align:middle"></a></td></tr></tbody></table></div></div></div></div>
<br><div class=3D"gmail_quote">On Fri, Feb 6, 2015 at 8:56 AM,  <span dir=
=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">=
internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 7<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-02-05<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document makes recommendations for how Forward Error Corr=
ection<br>
=C2=A0 =C2=A0(FEC) should be used by WebRTC applications.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" target=
=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00" target=3D"_=
blank">http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--001a1140e638271f51050ed2f615--


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On Wed, Feb 11, 2015 at 12:01 PM, Jozsef Vass <jvass@twilio.com> wrote:

> I just wondering what kind of FEC is recommended (or will be supported)
> for G.711. The draft does not recommend neither redundant encoding nor a
> separate FEC stream.


Isn't this what you're looking for?

3.1 <http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00#section-3.1>.
Separate FEC Stream

   This approach, as described in [RFC5956], Section 4.3
<http://tools.ietf.org/html/rfc5956#section-4.3>, sends FEC
   packets as an independent SSRC-multiplexed stream, with its own SSRC
   and payload type.  While by far the most flexible, each FEC packet
   will have its own IP+UDP+RTP+FEC header, leading to additional
   overhead of the FEC stream.


Simon

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Wed, Feb 11, 2015 at 12:01 PM, Jozsef Vass <span dir=3D"ltr">&lt;<a href=
=3D"mailto:jvass@twilio.com" target=3D"_blank">jvass@twilio.com</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">I just wondering what kind of FEC is recomme=
nded (or will be supported) for G.711. The draft does not recommend neither=
 redundant encoding nor a separate FEC stream.</blockquote><div><br></div><=
div>Isn&#39;t this what you&#39;re looking for?=C2=A0</div></div><br><pre c=
lass=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb=
(0,0,0)"><span class=3D"" style=3D"line-height:0pt;display:inline;font-size=
:1em;font-weight:bold"><h3 style=3D"line-height:0pt;display:inline;font-siz=
e:1em"><a class=3D"" name=3D"section-3.1" href=3D"http://tools.ietf.org/htm=
l/draft-ietf-rtcweb-fec-00#section-3.1" style=3D"color:black;text-decoratio=
n:none">3.1</a>.  Separate FEC Stream</h3></span>

   This approach, as described in <a href=3D"http://tools.ietf.org/html/rfc=
5956#section-4.3">[RFC5956], Section=C2=A04.3</a>, sends FEC
   packets as an independent SSRC-multiplexed stream, with its own SSRC
   and payload type.  While by far the most flexible, each FEC packet
   will have its own IP+UDP+RTP+FEC header, leading to additional
   overhead of the FEC stream.</pre><pre class=3D"" style=3D"font-size:1em;=
margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre>Simon</div></d=
iv>

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From: Jozsef Vass <jvass@twilio.com>
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Later, in 4.1 Recommended Mechanisms:

   When using constant-bitrate codecs, e.g.  PCMU, use of [RFC2198
<http://tools.ietf.org/html/rfc2198>]
   redundant encoding is NOT RECOMMENDED, as this will result in a
   potentially significant bitrate increase.  Furthermore, suddenly
   increasing the bitrate to deal with packet losses may actually make
   things worse.

   Because of the lower packet rate of audio encodings, usually a single
   packet per frame, use of a separate FEC stream comes with a higher
   overhead than other mechanisms, and therefore is NOT RECOMMENDED.


So what is recommended for PCMU?

Jozsef


Jozsef Vass
SDK Engineering
[image: twilio] <http://www.twilio.com/?utm_source=email_signature>
MOBILE(415) 216-6603EMAILjvass@twilio.com
SIGNAL: The modern communications conference
<http://signal.twilio.com/?utm_source=employee_signature&utm_medium=email&utm_content=signal>
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<https://www.linkedin.com/company/twilio-inc.>
<http://www.glassdoor.com/Overview/Working-at-Twilio-EI_IE410790.11,17.htm>

On Wed, Feb 11, 2015 at 11:39 AM, Simon Perreault <sperreault@jive.com>
wrote:

>
> On Wed, Feb 11, 2015 at 12:01 PM, Jozsef Vass <jvass@twilio.com> wrote:
>
>> I just wondering what kind of FEC is recommended (or will be supported)
>> for G.711. The draft does not recommend neither redundant encoding nor a
>> separate FEC stream.
>
>
> Isn't this what you're looking for?
>
> 3.1 <http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00#section-3.1>.  Separate FEC Stream
>
>    This approach, as described in [RFC5956], Section 4.3 <http://tools.ietf.org/html/rfc5956#section-4.3>, sends FEC
>    packets as an independent SSRC-multiplexed stream, with its own SSRC
>    and payload type.  While by far the most flexible, each FEC packet
>    will have its own IP+UDP+RTP+FEC header, leading to additional
>    overhead of the FEC stream.
>
>
> Simon
>

--089e015386d6c62995050ed53c4f
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<div dir=3D"ltr">Later, in 4.1 Recommended Mechanisms:<div><br></div><div><=
pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;colo=
r:rgb(0,0,0)">   When using constant-bitrate codecs, e.g.  PCMU, use of [<a=
 href=3D"http://tools.ietf.org/html/rfc2198" title=3D"&quot;RTP Payload for=
 Redundant Audio Data&quot;">RFC2198</a>]
   redundant encoding is NOT RECOMMENDED, as this will result in a
   potentially significant bitrate increase.  Furthermore, suddenly
   increasing the bitrate to deal with packet losses may actually make
   things worse.

   Because of the lower packet rate of audio encodings, usually a single
   packet per frame, use of a separate FEC stream comes with a higher
   overhead than other mechanisms, and therefore is NOT RECOMMENDED.</pre><=
div><br></div><div>So what is recommended for PCMU?</div><div><br></div><di=
v>Jozsef</div><div><br></div></div></div><div class=3D"gmail_extra"><br cle=
ar=3D"all"><div><div class=3D"gmail_signature"><div dir=3D"ltr"><div style=
=3D"color:rgb(45,44,42);font-family:&#39;Helvetica Neue&#39;,Helvetica,Aria=
l,sans-serif;font-size:1em;line-height:20px;padding-bottom:2px">Jozsef Vass=
</div><div style=3D"font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sa=
ns-serif;font-size:0.95em;padding-bottom:15px;color:rgb(178,173,162)">SDK E=
ngineering</div><div style=3D"color:rgb(45,44,42);font-family:&#39;Helvetic=
a Neue&#39;,Helvetica,Arial,sans-serif;font-size:14px;line-height:20px;padd=
ing-bottom:20px"><a href=3D"http://www.twilio.com/?utm_source=3Demail_signa=
ture" style=3D"color:red;text-decoration:none;font-size:32px;border:none;ou=
tline:none;background:0px 0px" target=3D"_blank"><img src=3D"https://s3.ama=
zonaws.com/ahoy-assets.twilio.com/Emails/Company-Signature/img/twilio-logo.=
png" width=3D"140" height=3D"42" alt=3D"twilio" style=3D"border:0px;vertica=
l-align:middle"></a></div><div style=3D"color:rgb(45,44,42);font-family:&#3=
9;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;font-size:14px;line-height=
:20px;padding-bottom:20px"><table style=3D"border-spacing:0px;border-collap=
se:collapse;background-color:transparent"><tbody><tr><td width=3D"60" style=
=3D"padding:4px 0px 5px;border:none;color:rgb(178,173,162);font-size:11px;l=
ine-height:12px;text-transform:uppercase;vertical-align:middle">MOBILE</td>=
<td style=3D"padding:0px;font-size:12px"><a href=3D"tel:(415)+216-6603" sty=
le=3D"color:rgb(45,44,42)!important;text-decoration:none!important;font-fam=
ily:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;background:0px 0px"=
 target=3D"_blank">(415) 216-6603</a></td></tr><tr><td width=3D"60" style=
=3D"padding:4px 0px 5px;border:none;color:rgb(178,173,162);font-size:11px;l=
ine-height:12px;text-transform:uppercase;vertical-align:middle">EMAIL</td><=
td style=3D"padding:0px;font-size:12px"><a href=3D"mailto:jvass@twilio.com"=
 style=3D"color:rgb(45,44,42)!important;text-decoration:none!important;font=
-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;background:0px =
0px" target=3D"_blank">jvass@twilio.com</a></td></tr></tbody></table></div>=
<div style=3D"font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-ser=
if;color:rgb(178,173,162);font-size:12px;padding-bottom:4px"><a href=3D"htt=
p://signal.twilio.com/?utm_source=3Demployee_signature&amp;utm_medium=3Dema=
il&amp;utm_content=3Dsignal" style=3D"color:rgb(178,173,162);text-decoratio=
n:none!important;background:0px 0px" target=3D"_blank"><span style=3D"color=
:rgb(235,100,97);font-weight:900">S</span><span style=3D"color:rgb(255,64,1=
09);font-weight:900">I</span><span style=3D"color:rgb(214,67,117);font-weig=
ht:900">G</span><span style=3D"color:rgb(122,99,126);font-weight:900">N</sp=
an><span style=3D"color:rgb(66,99,113);font-weight:900">A</span><span style=
=3D"color:rgb(43,154,137);font-weight:900">L</span>: The modern communicati=
ons conference</a></div><div style=3D"font-family:&#39;Helvetica Neue&#39;,=
Helvetica,Arial,sans-serif;color:rgb(178,173,162);font-size:12px;padding-bo=
ttom:20px"><a href=3D"http://signal.twilio.com/?utm_source=3Demployee_signa=
ture&amp;utm_medium=3Demail&amp;utm_content=3Dsignal" style=3D"color:rgb(17=
8,173,162);text-decoration:none!important;background:0px 0px" target=3D"_bl=
ank">May 19/20 2015, Fort Mason Festival Pavilion, SF, CA</a></div><div sty=
le=3D"color:rgb(45,44,42);font-family:&#39;Helvetica Neue&#39;,Helvetica,Ar=
ial,sans-serif;font-size:14px;line-height:20px;padding-bottom:20px"><table =
style=3D"border-spacing:0px;border-collapse:collapse;background-color:trans=
parent"><tbody><tr><td style=3D"padding:12px 16px 12px 12px;border-style:so=
lid;border-color:rgb(216,214,208);border-width:2px 0px"><a href=3D"https://=
www.twilio.com/" style=3D"color:rgb(66,139,202);text-decoration:none;backgr=
ound:0px 0px" target=3D"_blank"><img src=3D"https://s3.amazonaws.com/ahoy-a=
ssets.twilio.com/Emails/Company-Signature/img/icon-twilio.png" width=3D"20"=
 style=3D"border:0px;vertical-align:middle"></a></td><td style=3D"padding:1=
2px 16px;border-style:solid;border-color:rgb(216,214,208);border-width:2px =
0px"><a href=3D"https://www.facebook.com/TeamTwilio" style=3D"color:rgb(66,=
139,202);text-decoration:none;background:0px 0px" target=3D"_blank"><img sr=
c=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signatu=
re/img/icon-facebook.png" width=3D"20" style=3D"border:0px;vertical-align:m=
iddle"></a></td><td style=3D"padding:12px 16px;border-style:solid;border-co=
lor:rgb(216,214,208);border-width:2px 0px"><a href=3D"https://plus.google.c=
om/u/0/+twilio/posts" style=3D"color:rgb(66,139,202);text-decoration:none;b=
ackground:0px 0px" target=3D"_blank"><img src=3D"https://s3.amazonaws.com/a=
hoy-assets.twilio.com/Emails/Company-Signature/img/icon-google.png" width=
=3D"20" style=3D"border:0px;vertical-align:middle"></a></td><td style=3D"pa=
dding:12px 16px;border-style:solid;border-color:rgb(216,214,208);border-wid=
th:2px 0px"><a href=3D"https://twitter.com/twilio" style=3D"color:rgb(66,13=
9,202);text-decoration:none;background:0px 0px" target=3D"_blank"><img src=
=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signatur=
e/img/icon-twitter.png" width=3D"20" style=3D"border:0px;vertical-align:mid=
dle"></a></td><td style=3D"padding:12px 16px;border-style:solid;border-colo=
r:rgb(216,214,208);border-width:2px 0px"><a href=3D"https://www.linkedin.co=
m/company/twilio-inc." style=3D"color:rgb(66,139,202);text-decoration:none;=
background:0px 0px" target=3D"_blank"><img src=3D"https://s3.amazonaws.com/=
ahoy-assets.twilio.com/Emails/Company-Signature/img/icon-linkedin.png" widt=
h=3D"20" style=3D"border:0px;vertical-align:middle"></a></td><td style=3D"p=
adding:12px 12px 12px 16px;border-style:solid;border-color:rgb(216,214,208)=
;border-width:2px 0px"><a href=3D"http://www.glassdoor.com/Overview/Working=
-at-Twilio-EI_IE410790.11,17.htm" style=3D"color:rgb(66,139,202);text-decor=
ation:none;background:0px 0px" target=3D"_blank"><img src=3D"https://s3.ama=
zonaws.com/ahoy-assets.twilio.com/Emails/Company-Signature/img/icon-glassdo=
or.png" width=3D"20" style=3D"border:0px;vertical-align:middle"></a></td></=
tr></tbody></table></div></div></div></div>
<br><div class=3D"gmail_quote">On Wed, Feb 11, 2015 at 11:39 AM, Simon Perr=
eault <span dir=3D"ltr">&lt;<a href=3D"mailto:sperreault@jive.com" target=
=3D"_blank">sperreault@jive.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D=
"gmail_quote">On Wed, Feb 11, 2015 at 12:01 PM, Jozsef Vass <span dir=3D"lt=
r">&lt;<a href=3D"mailto:jvass@twilio.com" target=3D"_blank">jvass@twilio.c=
om</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,20=
4);border-left-style:solid;padding-left:1ex">I just wondering what kind of =
FEC is recommended (or will be supported) for G.711. The draft does not rec=
ommend neither redundant encoding nor a separate FEC stream.</blockquote><d=
iv><br></div><div>Isn&#39;t this what you&#39;re looking for?=C2=A0</div></=
div><br><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:=
rgb(0,0,0)"><span style=3D"line-height:0pt;display:inline;font-size:1em;fon=
t-weight:bold"><h3 style=3D"line-height:0pt;display:inline;font-size:1em"><=
a name=3D"14b7a271b939f06e_section-3.1" href=3D"http://tools.ietf.org/html/=
draft-ietf-rtcweb-fec-00#section-3.1" style=3D"color:black;text-decoration:=
none" target=3D"_blank">3.1</a>.  Separate FEC Stream</h3></span>

   This approach, as described in <a href=3D"http://tools.ietf.org/html/rfc=
5956#section-4.3" target=3D"_blank">[RFC5956], Section=C2=A04.3</a>, sends =
FEC
   packets as an independent SSRC-multiplexed stream, with its own SSRC
   and payload type.  While by far the most flexible, each FEC packet
   will have its own IP+UDP+RTP+FEC header, leading to additional
   overhead of the FEC stream.</pre><span class=3D"HOEnZb"><font color=3D"#=
888888"><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:=
rgb(0,0,0)"><br></pre>Simon</font></span></div></div>
</blockquote></div><br></div>

--089e015386d6c62995050ed53c4f--


From nobody Wed Feb 11 23:56:11 2015
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From: Suhas Nandakumar <suhasietf@gmail.com>
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Subject: [rtcweb] RTCWeb SDP Examples Draft
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Hello All

  We are looking to get feedback/thoughts on the RTCWeb Examples draft.

   Draft Location:
http://tools.ietf.org/id/draft-nandakumar-rtcweb-sdp-07.html
   (HTML version is more easier to read)

Below is the list of examples currently covered in this draft

5.2.  Basic Examples  . . . . . . . . . . . . . . . . . . . . .   8
       5.2.1.  Audio Only Session  . . . . . . . . . . . . . . . . .   8
       5.2.2.  Audio/Video Session . . . . . . . . . . . . . . . . .  11
       5.2.3.  Data Only Session . . . . . . . . . . . . . . . . . .  16
       5.2.4.  Audio Call On Hold  . . . . . . . . . . . . . . . . .  19
       5.2.5.  Audio with DTMF Session . . . . . . . . . . . . . . .  23
       5.2.6.  One Way Audio/Video Session - Document Camera . . . .  27
       5.2.7.  Audio, Video Session with BUNDLE Support Unknown  . .  31
       5.2.8.  Audio, Video and Data Session . . . . . . . . . . . .  38
       5.2.9.  Audio, Video Session with BUNDLE Unsupported  . . . .  43
       5.2.10. Audio, Video BUNDLED, but Data (Not BUNDLED)  . . . .  47
       5.2.11. Audio Only, Add Video to BUNDLE . . . . . . . . . . .  52
5.3.  MultiResolution, RTX, FEC Examples  . . . . . . . . . . .  59
       5.3.1.  Sendonly Simulcast Session with 2 cameras and 2
               encodings     per camera  . . . . . . . . . . . . . .  59
       5.3.2.  Successful SVC Video Session  . . . . . . . . . . . .  65
       5.3.3.  Successful Simulcast Video Session with
               Retransmission  . . . . . . . . . . . . . . . . . . .  70
       5.3.4.  Successful 1-way Simulcast Sessio with 2 resolutions
               and         RTX - One resolution rejected . . . . . .  74
       5.3.5.  Simulcast Video Session with Forward Error Correction  79
5.4.  Others  . . . . . . . . . . . . . . . . . . . . . . . . .  83
       5.4.1.  Audio Session - Voice Activity Detection  . . . . . .  83
       5.4.2.  Audio Conference - Voice Activity Detection . . . . .  86
       5.4.3.  Successful legacy Interop Fallaback with bundle-only   90



Given that it is quite impossible to cover all the possible combinations of
the use-cases , we would like to get the group's inputs on the following:

   - Do the examples capture the breadth of common RTCWeb use-cases ?
   - Are the examples clear enough in terms of text and SDP provided ?
   - Any new examples/use-cases that would be worthy of adding to the
existing draft ?
   - Is this draft helpful in meeting its said objective of Information
document ?


Look forward for the feedback.

Thanks in Advance
Suhas

--047d7bfeb34cb7836a050edf73de
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<div dir=3D"ltr">Hello All<div><br></div><div>=C2=A0 We are looking to get =
feedback/thoughts on the RTCWeb Examples draft.</div><div><br></div><div>=
=C2=A0 =C2=A0Draft Location: <a href=3D"http://tools.ietf.org/id/draft-nand=
akumar-rtcweb-sdp-07.html">http://tools.ietf.org/id/draft-nandakumar-rtcweb=
-sdp-07.html</a></div><div>=C2=A0 =C2=A0(HTML version is more easier to rea=
d)</div><div><br></div><div>Below is the list of examples currently covered=
 in this draft=C2=A0</div><div><pre style=3D"color:rgb(0,0,0);word-wrap:bre=
ak-word;white-space:pre-wrap"><pre style=3D"word-wrap:break-word;white-spac=
e:pre-wrap"><pre style=3D"word-wrap:break-word;white-space:pre-wrap">5.2.  =
Basic Examples  . . . . . . . . . . . . . . . . . . . . .   8
       5.2.1.  Audio Only Session  . . . . . . . . . . . . . . . . .   8
       5.2.2.  Audio/Video Session . . . . . . . . . . . . . . . . .  11
       5.2.3.  Data Only Session . . . . . . . . . . . . . . . . . .  16
       5.2.4.  Audio Call On Hold  . . . . . . . . . . . . . . . . .  19
       5.2.5.  Audio with DTMF Session . . . . . . . . . . . . . . .  23
       5.2.6.  One Way Audio/Video Session - Document Camera . . . .  27
       5.2.7.  Audio, Video Session with BUNDLE Support Unknown  . .  31
       5.2.8.  Audio, Video and Data Session . . . . . . . . . . . .  38
       5.2.9.  Audio, Video Session with BUNDLE Unsupported  . . . .  43
       5.2.10. Audio, Video BUNDLED, but Data (Not BUNDLED)  . . . .  47
       5.2.11. Audio Only, Add Video to BUNDLE . . . . . . . . . . .  52
5.3.  MultiResolution, RTX, FEC Examples  . . . . . . . . . . .  59
       5.3.1.  Sendonly Simulcast Session with 2 cameras and 2
               encodings     per camera  . . . . . . . . . . . . . .  59
       5.3.2.  Successful SVC Video Session  . . . . . . . . . . . .  65
       5.3.3.  Successful Simulcast Video Session with
               Retransmission  . . . . . . . . . . . . . . . . . . .  70
       5.3.4.  Successful 1-way Simulcast Sessio with 2 resolutions
               and         RTX - One resolution rejected . . . . . .  74
       5.3.5.  Simulcast Video Session with Forward Error Correction  79
5.4.  Others  . . . . . . . . . . . . . . . . . . . . . . . . .  83
       5.4.1.  Audio Session - Voice Activity Detection  . . . . . .  83
       5.4.2.  Audio Conference - Voice Activity Detection . . . . .  86
       5.4.3.  Successful legacy Interop Fallaback with bundle-only   90</p=
re></pre></pre></div><div><br></div><div><br></div><div>Given that it is qu=
ite impossible to cover all the possible combinations of the use-cases , we=
 would like to get the group&#39;s inputs on the following:</div><div><br><=
/div><div>=C2=A0 =C2=A0- Do the examples capture the breadth of common RTCW=
eb use-cases ?<br></div><div>=C2=A0 =C2=A0- Are the examples clear enough i=
n terms of text and SDP provided ?</div><div>=C2=A0 =C2=A0- Any new example=
s/use-cases that would be worthy of adding to the existing draft ?</div><di=
v>=C2=A0 =C2=A0- Is this draft helpful in meeting its said objective of Inf=
ormation document ?</div><div><br></div><div><br></div><div>Look forward fo=
r the feedback.</div><div><br></div><div>Thanks in Advance</div><div>Suhas<=
br></div><div>=C2=A0=C2=A0</div></div>

--047d7bfeb34cb7836a050edf73de--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] FW: SCTP-SDP and BUNDLE
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FYI,

I sent this to the MMUSIC list, but it may also be in interest to the RTCWE=
B community.

If you have any opinions regarding this, please post them on the MMUSIC lis=
t.

Regards,

Christer

From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christer Holmber=
g
Sent: 12. helmikuuta 2015 14:15
To: mmusic@ietf.org
Subject: [MMUSIC] SCTP-SDP and BUNDLE

Hi,

We may have discussed this before, but I'd like to verify.

According to the SCTP RFC, there can only be one SCTP association between t=
wo SCTP endpoints.

I ASSUME that applies also to DTLS/SCTP and SCTP/DTLS. If so, I would expli=
citly indicate that in the SCTP-SDP draft, as I have been asked to add a "B=
UNDLE considerations" section.

Regards,

Chrsiter

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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">FYI,<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 lang=3D"EN-US" style=3D"color:#1F497D">I sent =
this to the MMUSIC list, but it may also be in interest to the RTCWEB commu=
nity.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">If you =
have any opinions regarding this, please post them on the MMUSIC list.<o:p>=
</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">Regards=
,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">Christe=
r<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
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nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;mso-fareast-language:FI=
">From:</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-famil=
y:&quot;Tahoma&quot;,&quot;sans-serif&quot;;mso-fareast-language:FI"> mmusi=
c [mailto:mmusic-bounces@ietf.org]
<b>On Behalf Of </b>Christer Holmberg<br>
<b>Sent:</b> 12. helmikuuta 2015 14:15<br>
<b>To:</b> mmusic@ietf.org<br>
<b>Subject:</b> [MMUSIC] SCTP-SDP and BUNDLE<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">We may have discussed this befo=
re, but I&#8217;d like to verify.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">According to the SCTP RFC, ther=
e can only be one SCTP association between two SCTP endpoints.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I ASSUME that applies also to D=
TLS/SCTP and SCTP/DTLS. If so, I would explicitly indicate that in the SCTP=
-SDP draft, as I have been asked to add a &#8220;BUNDLE considerations&#822=
1; section.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Chrsiter<o:p></o:p></span></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D6CFDFDESESSMB209erics_--


From nobody Thu Feb 12 12:43:58 2015
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Date: Thu, 12 Feb 2015 14:43:54 -0600
From: Adam Roach <adam@nostrum.com>
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Subject: [rtcweb] Video Open issue #1: Color space
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Based on a lack of objection, I will be removing the "TODO" from the 
color space statement in section 3 of draft-ietf-rtcweb-video. I am also 
incorporating the suggestion to reference the ISO/IEC definition of 
media description code points for additional clarity. The revised text 
reads:

Unless specified otherwise by the SDP or codec, the color space SHOULD 
be sRGB
[SRGB].  For clarity, this the color space indicated by codepoint 1 from
"ColourPrimaries" as defined in [IEC23001-8].

I am removing this from my list of open issues.

/a


From nobody Thu Feb 12 12:48:58 2015
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Subject: [rtcweb] Video Open issue #4: VP8 filters
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In Toronto, there was a claim that specification that "bilinear" and 
"none" filters is superfluous, as they are already required as part of 
the VP8 spec. Since that time, I have repeatedly asked for a concrete 
pointer to the text that would make this unnecessary.

No one has come forth with such a pointer. Given the harmless nature of 
redundantly specifying this requirement on implementations and the lack 
of concrete evidence for its removal, I am leaving the text as-is and 
removing the open issue from my list.

/a


From nobody Thu Feb 12 13:00:54 2015
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On 02/12/2015 09:48 PM, Adam Roach wrote:
> In Toronto, there was a claim that specification that "bilinear" and 
> "none" filters is superfluous, as they are already required as part of 
> the VP8 spec. Since that time, I have repeatedly asked for a concrete 
> pointer to the text that would make this unnecessary.
>
> No one has come forth with such a pointer. Given the harmless nature 
> of redundantly specifying this requirement on implementations and the 
> lack of concrete evidence for its removal, I am leaving the text as-is 
> and removing the open issue from my list.

Please. This is not harmless.

The text under consideration is

6.1.  VP8

    For the VP8 codec, defined in [RFC6386], endpoints MUST support the
    payload formats defined in [I-D.ietf-payload-vp8].  In addition they
    MUST support the 'bilinear' and 'none' reconstruction filters.

The problem with the text is not that it specifies that one MUST support 
the "bilinear" and "none" filters.

The problem with the text is that in order for it to make any sense at 
all, one must believe that you can be conformant with RFC 6386 without 
the ability to decode all the specified reconstruction filters - that 
is, in order for the second sentence to make any sense at all, one must 
believe that filters are optional.

There is no hint in any place in RFC 6386 that a conformant decoder can 
be conformant without the ability to decode bitstreams using all the 
filter types. Filters are not optional for the decoder.

This is the point I'm objecting to.

Harald


From nobody Thu Feb 12 13:06:09 2015
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* Adam Roach wrote:
>Given the harmless nature of redundantly specifying this requirement

I do not see how that could be harmless, whatever the requirement.
-- 
Bjrn Hhrmann  mailto:bjoern@hoehrmann.de  http://bjoern.hoehrmann.de
D-10243 Berlin  PGP Pub. KeyID: 0xA4357E78  http://www.bjoernsworld.de
 Available for hire in Berlin (early 2015)   http://www.websitedev.de/ 


From nobody Thu Feb 12 13:25:38 2015
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--089e0122f58c6e3741050eeac275
Content-Type: text/plain; charset=UTF-8

Hi,

Why not remove the second sentence? It clearly introduces an implied
interpretation to RFC 6386 that is incorrect.

Mohammed

On Thu, Feb 12, 2015 at 11:00 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> On 02/12/2015 09:48 PM, Adam Roach wrote:
>
>> In Toronto, there was a claim that specification that "bilinear" and
>> "none" filters is superfluous, as they are already required as part of the
>> VP8 spec. Since that time, I have repeatedly asked for a concrete pointer
>> to the text that would make this unnecessary.
>>
>> No one has come forth with such a pointer. Given the harmless nature of
>> redundantly specifying this requirement on implementations and the lack of
>> concrete evidence for its removal, I am leaving the text as-is and removing
>> the open issue from my list.
>>
>
> Please. This is not harmless.
>
> The text under consideration is
>
> 6.1.  VP8
>
>    For the VP8 codec, defined in [RFC6386], endpoints MUST support the
>    payload formats defined in [I-D.ietf-payload-vp8].  In addition they
>    MUST support the 'bilinear' and 'none' reconstruction filters.
>
> The problem with the text is not that it specifies that one MUST support
> the "bilinear" and "none" filters.
>
> The problem with the text is that in order for it to make any sense at
> all, one must believe that you can be conformant with RFC 6386 without the
> ability to decode all the specified reconstruction filters - that is, in
> order for the second sentence to make any sense at all, one must believe
> that filters are optional.
>
> There is no hint in any place in RFC 6386 that a conformant decoder can be
> conformant without the ability to decode bitstreams using all the filter
> types. Filters are not optional for the decoder.
>
> This is the point I'm objecting to.
>
> Harald
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



-- 
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Email: mohammedsraad@raadtech.com

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<div dir=3D"ltr">Hi,<div><br></div><div>Why not remove the second sentence?=
 It clearly introduces an implied interpretation to RFC 6386 that is incorr=
ect.</div><div><br></div><div>Mohammed</div></div><div class=3D"gmail_extra=
"><br><div class=3D"gmail_quote">On Thu, Feb 12, 2015 at 11:00 PM, Harald A=
lvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" tar=
get=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><span class=3D"">On 02/12/2015 09:48 PM, Adam Roach wrote:=
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
In Toronto, there was a claim that specification that &quot;bilinear&quot; =
and &quot;none&quot; filters is superfluous, as they are already required a=
s part of the VP8 spec. Since that time, I have repeatedly asked for a conc=
rete pointer to the text that would make this unnecessary.<br>
<br>
No one has come forth with such a pointer. Given the harmless nature of red=
undantly specifying this requirement on implementations and the lack of con=
crete evidence for its removal, I am leaving the text as-is and removing th=
e open issue from my list.<br>
</blockquote>
<br></span>
Please. This is not harmless.<br>
<br>
The text under consideration is<br>
<br>
6.1.=C2=A0 VP8<br>
<br>
=C2=A0 =C2=A0For the VP8 codec, defined in [RFC6386], endpoints MUST suppor=
t the<br>
=C2=A0 =C2=A0payload formats defined in [I-D.ietf-payload-vp8].=C2=A0 In ad=
dition they<br>
=C2=A0 =C2=A0MUST support the &#39;bilinear&#39; and &#39;none&#39; reconst=
ruction filters.<br>
<br>
The problem with the text is not that it specifies that one MUST support th=
e &quot;bilinear&quot; and &quot;none&quot; filters.<br>
<br>
The problem with the text is that in order for it to make any sense at all,=
 one must believe that you can be conformant with RFC 6386 without the abil=
ity to decode all the specified reconstruction filters - that is, in order =
for the second sentence to make any sense at all, one must believe that fil=
ters are optional.<br>
<br>
There is no hint in any place in RFC 6386 that a conformant decoder can be =
conformant without the ability to decode bitstreams using all the filter ty=
pes. Filters are not optional for the decoder.<br>
<br>
This is the point I&#39;m objecting to.<span class=3D"HOEnZb"><font color=
=3D"#888888"><br>
<br>
Harald</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div class=3D"gmail_signature">Mohammed Raad, PhD.<br>Partner<br>RAADTECH C=
ONSULTING<br>P.O. Box 113<br>Warrawong<br>NSW 2502 Australia<br>Phone: +61 =
414451478<br>Email: <a href=3D"mailto:mohammedsraad@raadtech.com" target=3D=
"_blank">mohammedsraad@raadtech.com</a></div>
</div>

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Subject: [rtcweb] Stephan Wenger's Video Document Comments
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Splitting this off from the thread it was posted on for the sake of 
cleanliness.

Thanks, Stefan, for the very useful feedback. I've incorporated it into 
the upcoming -04 version of the document as described below.

On 11/25/14 23:16, Stephan Wenger wrote:
> 1. Section 3: sRGB is fine.

Done.

> 2. Section 3: Add something like dynamic frame rate / frame interval based
> on user locale, available bandwidth, and so on.  For example, in low light
> conditions, it almost never makes sense to run a full 30 fps camera when
> your transport has only bandwidth for 15 fps.

I think I follow what you're saying, but not 100% sure. What I've added is:


* Dynamic frame rate for video capture based on actual encoding in use
   (e.g., if encoding at 15 fps due to bandwidth constraints, the camera 
will
   ideally capture at 15 fps rather than a higher rate).


> 3. Section 3: Possibly missing: mention that the video scan pattern for
> both VP8 and H.264 is YCrCb 4:2:0.

Added: "Unless specified otherwise by the SDP or codec, the video scan 
pattern for
video codecs is Y'CbCr 4:2:0".

> 4. Section 3: Possibly missing: the screen content section should mention
> that the scan format of video (YCrCb 4:2:0) is known to be suboptimal for
> the representation of screen content as produced by systems at the time of
> publication, which generally use at least 24 bits per sample RGB.  Perhaps
> a forward looking statement: ³In the future, it may become advisable to
> use video codecs optimized for screen content (scan pattern, color space,
> signal characteristics) for the representation of this type of content.²

Added:

Note that the default video scan format (Y'CrCb 4:2:0) is known to be
less than optimal for the representation of screen content produced by
most systems in use at the time of this document's publication, which
generally use RGB with at least 24 bits per sample. In the future, it
may be advisable to use video codecs optimized for screen content for the
representation of this type of content.

> 5. Section 5 second and third paragraphs: Mention Constrained Baseline
> Profile here.   Otherwise, section 5 seems fine to me.  (I know that
> constrained baseline is mentioned as a MUST later, but H.264 in isolation
> is insufficient even in this overview part of the document).

Done.

> 6. Section 6, 3rd para, 1st and 2nd line, nit: "are MUST²

Fixed.

> 5. Section 6.2, profile_level_id, suggest to change the receiving side to
> a MUST.  The decoder interprets the value anyway as presented in-band.
> What¹s the point that a receiver accepts a bitstream (without checking
> profile_level_id), and the croaks because the decoder cannot handle the
> bitstream complexity as communicated in-band?

done.

> 6. Section 6.2, max_mbps..., nit: par ameters (1st and second line)

That's a very strange tooling issue (it's fine in the source). I'm going 
to see if I can untangle what's going on here. Thanks for catching it.

>
> Normative reference to H.264: depending on the timing, it may be better to
> reference H.264v9.  That version is currently in ³pre-published² state and
> not freely downloadable, but should be generally available rather sooner
> than later (and almost certainly before we are through with this draft).

Updated the reference. Thanks for catching this.

> 7. Section 6.2, sprop_parameter_sets: For avoidance of doubt, I would
> rephrase that as ³this parameter MUST NOT be present².  It is possible to
> have that parameter and later change parameter sets in-band, and that is a
> condition I think you want to avoid.

That seems like a good point. I've taken the suggestion as:

: H.264 allows sequence and picture information to be sent both in-band, and
   out-of-band.  WebRTC implementations MUST signal this information 
in-band.
   This means that WebRTC implementations MUST NOT include this parameter in
   the SDP they generate.


> 8. Section 6.2, Possibly missing: for H.264, should we specify a pixel
> aspect ratio?  VP8 uses square pixels, right?  Perhaps follow that lead?

Added: "Unless otherwise signaled, implementations that use H.264 MUST 
encode and decode pixels with a implied 1:1 (square) aspect ratio."

> 9. Section 7: point to the payload formats also.  H.264 allows for active
> content in SEI messages... and the payload formats contain the necessary
> caveats.

Added:

Implementors making use of H.264 are also advised to take careful note 
of the
"Security Considerations" section of {{RFC6184}}, paying special regard 
to the
normative requirement pertaining to SEI messages.

/a

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<html>
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    <meta http-equiv="content-type" content="text/html; charset=utf-8">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    Splitting this off from the thread it was posted on for the sake of
    cleanliness.<br>
    <br>
    Thanks, Stefan, for the very useful feedback. I've incorporated it
    into the upcoming -04 version of the document as described below.<br>
    <br>
    <div class="moz-cite-prefix">On 11/25/14 23:16, Stephan Wenger
      wrote:<br>
    </div>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">1. Section 3: sRGB is fine.</pre>
      </div>
    </blockquote>
    <br>
    Done.<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">2. Section 3: Add something like dynamic frame rate / frame interval based
on user locale, available bandwidth, and so on.  For example, in low light
conditions, it almost never makes sense to run a full 30 fps camera when
your transport has only bandwidth for 15 fps.</pre>
      </div>
    </blockquote>
    <br>
    I think I follow what you're saying, but not 100% sure. What I've
    added is:<br>
    <br>
    <br>
    * Dynamic frame rate for video capture based on actual encoding in
    use<br>
      (e.g., if encoding at 15 fps due to bandwidth constraints, the
    camera will<br>
      ideally capture at 15 fps rather than a higher rate).<br>
    <br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">3. Section 3: Possibly missing: mention that the video scan pattern for
both VP8 and H.264 is YCrCb 4:2:0.</pre>
      </div>
    </blockquote>
    <br>
    Added: "Unless specified otherwise by the SDP or codec, the video
    scan pattern for<br>
    video codecs is Y'CbCr 4:2:0".<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">
4. Section 3: Possibly missing: the screen content section should mention
that the scan format of video (YCrCb 4:2:0) is known to be suboptimal for
the representation of screen content as produced by systems at the time of
publication, which generally use at least 24 bits per sample RGB.  Perhaps
a forward looking statement: ³In the future, it may become advisable to
use video codecs optimized for screen content (scan pattern, color space,
signal characteristics) for the representation of this type of content.²</pre>
      </div>
    </blockquote>
    <br>
    Added:<br>
    <br>
    Note that the default video scan format (Y'CrCb 4:2:0) is known to
    be<br>
    less than optimal for the representation of screen content produced
    by<br>
    most systems in use at the time of this document's publication,
    which<br>
    generally use RGB with at least 24 bits per sample. In the future,
    it<br>
    may be advisable to use video codecs optimized for screen content
    for the<br>
    representation of this type of content.<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">
5. Section 5 second and third paragraphs: Mention Constrained Baseline
Profile here.   Otherwise, section 5 seems fine to me.  (I know that
constrained baseline is mentioned as a MUST later, but H.264 in isolation
is insufficient even in this overview part of the document).
</pre>
      </div>
    </blockquote>
    <br>
    Done.<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">
6. Section 6, 3rd para, 1st and 2nd line, nit: "are MUST²
</pre>
      </div>
    </blockquote>
    <br>
    Fixed.<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">
5. Section 6.2, profile_level_id, suggest to change the receiving side to
a MUST.  The decoder interprets the value anyway as presented in-band.
What¹s the point that a receiver accepts a bitstream (without checking
profile_level_id), and the croaks because the decoder cannot handle the
bitstream complexity as communicated in-band?
</pre>
      </div>
    </blockquote>
    <br>
    done.<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">
6. Section 6.2, max_mbps..., nit: par ameters (1st and second line)</pre>
      </div>
    </blockquote>
    <br>
    That's a very strange tooling issue (it's fine in the source). I'm
    going to see if I can untangle what's going on here. Thanks for
    catching it.<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">

Normative reference to H.264: depending on the timing, it may be better to
reference H.264v9.  That version is currently in ³pre-published² state and
not freely downloadable, but should be generally available rather sooner
than later (and almost certainly before we are through with this draft).</pre>
      </div>
    </blockquote>
    <br>
    Updated the reference. Thanks for catching this.<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">
7. Section 6.2, sprop_parameter_sets: For avoidance of doubt, I would
rephrase that as ³this parameter MUST NOT be present².  It is possible to
have that parameter and later change parameter sets in-band, and that is a
condition I think you want to avoid.</pre>
      </div>
    </blockquote>
    <br>
    That seems like a good point. I've taken the suggestion as:<br>
    <br>
    : H.264 allows sequence and picture information to be sent both
    in-band, and<br>
      out-of-band.  WebRTC implementations MUST signal this information
    in-band.<br>
      This means that WebRTC implementations MUST NOT include this
    parameter in<br>
      the SDP they generate.<br>
    <br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">8. Section 6.2, Possibly missing: for H.264, should we specify a pixel
aspect ratio?  VP8 uses square pixels, right?  Perhaps follow that lead?</pre>
      </div>
    </blockquote>
    <br>
    Added: "Unless otherwise signaled, implementations that use H.264
    MUST encode and decode pixels with a implied 1:1 (square) aspect
    ratio."<br>
    <br>
    <blockquote cite="mid:D09A9942.4BF0D%25stewe@stewe.org" type="cite">
      <div class="moz-text-plain" wrap="true" graphical-quote="true"
        style="font-family: -moz-fixed; font-size: 12px;"
        lang="x-western">
        <pre wrap="">
9. Section 7: point to the payload formats also.  H.264 allows for active
content in SEI messages... and the payload formats contain the necessary
caveats.</pre>
      </div>
    </blockquote>
    <br>
    Added:<br>
    <br>
    Implementors making use of H.264 are also advised to take careful
    note of the<br>
    "Security Considerations" section of {{RFC6184}}, paying special
    regard to the<br>
    normative requirement pertaining to SEI messages.<br>
    <br>
    /a<br>
  </body>
</html>

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Also breaking this one off from the previous thread...

On 11/25/14 23:27, Stephan Wenger wrote:
> Hi,
>
> I meant to copy the mailing list on two emails regarding my (and
> Bernard’s) action items.  I mis-spelled rtcweb.  So here the content
> integrated into a single email.  (Bernard and Adam: there’s nothing new
> herein.)
>
> Hi,
> So here my proposal, with respect to H.264 baseline:
>
> MUST:
> Filler payload.  You see this occasionally from mostly early generation)
> encoder chips.  It¹s basically a way to waste bits so to fulfill buffer
> constraints.  Implementation complexity is zero: throw away bits.
>
> Full frame freeze: used in video switching MCUs, to ensure a stable
> decoded displayed picture while switching channels in probably a
> media-unaware way.  Commonly used in H.32x legacy environments and also in
> some SIP MCUs.

To be clear, you are specifically leaving off "full frame freeze 
release" on purpose, right?

> MAY:
> User data T.35 registered, and user data unregistered
> Especially the latter you see occasionally for proprietary extensions.
> It¹s a good idea to advise implementers that a bitstream may contain that
> kind of stuff, even if they don¹t act on it.
>
> Display orientation SEI, as it is referenced in section 4.

Seems reasonable to me.

Based on this advice, I have added the following text to -04:

H.264 codecs MAY send and MUST support proper interpretation of SEI "filler
payload" and "full frame freeze" messages. "Full frame freeze" messages are
used in video switching MCUs, to ensure a stable decoded displayed picture
while switching among various input streams.

When video orientation (CVO) is not signaled as part of the SDP, H.264
implementations MAY send and SHOULD support proper interpretation of Display
Orientation SEI messages.

Implementations MAY send and act upon "User data registered by Rec. ITU-T
T.35" and "User data unregistered" messages. Even if they do not act on 
them,
implementations MUST be prepared to receive such messages without any ill
effects.


Stephan: could you sanity-check that this captures your recommendations 
accurately? Thanks!

/a


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If we do any FEC with PCMU, it would probably be 2198-style for simplicity
(counter to what the current text says). Is this something that you would
use?

On Wed, Feb 11, 2015 at 11:44 AM, Jozsef Vass <jvass@twilio.com> wrote:

> Later, in 4.1 Recommended Mechanisms:
>
>    When using constant-bitrate codecs, e.g.  PCMU, use of [RFC2198 <http://tools.ietf.org/html/rfc2198>]
>    redundant encoding is NOT RECOMMENDED, as this will result in a
>    potentially significant bitrate increase.  Furthermore, suddenly
>    increasing the bitrate to deal with packet losses may actually make
>    things worse.
>
>    Because of the lower packet rate of audio encodings, usually a single
>    packet per frame, use of a separate FEC stream comes with a higher
>    overhead than other mechanisms, and therefore is NOT RECOMMENDED.
>
>
> So what is recommended for PCMU?
>
> Jozsef
>
>
> Jozsef Vass
> SDK Engineering
> [image: twilio] <http://www.twilio.com/?utm_source=email_signature>
> MOBILE(415) 216-6603EMAILjvass@twilio.com
> SIGNAL: The modern communications conference
> <http://signal.twilio.com/?utm_source=employee_signature&utm_medium=email&utm_content=signal>
> May 19/20 2015, Fort Mason Festival Pavilion, SF, CA
> <http://signal.twilio.com/?utm_source=employee_signature&utm_medium=email&utm_content=signal>
> <https://www.twilio.com/> <https://www.facebook.com/TeamTwilio>
> <https://plus.google.com/u/0/+twilio/posts> <https://twitter.com/twilio>
> <https://www.linkedin.com/company/twilio-inc.>
> <http://www.glassdoor.com/Overview/Working-at-Twilio-EI_IE410790.11,17.htm>
>
> On Wed, Feb 11, 2015 at 11:39 AM, Simon Perreault <sperreault@jive.com>
> wrote:
>
>>
>> On Wed, Feb 11, 2015 at 12:01 PM, Jozsef Vass <jvass@twilio.com> wrote:
>>
>>> I just wondering what kind of FEC is recommended (or will be supported)
>>> for G.711. The draft does not recommend neither redundant encoding nor a
>>> separate FEC stream.
>>
>>
>> Isn't this what you're looking for?
>>
>> 3.1 <http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00#section-3.1>.  Separate FEC Stream
>>
>>    This approach, as described in [RFC5956], Section 4.3 <http://tools.ietf.org/html/rfc5956#section-4.3>, sends FEC
>>    packets as an independent SSRC-multiplexed stream, with its own SSRC
>>    and payload type.  While by far the most flexible, each FEC packet
>>    will have its own IP+UDP+RTP+FEC header, leading to additional
>>    overhead of the FEC stream.
>>
>>
>> Simon
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a113f8fd2d719b0050eecf13d
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<div dir=3D"ltr">If we do any FEC with PCMU, it would probably be 2198-styl=
e for simplicity (counter to what the current text says). Is this something=
 that you would use?</div><div class=3D"gmail_extra"><br><div class=3D"gmai=
l_quote">On Wed, Feb 11, 2015 at 11:44 AM, Jozsef Vass <span dir=3D"ltr">&l=
t;<a href=3D"mailto:jvass@twilio.com" target=3D"_blank">jvass@twilio.com</a=
>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">Late=
r, in 4.1 Recommended Mechanisms:<div><br></div><div><pre style=3D"font-siz=
e:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   When using cons=
tant-bitrate codecs, e.g.  PCMU, use of [<a href=3D"http://tools.ietf.org/h=
tml/rfc2198" title=3D"&quot;RTP Payload for Redundant Audio Data&quot;" tar=
get=3D"_blank">RFC2198</a>]
   redundant encoding is NOT RECOMMENDED, as this will result in a
   potentially significant bitrate increase.  Furthermore, suddenly
   increasing the bitrate to deal with packet losses may actually make
   things worse.

   Because of the lower packet rate of audio encodings, usually a single
   packet per frame, use of a separate FEC stream comes with a higher
   overhead than other mechanisms, and therefore is NOT RECOMMENDED.</pre><=
div><br></div><div>So what is recommended for PCMU?</div><span class=3D"HOE=
nZb"><font color=3D"#888888"><div><br></div><div>Jozsef</div><div><br></div=
></font></span></div></div><div class=3D"gmail_extra"><span class=3D""><br =
clear=3D"all"><div><div><div dir=3D"ltr"><div style=3D"color:rgb(45,44,42);=
font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;font-size:1=
em;line-height:20px;padding-bottom:2px">Jozsef Vass</div><div style=3D"font=
-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;font-size:0.95e=
m;padding-bottom:15px;color:rgb(178,173,162)">SDK Engineering</div><div sty=
le=3D"color:rgb(45,44,42);font-family:&#39;Helvetica Neue&#39;,Helvetica,Ar=
ial,sans-serif;font-size:14px;line-height:20px;padding-bottom:20px"><a href=
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lvetica,Arial,sans-serif;font-size:14px;line-height:20px;padding-bottom:20p=
x"><table style=3D"border-spacing:0px;border-collapse:collapse;background-c=
olor:transparent"><tbody><tr><td width=3D"60" style=3D"padding:4px 0px 5px;=
border:none;color:rgb(178,173,162);font-size:11px;line-height:12px;text-tra=
nsform:uppercase;vertical-align:middle">MOBILE</td><td style=3D"padding:0px=
;font-size:12px"><a href=3D"tel:(415)+216-6603" style=3D"color:rgb(45,44,42=
)!important;text-decoration:none!important;font-family:&#39;Helvetica Neue&=
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 216-6603</a></td></tr><tr><td width=3D"60" style=3D"padding:4px 0px 5px;bo=
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d:0px 0px" target=3D"_blank"><span style=3D"color:rgb(235,100,97);font-weig=
ht:900">S</span><span style=3D"color:rgb(255,64,109);font-weight:900">I</sp=
an><span style=3D"color:rgb(214,67,117);font-weight:900">G</span><span styl=
e=3D"color:rgb(122,99,126);font-weight:900">N</span><span style=3D"color:rg=
b(66,99,113);font-weight:900">A</span><span style=3D"color:rgb(43,154,137);=
font-weight:900">L</span>: The modern communications conference</a></div><d=
iv style=3D"font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif=
;color:rgb(178,173,162);font-size:12px;padding-bottom:20px"><a href=3D"http=
://signal.twilio.com/?utm_source=3Demployee_signature&amp;utm_medium=3Demai=
l&amp;utm_content=3Dsignal" style=3D"color:rgb(178,173,162);text-decoration=
:none!important;background:0px 0px" target=3D"_blank">May 19/20 2015, Fort =
Mason Festival Pavilion, SF, CA</a></div><div style=3D"color:rgb(45,44,42);=
font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;font-size:1=
4px;line-height:20px;padding-bottom:20px"><table style=3D"border-spacing:0p=
x;border-collapse:collapse;background-color:transparent"><tbody><tr><td sty=
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mpany-Signature/img/icon-twilio.png" width=3D"20" style=3D"border:0px;verti=
cal-align:middle"></a></td><td style=3D"padding:12px 16px;border-style:soli=
d;border-color:rgb(216,214,208);border-width:2px 0px"><a href=3D"https://ww=
w.facebook.com/TeamTwilio" style=3D"color:rgb(66,139,202);text-decoration:n=
one;background:0px 0px" target=3D"_blank"><img src=3D"https://s3.amazonaws.=
com/ahoy-assets.twilio.com/Emails/Company-Signature/img/icon-facebook.png" =
width=3D"20" style=3D"border:0px;vertical-align:middle"></a></td><td style=
=3D"padding:12px 16px;border-style:solid;border-color:rgb(216,214,208);bord=
er-width:2px 0px"><a href=3D"https://plus.google.com/u/0/+twilio/posts" sty=
le=3D"color:rgb(66,139,202);text-decoration:none;background:0px 0px" target=
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ils/Company-Signature/img/icon-google.png" width=3D"20" style=3D"border:0px=
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h=3D"20" style=3D"border:0px;vertical-align:middle"></a></td><td style=3D"p=
adding:12px 16px;border-style:solid;border-color:rgb(216,214,208);border-wi=
dth:2px 0px"><a href=3D"https://www.linkedin.com/company/twilio-inc." style=
=3D"color:rgb(66,139,202);text-decoration:none;background:0px 0px" target=
=3D"_blank"><img src=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Ema=
ils/Company-Signature/img/icon-linkedin.png" width=3D"20" style=3D"border:0=
px;vertical-align:middle"></a></td><td style=3D"padding:12px 12px 12px 16px=
;border-style:solid;border-color:rgb(216,214,208);border-width:2px 0px"><a =
href=3D"http://www.glassdoor.com/Overview/Working-at-Twilio-EI_IE410790.11,=
17.htm" style=3D"color:rgb(66,139,202);text-decoration:none;background:0px =
0px" target=3D"_blank"><img src=3D"https://s3.amazonaws.com/ahoy-assets.twi=
lio.com/Emails/Company-Signature/img/icon-glassdoor.png" width=3D"20" style=
=3D"border:0px;vertical-align:middle"></a></td></tr></tbody></table></div><=
/div></div></div>
<br></span><div><div class=3D"h5"><div class=3D"gmail_quote">On Wed, Feb 11=
, 2015 at 11:39 AM, Simon Perreault <span dir=3D"ltr">&lt;<a href=3D"mailto=
:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gma=
il_extra"><br><div class=3D"gmail_quote">On Wed, Feb 11, 2015 at 12:01 PM, =
Jozsef Vass <span dir=3D"ltr">&lt;<a href=3D"mailto:jvass@twilio.com" targe=
t=3D"_blank">jvass@twilio.com</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">I j=
ust wondering what kind of FEC is recommended (or will be supported) for G.=
711. The draft does not recommend neither redundant encoding nor a separate=
 FEC stream.</blockquote><div><br></div><div>Isn&#39;t this what you&#39;re=
 looking for?=C2=A0</div></div><br><pre style=3D"font-size:1em;margin-top:0=
px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"line-height:0pt;displ=
ay:inline;font-size:1em;font-weight:bold"><h3 style=3D"line-height:0pt;disp=
lay:inline;font-size:1em"><a name=3D"14b7a2c5f2019384_14b7a271b939f06e_sect=
ion-3.1" href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00#sectio=
n-3.1" style=3D"color:black;text-decoration:none" target=3D"_blank">3.1</a>=
.  Separate FEC Stream</h3></span>

   This approach, as described in <a href=3D"http://tools.ietf.org/html/rfc=
5956#section-4.3" target=3D"_blank">[RFC5956], Section=C2=A04.3</a>, sends =
FEC
   packets as an independent SSRC-multiplexed stream, with its own SSRC
   and payload type.  While by far the most flexible, each FEC packet
   will have its own IP+UDP+RTP+FEC header, leading to additional
   overhead of the FEC stream.</pre><span><font color=3D"#888888"><pre styl=
e=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br><=
/pre>Simon</font></span></div></div>
</blockquote></div><br></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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From: Stephan Wenger <stewe@stewe.org>
To: Adam Roach <adam@nostrum.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Stephan Wenger's Video Document Comments
Thread-Index: AQHQRwt38145cLuh5kSTeQXJxcXOuZzuZucA
Date: Fri, 13 Feb 2015 09:47:48 +0000
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Subject: Re: [rtcweb] Stephan Wenger's Video Document Comments
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--_000_D10384D750017stewesteweorg_
Content-Type: text/plain; charset="iso-8859-1"
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Hi,
We seem to be in agreement.  There is just one point where I have additiona=
l input; see below (including suggested text).
Stephan


From: Adam Roach <adam@nostrum.com<mailto:adam@nostrum.com>>
Date: Thursday, February 12, 2015 at 22:32
To: "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Cc: Stephan Wenger <stewe@stewe.org<mailto:stewe@stewe.org>>
Subject: Stephan Wenger's Video Document Comments

Splitting this off from the thread it was posted on for the sake of cleanli=
ness.

Thanks, Stefan, for the very useful feedback. I've incorporated it into the=
 upcoming -04 version of the document as described below.

[...]


2. Section 3: Add something like dynamic frame rate / frame interval based
on user locale, available bandwidth, and so on.  For example, in low light
conditions, it almost never makes sense to run a full 30 fps camera when
your transport has only bandwidth for 15 fps.

I think I follow what you're saying, but not 100% sure. What I've added is:


* Dynamic frame rate for video capture based on actual encoding in use
  (e.g., if encoding at 15 fps due to bandwidth constraints, the camera wil=
l
  ideally capture at 15 fps rather than a higher rate).


Transmission bandwidth (which comes to play only after encoding, and where =
the encoder generally has many choices in addition to dealing down the fram=
e rate) is just one reason to go down with the capture frame rate.  Applica=
tion needs/settings is another and probably a more common one.  Yet another=
 is light conditions: in low light conditions, one is better off to sample =
at low frame rate, for two reasons: first, the human eye is not particularl=
y good in observing fast movement in low light, so why bother capturing/cod=
ing it?  And second, a camera sensor creates more noise at low light condit=
ions, which can be mitigated by allowing more photons to hit the sensor ele=
ment in question over time.
Also, capture frame rate can be dictated not only by the codec (as a result=
 of bandwidth starvation), but also by the environment (light/applications =
setting).  Your text suggests that the frame rate adjustment trickles down =
from transmission needs, but that's not the only source (and perhaps not ev=
en the most relevant one).

Suggested text:


  *   Dynamic frame rate for video capture based on actual encoding in use =
(e.g., if encoding at 15 fps due to bandwidth constraints, low light condit=
ions, or application settings, the camera will ideally capture at 15 fps ra=
ther than a higher rate).


--_000_D10384D750017stewesteweorg_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;"><br>
</blockquote>
<div>Hi,</div>
<div>We seem to be in agreement. &nbsp;There is just one point where I have=
 additional input; see below (including suggested text).</div>
<div>Stephan</div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<div><br>
</div>
<div><br>
</div>
</blockquote>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Adam Roach &lt;<a href=3D"mai=
lto:adam@nostrum.com">adam@nostrum.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, February 12, 2015 a=
t 22:32<br>
<span style=3D"font-weight:bold">To: </span>&quot;<a href=3D"mailto:rtcweb@=
ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">=
rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Stephan Wenger &lt;<a href=3D"m=
ailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Stephan Wenger's Video Doc=
ument Comments<br>
</div>
<div><br>
</div>
</blockquote>
<div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<div text=3D"#000000" bgcolor=3D"#FFFFFF">Splitting this off from the threa=
d it was posted on for the sake of cleanliness.<br>
<br>
Thanks, Stefan, for the very useful feedback. I've incorporated it into the=
 upcoming -04 version of the document as described below.<br>
<br>
<div class=3D"moz-cite-prefix">[&#8230;]</div>
<br>
<blockquote cite=3D"mid:D09A9942.4BF0D%25stewe@stewe.org" type=3D"cite">
<div class=3D"moz-text-plain" wrap=3D"true" graphical-quote=3D"true" style=
=3D"font-family: -moz-fixed; font-size: 12px;" lang=3D"x-western">
<pre wrap=3D"">2. Section 3: Add something like dynamic frame rate / frame =
interval based
on user locale, available bandwidth, and so on.  For example, in low light
conditions, it almost never makes sense to run a full 30 fps camera when
your transport has only bandwidth for 15 fps.</pre>
</div>
</blockquote>
<br>
I think I follow what you're saying, but not 100% sure. What I've added is:=
<br>
<br>
<br>
* Dynamic frame rate for video capture based on actual encoding in use<br>
&nbsp; (e.g., if encoding at 15 fps due to bandwidth constraints, the camer=
a will<br>
&nbsp; ideally capture at 15 fps rather than a higher rate).<br>
<br>
</div>
</blockquote>
</div>
</span>
<div><br>
</div>
<div>
<div>Transmission bandwidth (which comes to play only after encoding, and w=
here the encoder generally has many choices in addition to dealing down the=
 frame rate) is just one reason to go down with the capture frame rate. &nb=
sp;Application needs/settings is another
 and probably a more common one. &nbsp;Yet another is light conditions: in =
low light conditions, one is better off to sample at low frame rate, for tw=
o reasons: first, the human eye is not particularly good in observing fast =
movement in low light, so why bother
 capturing/coding it? &nbsp;And second, a camera sensor creates more noise =
at low light conditions, which can be mitigated by allowing more photons to=
 hit the sensor element in question over time. &nbsp;</div>
<div>Also, capture frame rate can be dictated not only by the codec (as a r=
esult of bandwidth starvation), but also by the environment (light/applicat=
ions setting). &nbsp;Your text suggests that the frame rate adjustment tric=
kles down from transmission needs, but
 that&#8217;s not the only source (and perhaps not even the most relevant o=
ne).&nbsp;</div>
<div><br>
</div>
<div>Suggested text:</div>
<div><br>
</div>
<ul>
<li>Dynamic frame rate for video capture based on actual encoding in use&nb=
sp;(e.g., if encoding at 15 fps due to bandwidth constraints, low light con=
ditions, or application settings, the camera will ideally capture at 15 fps=
 rather than a higher rate).&nbsp;</li></ul>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<div text=3D"#000000" bgcolor=3D"#FFFFFF"><br>
</div>
</blockquote>
</div>
</span>
</body>
</html>

--_000_D10384D750017stewesteweorg_--


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From: Stephan Wenger <stewe@stewe.org>
To: Adam Roach <adam@nostrum.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Video Open Issue #6: H.264 SEI messages
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Date: Fri, 13 Feb 2015 09:54:41 +0000
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: SDP-SCTP: Max value for SDP max-message-size attribute?
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Date: Fri, 13 Feb 2015 13:28:29 +0000
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Subject: [rtcweb] SDP-SCTP: Max value for SDP max-message-size attribute?
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Hi,

The MMUSIC SCTP-SDP draft defines the SDP max-message-size attribute, which=
 can be used to indicate the maximum size of message that an entity is will=
ing to receive on an SCTP association

I have received a comment on whether we should define a max value for the a=
ttribute.

When reading the rtcweb-data-channel draft, it says that a sender should no=
t send messages bigger than 16 KB, unless interleaving is used.

As an entity can choose how big messages to send (as long as the message si=
ze is smaller than the max-message-size value), but suggestion would be to =
NOT define an attribute max value in the SCTP-SDP draft. Different usages (=
e.g. webrtc data channels) need to choose the size based on SCTP features s=
upported etc.

Regards,

Christer

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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">The MMUSIC SCTP-SDP draft defin=
es the SDP max-message-size attribute, which can be used to indicate the ma=
ximum size of message that an entity is willing to receive on an SCTP assoc=
iation<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have received a comment on wh=
ether we should define a max value for the attribute.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">When reading the rtcweb-data-ch=
annel draft, it says that a sender should not send messages bigger than 16 =
KB, unless interleaving is used.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">As an entity can choose how big=
 messages to send (as long as the message size is smaller than the max-mess=
age-size value), but suggestion would be to NOT define an attribute max val=
ue in the SCTP-SDP draft. Different
 usages (e.g. webrtc data channels) need to choose the size based on SCTP f=
eatures supported etc.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<o:p></o:p></span></p>
</div>
</body>
</html>

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From nobody Fri Feb 13 05:55:58 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Subject: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Hi,

In the TLS role determination section of the SDP-SCTP draft, the text says:

                             "Once a DTLS connection has been established, =
if the 'active/passive' status of the
                             endpoints change (as result of an offer/answer=
 exchange) during a session, a
                             new DTLS connection MUST be established. There=
fore, endpoints SHOULD NOT change
                             the 'active/passive' status during a session, =
unless they want to establish a
                             new DTLS connection."

I have received a question whether this also affects the underlying TCP con=
nection (in case of 'TCP/DTLS/SCTP') and the SCTP association (in case of '=
SCTP/DTLS').

My suggestion would be to NOT re-establish the underlying TCP/SCTP associat=
ion just because the DTLS connection is re-established. Also, I could not f=
ind anything in RFC 6083 that would indicate that the SCTP association woul=
d have to be re-established.

(IF a user also wants the underlying TCP/SCTP association to be re-establis=
hed, he/she uses the SDP connection attribute with a 'new' value.)

Regards,

Christer

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<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">In the TLS role determination s=
ection of the SDP-SCTP draft, the text says:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;Once =
a DTLS connection has been established, if the 'active/passive' status of t=
he<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; endpoints ch=
ange (as result of an offer/answer exchange) during a session, a
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; new DTLS con=
nection MUST be established. Therefore, endpoints SHOULD NOT change
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the 'active/=
passive' status during a session, unless they want to establish a
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; new DTLS con=
nection.&#8221;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have received a question whet=
her this also affects the underlying TCP connection (in case of &#8216;TCP/=
DTLS/SCTP&#8217;) and the SCTP association (in case of &#8216;SCTP/DTLS&#82=
17;).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">My suggestion would be to NOT r=
e-establish the underlying TCP/SCTP association just because the DTLS conne=
ction is re-established. Also, I could not find anything in RFC 6083 that w=
ould indicate that the SCTP association
 would have to be re-established.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">(IF a user also wants the under=
lying TCP/SCTP association to be re-established, he/she uses the SDP connec=
tion attribute with a &#8216;new&#8217; value.)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<o:p></o:p></span></p>
</div>
</body>
</html>

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Date: Fri, 13 Feb 2015 16:29:33 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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On 02/13/2015 02:55 PM, Christer Holmberg wrote:
>
> Hi,
>
> In the TLS role determination section of the SDP-SCTP draft, the text 
> says:
>
> Once a DTLS connection has been established, if the 'active/passive' 
> status of the
>
> endpoints change (as result of an offer/answer exchange) during a 
> session, a
>
> new DTLS connection MUST be established. Therefore, endpoints SHOULD 
> NOT change
>
> the 'active/passive' status during a session, unless they want to 
> establish a
>
> new DTLS connection.
>
> I have received a question whether this also affects the underlying 
> TCP connection (in case of TCP/DTLS/SCTP) and the SCTP association 
> (in case of SCTP/DTLS).
>
> My suggestion would be to NOT re-establish the underlying TCP/SCTP 
> association just because the DTLS connection is re-established. Also, 
> I could not find anything in RFC 6083 that would indicate that the 
> SCTP association would have to be re-established.
>

Seems reasonable to me - but since this is an MMUSIC draft 
(draft-ietf-mmusic-sctp-sdp) and has the potential to be used outside of 
RTCWEB, shoudn't this go on the mmusic mailing list?


> (IF a user also wants the underlying TCP/SCTP association to be 
> re-established, he/she uses the SDP connection attribute with a new 
> value.)
>
> Regards,
>
> Christer
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">On 02/13/2015 02:55 PM, Christer
      Holmberg wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1D6D1E74@ESESSMB209.ericsson.se"
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      <div class="WordSection1">
        <p class="MsoNormal"><span lang="EN-US">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">In the TLS role
            determination section of the SDP-SCTP draft, the text says:<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">
            Once a DTLS connection has been established, if the
            'active/passive' status of the<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">
            endpoints change (as result of an offer/answer exchange)
            during a session, a
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">
            new DTLS connection MUST be established. Therefore,
            endpoints SHOULD NOT change
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">
            the 'active/passive' status during a session, unless they
            want to establish a
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">
            new DTLS connection.<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">I have received a
            question whether this also affects the underlying TCP
            connection (in case of TCP/DTLS/SCTP) and the SCTP
            association (in case of SCTP/DTLS).<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">My suggestion would be
            to NOT re-establish the underlying TCP/SCTP association just
            because the DTLS connection is re-established. Also, I could
            not find anything in RFC 6083 that would indicate that the
            SCTP association would have to be re-established.</span></p>
      </div>
    </blockquote>
    <br>
    Seems reasonable to me - but since this is an MMUSIC draft
    (draft-ietf-mmusic-sctp-sdp) and has the potential to be used
    outside of RTCWEB, shoudn't this go on the mmusic mailing list?<br>
    <br>
    <br>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1D6D1E74@ESESSMB209.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">(IF a user also wants
            the underlying TCP/SCTP association to be re-established,
            he/she uses the SDP connection attribute with a new
            value.)<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="EN-US">Christer<o:p></o:p></span></p>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
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--------------070906000807080205050003--


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Date: Fri, 13 Feb 2015 11:23:07 -0500
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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On Fri, Feb 13, 2015 at 8:55 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  In the TLS role determination section of the SDP-SCTP draft, the text
> says:
>
>
>
>                              =E2=80=9COnce a DTLS connection has been est=
ablished,
> if the 'active/passive' status of the
>
>                              endpoints change (as result of an
> offer/answer exchange) during a session, a
>
>                              new DTLS connection MUST be established.
> Therefore, endpoints SHOULD NOT change
>
>                              the 'active/passive' status during a session=
,
> unless they want to establish a
>
>                              new DTLS connection.=E2=80=9D
>
>
>
> I have received a question whether this also affects the underlying TCP
> connection (in case of =E2=80=98TCP/DTLS/SCTP=E2=80=99) and the SCTP asso=
ciation (in case
> of =E2=80=98SCTP/DTLS=E2=80=99).
>
>
>
> My suggestion would be to NOT re-establish the underlying TCP/SCTP
> association just because the DTLS connection is re-established. Also, I
> could not find anything in RFC 6083 that would indicate that the SCTP
> association would have to be re-established.
>
>
>
> (IF a user also wants the underlying TCP/SCTP association to be
> re-established, he/she uses the SDP connection attribute with a =E2=80=98=
new=E2=80=99
> value.)
>
>
>
>
>
As far as I know the only way to demultiplex DTLS packets is based on the
underlying transport. Unless you can guarantee that no packets from the
previous DTLS connection will arrive on the new connection reusing the same
underlying transport some interesting problems can occur. Unless someone
proposes a way to demux two DTLS connections on the same transport, I would
suggest that new underlying transport should always be used for new DTLS
connection and that DTLS connection settings should not be changed unless
the transport is changed as well.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Feb 13, 2015 at 8:55 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holm=
berg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color=
:rgb(204,204,204);border-left-style:solid;padding-left:1ex">





<div lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">In the TLS role determination section of the SDP-SCT=
P draft, the text says:<br></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =E2=80=9COn=
ce a DTLS connection has been established, if the &#39;active/passive&#39; =
status of the<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 endpoints c=
hange (as result of an offer/answer exchange) during a session, a
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 new DTLS co=
nnection MUST be established. Therefore, endpoints SHOULD NOT change
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 the &#39;ac=
tive/passive&#39; status during a session, unless they want to establish a
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 new DTLS co=
nnection.=E2=80=9D<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have received a question whet=
her this also affects the underlying TCP connection (in case of =E2=80=98TC=
P/DTLS/SCTP=E2=80=99) and the SCTP association (in case of =E2=80=98SCTP/DT=
LS=E2=80=99).<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">My suggestion would be to NOT r=
e-establish the underlying TCP/SCTP association just because the DTLS conne=
ction is re-established. Also, I could not find anything in RFC 6083 that w=
ould indicate that the SCTP association
 would have to be re-established.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">(IF a user also wants the under=
lying TCP/SCTP association to be re-established, he/she uses the SDP connec=
tion attribute with a =E2=80=98new=E2=80=99 value.)<u></u><u></u></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>As far as I know the only way to demultiplex DTLS packets is based on the =
underlying transport. Unless you can guarantee that no packets from the pre=
vious DTLS connection will arrive on the new connection reusing the same un=
derlying transport some interesting problems can occur. Unless someone prop=
oses a way to demux two DTLS connections on the same transport, I would sug=
gest that new underlying transport should always be used for new DTLS conne=
ction and that DTLS connection settings should not be changed unless the tr=
ansport is changed as well.=C2=A0</div><div><div class=3D"gmail_signature">=
_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></di=
v>

--089e015383fcdf7ff6050efaa6af--


From nobody Fri Feb 13 09:10:21 2015
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Subject: Re: [rtcweb] Video Open issue #4: VP8 filters
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Well, I guess fifth time asking is the charm.

Consensus seems to be to remove this sentence -- so it's gone.

/a

On 2/12/15 15:00, Harald Alvestrand wrote:
> On 02/12/2015 09:48 PM, Adam Roach wrote:
>> In Toronto, there was a claim that specification that "bilinear" and 
>> "none" filters is superfluous, as they are already required as part 
>> of the VP8 spec. Since that time, I have repeatedly asked for a 
>> concrete pointer to the text that would make this unnecessary.
>>
>> No one has come forth with such a pointer. Given the harmless nature 
>> of redundantly specifying this requirement on implementations and the 
>> lack of concrete evidence for its removal, I am leaving the text 
>> as-is and removing the open issue from my list.
>
> Please. This is not harmless.
>
> The text under consideration is
>
> 6.1.  VP8
>
>    For the VP8 codec, defined in [RFC6386], endpoints MUST support the
>    payload formats defined in [I-D.ietf-payload-vp8].  In addition they
>    MUST support the 'bilinear' and 'none' reconstruction filters.
>
> The problem with the text is not that it specifies that one MUST 
> support the "bilinear" and "none" filters.
>
> The problem with the text is that in order for it to make any sense at 
> all, one must believe that you can be conformant with RFC 6386 without 
> the ability to decode all the specified reconstruction filters - that 
> is, in order for the second sentence to make any sense at all, one 
> must believe that filters are optional.
>
> There is no hint in any place in RFC 6386 that a conformant decoder 
> can be conformant without the ability to decode bitstreams using all 
> the filter types. Filters are not optional for the decoder.
>
> This is the point I'm objecting to.
>
> Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Fri Feb 13 09:56:33 2015
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On 2/13/15 03:54, Stephan Wenger wrote:
> Sounds OK. Quick question for my education and to make sure: SDP CVO 
> refers to the presence of that RTP header extension that indicates 
> picture orientation, and not to a fixed orientation signaled in the 
> SDP, right?

Yes. I've added some more text to hopefully make this clearer:

When the use of the video orientation (CVO) RTP header extension is not
signaled as part of the SDP, H.264 implementations MAY send and SHOULD 
support
proper interpretation of Display Orientation SEI messages.

/a


From nobody Fri Feb 13 10:13:09 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Video Processing and Codec Requirements
        Author          : Adam Roach
	Filename        : draft-ietf-rtcweb-video-04.txt
	Pages           : 9
	Date            : 2015-02-13

Abstract:
   This specification provides the requirements and considerations for
   WebRTC applications to send and receive video across a network.  It
   specifies the video processing that is required, as well as video
   codecs and their parameters.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-video/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-video-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-video-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Fri Feb 13 10:21:24 2015
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I just uploaded a -04 version of the video draft. This closes all of the 
open issues of which I am aware (see yesterday's messages for the 
resolution to the few items that had remained open in -03).

New version here: https://tools.ietf.org/html/draft-ietf-rtcweb-video-04
Diff from previous here: 
https://tools.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-video-04.txt

At this point, I consider the document functionally complete and ready 
for WGLC.

/a


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To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Date: Fri, 13 Feb 2015 23:35:22 +0000
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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From nobody Fri Feb 13 15:38:17 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Date: Fri, 13 Feb 2015 23:38:06 +0000
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Hi Harald,
=A0
>>In the TLS role determination section of the SDP-SCTP draft, the text say=
s:
>>=A0
>>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 "Once a DTLS connection has been established, if the 'active/p=
assive' status of the
>>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 endpoints change (as result of an offer/answer exchange) durin=
g a session, a=20
>>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 new DTLS connection MUST be established. Therefore, endpoints =
SHOULD NOT change=20
>>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 the 'active/passive' status during a session, unless they want=
 to establish a=20
>>=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 new DTLS connection."
>>=A0
>> I have received a question whether this also affects the underlying TCP =
connection (in case of 'TCP/DTLS/SCTP') and the SCTP=20
>> association (in case of 'SCTP/DTLS').
>>
>>My suggestion would be to NOT re-establish the underlying TCP/SCTP associ=
ation just because the DTLS connection is re->>established. Also, I could n=
ot find anything in RFC 6083 that would indicate that the SCTP association =
would have to be re->>established.
>
>Seems reasonable to me - but since this is an MMUSIC draft (draft-ietf-mmu=
sic-sctp-sdp) and has the potential to be used >outside of RTCWEB, shoudn't=
 this go on the mmusic mailing list?

They will have their chance to comment during WGLC :)

Seriously, the reason I brought it here first is because the people actuall=
y implementing this (for WebRTC Data Channels) are more likely to be found =
here.

Regards,

Christer


From nobody Fri Feb 13 16:14:35 2015
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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On Fri, Feb 13, 2015 at 6:35 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Don=E2=80=99t you have the same problem in case of DTLS re-keying?
>

You do have the same problem in case of DTLS re-key. This is why none of
the current browser implementations actually support it. At this point I
would be very curious to see if anybody implemented DTLS re-key or
re-negotiation over the same transport and would very much appreciate if
they tell us how exactly they did this.

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Feb 13, 2015 at 6:35 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holm=
berg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color=
:rgb(204,204,204);border-left-style:solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125);font-family:Cali=
bri,sans-serif;font-size:11pt">Don=E2=80=99t you have the same problem in c=
ase of DTLS re-keying?</span></p></div></div></blockquote><div><br></div><d=
iv>You do have the same problem in case of DTLS re-key. This is why none of=
 the current browser implementations actually support it. At this point I w=
ould be very curious to see if anybody implemented DTLS re-key or re-negoti=
ation over the same transport and would very much appreciate if they tell u=
s how exactly they did this.</div><div><br></div><div>Regards,=C2=A0</div><=
div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></di=
v><div>=C2=A0</div></div></div></div>

--001a113ed13094f157050f013c14--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Fri, 13 Feb 2015 16:29:38 -0800
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Harald said:

"My suggestion would be to NOT re-establish the underlying TCP/SCTP
association just because the DTLS connection is re->>established. Also, I
could not find anything in RFC 6083 that would indicate that the SCTP
association would have to be re->>established."

[BA] There is no reason to re-establish the underlying TCP/SCTP association
due to DTLS connection re-establishment or ICE re-start.  Similarly, there
is no reason a DTLS connection should need to be re-established just
because of an ICE re-start.

On Fri, Feb 13, 2015 at 3:38 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi Harald,
>
> >>In the TLS role determination section of the SDP-SCTP draft, the text
> says:
> >>
> >>                             "Once a DTLS connection has been
> established, if the 'active/passive' status of the
> >>                             endpoints change (as result of an
> offer/answer exchange) during a session, a
> >>                             new DTLS connection MUST be established.
> Therefore, endpoints SHOULD NOT change
> >>                             the 'active/passive' status during a
> session, unless they want to establish a
> >>                             new DTLS connection."
> >>
> >> I have received a question whether this also affects the underlying TCP
> connection (in case of 'TCP/DTLS/SCTP') and the SCTP
> >> association (in case of 'SCTP/DTLS').
> >>
> >>My suggestion would be to NOT re-establish the underlying TCP/SCTP
> association just because the DTLS connection is re->>established. Also, I
> could not find anything in RFC 6083 that would indicate that the SCTP
> association would have to be re->>established.
> >
> >Seems reasonable to me - but since this is an MMUSIC draft
> (draft-ietf-mmusic-sctp-sdp) and has the potential to be used >outside of
> RTCWEB, shoudn't this go on the mmusic mailing list?
>
> They will have their chance to comment during WGLC :)
>
> Seriously, the reason I brought it here first is because the people
> actually implementing this (for WebRTC Data Channels) are more likely to be
> found here.
>
> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Harald said:=C2=A0<div><br></div><div>&quot;<span style=3D=
"font-size:12.8000001907349px">My suggestion would be to NOT re-establish t=
he underlying TCP/SCTP association just because the DTLS connection is re-&=
gt;&gt;established. Also, I could not find anything in RFC 6083 that would =
indicate that the SCTP association would have to be re-&gt;&gt;established.=
</span>&quot;</div><div><br></div><div>[BA] There is no reason to re-establ=
ish the underlying TCP/SCTP association due to DTLS connection re-establish=
ment or ICE re-start.=C2=A0 Similarly, there is no reason a DTLS connection=
 should need to be re-established just because of an ICE re-start. =C2=A0</=
div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri,=
 Feb 13, 2015 at 3:38 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi=
 Harald,<br>
<span class=3D"">=C2=A0<br>
&gt;&gt;In the TLS role determination section of the SDP-SCTP draft, the te=
xt says:<br>
&gt;&gt;=C2=A0<br>
&gt;&gt;=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0 &quot;Once a DTLS connection has been establish=
ed, if the &#39;active/passive&#39; status of the<br>
&gt;&gt;=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0 endpoints change (as result of an offer/answer =
exchange) during a session, a<br>
&gt;&gt;=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0 new DTLS connection MUST be established. Theref=
ore, endpoints SHOULD NOT change<br>
&gt;&gt;=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0 the &#39;active/passive&#39; status during a se=
ssion, unless they want to establish a<br>
&gt;&gt;=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0 new DTLS connection.&quot;<br>
&gt;&gt;=C2=A0<br>
&gt;&gt; I have received a question whether this also affects the underlyin=
g TCP connection (in case of &#39;TCP/DTLS/SCTP&#39;) and the SCTP<br>
&gt;&gt; association (in case of &#39;SCTP/DTLS&#39;).<br>
&gt;&gt;<br>
</span>&gt;&gt;My suggestion would be to NOT re-establish the underlying TC=
P/SCTP association just because the DTLS connection is re-&gt;&gt;establish=
ed. Also, I could not find anything in RFC 6083 that would indicate that th=
e SCTP association would have to be re-&gt;&gt;established.<br>
&gt;<br>
&gt;Seems reasonable to me - but since this is an MMUSIC draft (draft-ietf-=
mmusic-sctp-sdp) and has the potential to be used &gt;outside of RTCWEB, sh=
oudn&#39;t this go on the mmusic mailing list?<br>
<br>
They will have their chance to comment during WGLC :)<br>
<br>
Seriously, the reason I brought it here first is because the people actuall=
y implementing this (for WebRTC Data Channels) are more likely to be found =
here.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--bcaec53f3717fffc93050f0173fc--


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--089e013d1d528b1ed0050f13befd
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>From Section 4.1:

   Initial consent to send traffic is obtained using ICE.  Consent
   expires after 30 seconds.  That is, if a valid STUN binding response
   corresponding to any STUN request sent in the last 30 seconds has not
   been received from the remote peer's transport address, the endpoint
   MUST cease transmission on that 5-tuple.  STUN consent responses
   received after consent expiry do not re-establish consent, and may be
   discarded or cause an ICMP error.


[BA] Let us say that peer A sends STUN requests to peer B.  Request 1

does not receive a response within 30 seconds, but Request 2 (sent 5
seconds later) does receive a response.  Does that mean that peer A
considers consent revoked 30 seconds after sending Request 1, even
though it has since gotten a response?


This does not make sense to me.

--089e013d1d528b1ed0050f13befd
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<div dir=3D"ltr">From Section 4.1:=C2=A0<div><br></div><div><pre class=3D""=
 style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">=
   Initial consent to send traffic is obtained using ICE.  Consent
   expires after 30 seconds.  That is, if a valid STUN binding response
   corresponding to any STUN request sent in the last 30 seconds has not
   been received from the remote peer&#39;s transport address, the endpoint
   MUST cease transmission on that 5-tuple.  STUN consent responses
   received after consent expiry do not re-establish consent, and may be
   discarded or cause an ICMP error.</pre><pre class=3D"" style=3D"font-siz=
e:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre cla=
ss=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0=
,0,0)">[BA] Let us say that peer A sends STUN requests to peer B.  Request =
1</pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:=
0px;color:rgb(0,0,0)">does not receive a response within 30 seconds, but Re=
quest 2 (sent 5 seconds later) does receive a response.  Does that mean tha=
t peer A considers consent revoked 30 seconds after sending Request 1, even=
 though it has since gotten a response? =C2=A0</pre><pre class=3D"" style=
=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></=
pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px=
;color:rgb(0,0,0)">This does not make sense to me. </pre></div></div>

--089e013d1d528b1ed0050f13befd--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Question about draft-ietf-rtcweb-stun-consent-freshness-11
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Subject: Re: [rtcweb] Question about draft-ietf-rtcweb-stun-consent-freshness-11
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--_000_7594FB04B1934943A5C02806D1A2204B1D6ED86EESESSMB209erics_
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Hi Bernard,

I had the same comment some time ago.

But, note that the text now says "response to *any* STUN request", meaning =
that as long as you receive a response to at least one STUN request that wa=
s sent in the last 30 seconds you are ok, and consent will not expire.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Bernard Aboba<mailto:bernard.aboba@gmail.com>
Sent: =FD15/=FD02/=FD2015 00:19
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Question about draft-ietf-rtcweb-stun-consent-freshness-1=
1

>From Section 4.1:


   Initial consent to send traffic is obtained using ICE.  Consent
   expires after 30 seconds.  That is, if a valid STUN binding response
   corresponding to any STUN request sent in the last 30 seconds has not
   been received from the remote peer's transport address, the endpoint
   MUST cease transmission on that 5-tuple.  STUN consent responses
   received after consent expiry do not re-establish consent, and may be
   discarded or cause an ICMP error.


[BA] Let us say that peer A sends STUN requests to peer B.  Request 1

does not receive a response within 30 seconds, but Request 2 (sent 5 second=
s later) does receive a response.  Does that mean that peer A considers con=
sent revoked 30 seconds after sending Request 1, even though it has since g=
otten a response?


This does not make sense to me.

--_000_7594FB04B1934943A5C02806D1A2204B1D6ED86EESESSMB209erics_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi Bernard,<b=
r>
<br>
I had the same comment some time ago.<br>
<br>
But, note that the text now says &quot;response to *any* STUN request&quot;=
, meaning that as long as you receive a response to at least one STUN reque=
st that was sent in the last 30 seconds you are ok, and consent will not ex=
pire.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:bernard.aboba@gmail.com">Bernard Aboba</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD15=
/=FD02/=FD2015 00:19</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">[rtcw=
eb] Question about draft-ietf-rtcweb-stun-consent-freshness-11</span><br>
<br>
</div>
<div>
<div dir=3D"ltr">From Section 4.1:&nbsp;
<div><br>
</div>
<div>
<pre class=3D"" style=3D"font-size:1em; margin-top:0px; margin-bottom:0px; =
color:rgb(0,0,0)">   Initial consent to send traffic is obtained using ICE.=
  Consent
   expires after 30 seconds.  That is, if a valid STUN binding response
   corresponding to any STUN request sent in the last 30 seconds has not
   been received from the remote peer's transport address, the endpoint
   MUST cease transmission on that 5-tuple.  STUN consent responses
   received after consent expiry do not re-establish consent, and may be
   discarded or cause an ICMP error.</pre>
<pre class=3D"" style=3D"font-size:1em; margin-top:0px; margin-bottom:0px; =
color:rgb(0,0,0)"><br></pre>
<pre class=3D"" style=3D"font-size:1em; margin-top:0px; margin-bottom:0px; =
color:rgb(0,0,0)">[BA] Let us say that peer A sends STUN requests to peer B=
.  Request 1</pre>
<pre class=3D"" style=3D"font-size:1em; margin-top:0px; margin-bottom:0px; =
color:rgb(0,0,0)">does not receive a response within 30 seconds, but Reques=
t 2 (sent 5 seconds later) does receive a response.  Does that mean that pe=
er A considers consent revoked 30 seconds after sending Request 1, even tho=
ugh it has since gotten a response? &nbsp;</pre>
<pre class=3D"" style=3D"font-size:1em; margin-top:0px; margin-bottom:0px; =
color:rgb(0,0,0)"><br></pre>
<pre class=3D"" style=3D"font-size:1em; margin-top:0px; margin-bottom:0px; =
color:rgb(0,0,0)">This does not make sense to me. </pre>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D6ED86EESESSMB209erics_--


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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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On Fri, Feb 13, 2015 at 7:14 PM, Roman Shpount <roman@telurix.com> wrote:

> On Fri, Feb 13, 2015 at 6:35 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>>  Don=E2=80=99t you have the same problem in case of DTLS re-keying?
>>
>
> You do have the same problem in case of DTLS re-key. This is why none of
> the current browser implementations actually support it. At this point I
> would be very curious to see if anybody implemented DTLS re-key or
> re-negotiation over the same transport and would very much appreciate if
> they tell us how exactly they did this.
>

Uh?

I did implement DTLS re-key / re-negotiation as specified. Then I tested
it. It works. I don't know what problem you're alluding to.

Simon

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Fri, Feb 13, 2015 at 7:14 PM, Roman Shpount <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><span class=3D"">On Fri, Fe=
b 13, 2015 at 6:35 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@e=
ricsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(2=
04,204,204);border-left-style:solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125);font-family:Cali=
bri,sans-serif;font-size:11pt">Don=E2=80=99t you have the same problem in c=
ase of DTLS re-keying?</span></p></div></div></blockquote><div><br></div></=
span><div>You do have the same problem in case of DTLS re-key. This is why =
none of the current browser implementations actually support it. At this po=
int I would be very curious to see if anybody implemented DTLS re-key or re=
-negotiation over the same transport and would very much appreciate if they=
 tell us how exactly they did this.</div></blockquote></div><br>Uh?</div><d=
iv class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">I did impleme=
nt DTLS re-key / re-negotiation as specified. Then I tested it. It works. I=
 don&#39;t know what problem you&#39;re alluding to.</div><div class=3D"gma=
il_extra"><br></div><div class=3D"gmail_extra">Simon</div></div>

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To: Bernard Aboba <bernard.aboba@gmail.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Date: Mon, 16 Feb 2015 12:47:40 -0500
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From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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On Mon, Feb 16, 2015 at 8:55 AM, Simon Perreault <sperreault@jive.com>
wrote:

>
> On Fri, Feb 13, 2015 at 7:14 PM, Roman Shpount <roman@telurix.com> wrote:
>
>> On Fri, Feb 13, 2015 at 6:35 PM, Christer Holmberg <
>> christer.holmberg@ericsson.com> wrote:
>>
>>>  Don=E2=80=99t you have the same problem in case of DTLS re-keying?
>>>
>>
>> You do have the same problem in case of DTLS re-key. This is why none of
>> the current browser implementations actually support it. At this point I
>> would be very curious to see if anybody implemented DTLS re-key or
>> re-negotiation over the same transport and would very much appreciate if
>> they tell us how exactly they did this.
>>
>
> Uh?
>
> I did implement DTLS re-key / re-negotiation as specified. Then I tested
> it. It works. I don't know what problem you're alluding to.
>
>
>
How do you deal with late DTLS data packets or alerts for the previous
connection received on the new connection? If it is an open source, you can
simply point me to the relevant code.

I do know how to implement DTLS-SRTP re-key. It is dealing with new DTLS
session running over existing connection where packets from the previous
DTLS session can be present which is not completely clear to me.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Feb 16, 2015 at 8:55 AM, Simon Perreault <span dir=3D"ltr">&lt=
;<a href=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.c=
om</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wid=
th:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-l=
eft:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"h5"=
><br><div class=3D"gmail_quote">On Fri, Feb 13, 2015 at 7:14 PM, Roman Shpo=
unt <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_b=
lank">roman@telurix.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail=
_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left=
-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span>On =
Fri, Feb 13, 2015 at 6:35 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.hol=
mberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-colo=
r:rgb(204,204,204);border-left-style:solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125);font-family:Cali=
bri,sans-serif;font-size:11pt">Don=E2=80=99t you have the same problem in c=
ase of DTLS re-keying?</span></p></div></div></blockquote><div><br></div></=
span><div>You do have the same problem in case of DTLS re-key. This is why =
none of the current browser implementations actually support it. At this po=
int I would be very curious to see if anybody implemented DTLS re-key or re=
-negotiation over the same transport and would very much appreciate if they=
 tell us how exactly they did this.</div></blockquote></div><br></div></div=
>Uh?</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">I=
 did implement DTLS re-key / re-negotiation as specified. Then I tested it.=
 It works. I don&#39;t know what problem you&#39;re alluding to.</div><span=
 class=3D""><font color=3D"#888888"><div class=3D"gmail_extra"><br></div><d=
iv class=3D"gmail_extra"><br></div></font></span></div></blockquote><div><b=
r></div><div>How do you deal with late DTLS data packets or alerts for the =
previous connection received on the new connection? If it is an open source=
, you can simply point me to the relevant code.</div><div><br></div><div>I =
do know how to implement DTLS-SRTP re-key. It is dealing with new DTLS sess=
ion running over existing connection where packets from the previous DTLS s=
ession can be present which is not completely clear to me.</div><div><div c=
lass=3D"gmail_signature">_____________<br>Roman Shpount</div></div><br><div=
>=C2=A0</div></div></div></div>

--001a113ed130c72bf8050f382e18--


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Date: Mon, 16 Feb 2015 12:51:19 -0500
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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On Mon, Feb 16, 2015 at 9:44 AM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>
>
> >As far as I know the only way to demultiplex DTLS packets is based on
> the underlying transport. >Unless you can >guarantee that no packets from
> the previous DTLS connection will arrive on the >new connection reusing t=
he
> same >underlying transport some interesting problems can occur.
>
> [Raju] I believe packets from old DTLS connection appearing on new will b=
e
> dropped the same way DTLS rekeying handles packets using old key after MS=
L
> timeout [1]. So, I don=E2=80=99t think a new TCP/SCTP transport connectio=
n need to
> be established.
>
>
>
> [1] https://tools.ietf.org/html/rfc5764#section-5.2
>
>    Because of packet reordering, packets protected by the previous set
>
>    of keys can appear on the wire after the handshake has completed.  To
>
>    compensate for this fact, receivers SHOULD maintain both sets of keys
>
>    for some time in order to be able to decrypt and verify older
>
>    packets.  The keys should be maintained for the duration of the
>
>    maximum segment lifetime (MSL).
>
> =E2=80=A6
>
> If the authentication check on the packet fails
>
>    and no MKI is being used, then the receiver MAY process the packet
>
>    with the older set of keys.  If that authentication check indicates
>
>    that the packet is valid, the packet should be accepted; otherwise,
>
>    the packet MUST be discarded and rejected.
>
>
>
This actually talks about DTLS-SRTP re-key that I have implemented before.
What is unclear to me is how DTLS is dealing with multiple DTLS sessions
(i.e. DTLS data and alerts) on the same transport connection. I cannot find
any documentation or implementations of such feature.
_____________
Roman Shpount

--001a11403ab2c8316b050f383b5c
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Feb 16, 2015 at 9:44 AM, Makaraju, Maridi Raju (Raju) <span di=
r=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=3D=
"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br></div></=
div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204=
,204);border-left-style:solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div><span class=3D"">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:rgb(31,73,125)">=
&gt;</span><span lang=3D"EN-GB">As far as I know the only way to demultiple=
x DTLS packets is based on the underlying transport. &gt;Unless you can
<span style=3D"color:rgb(31,73,125)">&gt;</span>guarantee that no packets f=
rom the previous DTLS connection will arrive on the &gt;new connection reus=
ing the same
<span style=3D"color:rgb(31,73,125)">&gt;</span>underlying transport some i=
nteresting problems can occur.</span><br></p></span><p class=3D"MsoNormal">=
<span style=3D"font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,7=
3,125)">[Raju] I believe packets from old DTLS connection appearing on new =
will be dropped the same way DTLS rekeying handles packets using old key af=
ter MSL timeout
 [1]. So, I don=E2=80=99t think a new TCP/SCTP transport connection need to=
 be established.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">[1]
<a href=3D"https://tools.ietf.org/html/rfc5764#section-5.2" target=3D"_blan=
k">https://tools.ietf.org/html/rfc5764#section-5.2</a><u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-family:&#39;Courier New&#39;;col=
or:black">=C2=A0=C2=A0 Because of packet reordering, packets protected by t=
he previous set<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&#39;Courier New&#39;;col=
or:black">=C2=A0=C2=A0 of keys can appear on the wire after the handshake h=
as completed.=C2=A0 To<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&#39;Courier New&#39;;col=
or:black">=C2=A0=C2=A0 compensate for this fact, receivers SHOULD maintain =
both sets of keys<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&#39;Courier New&#39;;col=
or:black">=C2=A0=C2=A0 for some time in order to be able to decrypt and ver=
ify older<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&#39;Courier New&#39;;col=
or:black">=C2=A0=C2=A0 packets.=C2=A0 The keys should be maintained for the=
 duration of the<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&#39;Courier New&#39;;col=
or:black">=C2=A0=C2=A0 maximum segment lifetime (MSL).<u></u><u></u></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">=E2=80=A6<u></u><u></u></span></p>
<pre><span style=3D"font-size:12pt;color:black">If the authentication check=
 on the packet fails<u></u><u></u></span></pre>
<pre><span style=3D"font-size:12pt;color:black">=C2=A0=C2=A0 and no MKI is =
being used, then the receiver MAY process the packet<u></u><u></u></span></=
pre>
<pre><span style=3D"font-size:12pt;color:black">=C2=A0=C2=A0 with the older=
 set of keys.=C2=A0 If that authentication check indicates<u></u><u></u></s=
pan></pre>
<pre><span style=3D"font-size:12pt;color:black">=C2=A0=C2=A0 that the packe=
t is valid, the packet should be accepted; otherwise,<u></u><u></u></span><=
/pre>
<pre><span style=3D"font-size:12pt;color:black">=C2=A0=C2=A0 the packet MUS=
T be discarded and rejected.<u></u><u></u></span></pre>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>This actually talks about DTLS-SRTP re-key that I have implemented before.=
 What is unclear to me is how DTLS is dealing with multiple DTLS sessions (=
i.e. DTLS data and alerts) on the same transport connection. I cannot find =
any documentation or implementations of such feature.</div><div><div class=
=3D"gmail_signature">_____________<br>Roman Shpount</div></div><br><div>=C2=
=A0</div></div></div></div>

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From: Simon Perreault <sperreault@jive.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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On Mon, Feb 16, 2015 at 12:47 PM, Roman Shpount <roman@telurix.com> wrote:
>
> I did implement DTLS re-key / re-negotiation as specified. Then I tested
>> it. It works. I don't know what problem you're alluding to.
>>
>
> How do you deal with late DTLS data packets or alerts for the previous
> connection received on the new connection? If it is an open source, you can
> simply point me to the relevant code.
>

This may just be a terminology issue... DTLS re-key / re-negotiation
doesn't create a new connection.

I do know how to implement DTLS-SRTP re-key. It is dealing with new DTLS
> session running over existing connection where packets from the previous
> DTLS session can be present which is not completely clear to me.
>

Packets from an unrelated DTLS connection will simply be discarded. What
else are you looking for?

Simon

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Mon, Feb 16, 2015 at 12:47 PM, Roman Shpount <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;=
</span> wrote:<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div class=3D"gmail_quote"><di=
v><div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:=
solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">I did i=
mplement DTLS re-key / re-negotiation as specified. Then I tested it. It wo=
rks. I don&#39;t know what problem you&#39;re alluding to.</div></div></blo=
ckquote><div><br></div></div></div><div>How do you deal with late DTLS data=
 packets or alerts for the previous connection received on the new connecti=
on? If it is an open source, you can simply point me to the relevant code.<=
/div></div></blockquote><div><br></div><div>This may just be a terminology =
issue... DTLS re-key / re-negotiation doesn&#39;t create a new connection.=
=C2=A0</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=3D"gma=
il_quote"><div>I do know how to implement DTLS-SRTP re-key. It is dealing w=
ith new DTLS session running over existing connection where packets from th=
e previous DTLS session can be present which is not completely clear to me.=
</div></div></blockquote></div><br>Packets from an unrelated DTLS connectio=
n will simply be discarded. What else are you looking for?</div><div class=
=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Simon</div></div>

--001a11c29ce49d5882050f384f61--


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From: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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--089e0160a9d2c0b8b5050f385338
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On Mon, Feb 16, 2015 at 12:51 PM, Roman Shpount <roman@telurix.com> wrote:

> What is unclear to me is how DTLS is dealing with multiple DTLS sessions
> (i.e. DTLS data and alerts) on the same transport connection. I cannot find
> any documentation or implementations of such feature.


You can't do this, just like you can't have more than one TCP connection
per 5-tuple.

Simon

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Mon, Feb 16, 2015 at 12:51 PM, Roman Shpount <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex">What is unclear to me is h=
ow DTLS is dealing with multiple DTLS sessions (i.e. DTLS data and alerts) =
on the same transport connection. I cannot find any documentation or implem=
entations of such feature.</blockquote></div><br>You can&#39;t do this, jus=
t like you can&#39;t have more than one TCP connection per 5-tuple.</div><d=
iv class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Simon</div></=
div>

--089e0160a9d2c0b8b5050f385338--


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From: Jozsef Vass <jvass@twilio.com>
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Yes, that would work for us. Our main use case is audio only. In addition,
implementers may add PLC (Appendix I) to G.711.

Jozsef

Jozsef Vass
SDK Engineering
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MOBILE(415) 216-6603EMAILjvass@twilio.com
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On Thu, Feb 12, 2015 at 4:01 PM, Justin Uberti <juberti@google.com> wrote:

> If we do any FEC with PCMU, it would probably be 2198-style for simplicity
> (counter to what the current text says). Is this something that you would
> use?
>
> On Wed, Feb 11, 2015 at 11:44 AM, Jozsef Vass <jvass@twilio.com> wrote:
>
>> Later, in 4.1 Recommended Mechanisms:
>>
>>    When using constant-bitrate codecs, e.g.  PCMU, use of [RFC2198 <http://tools.ietf.org/html/rfc2198>]
>>    redundant encoding is NOT RECOMMENDED, as this will result in a
>>    potentially significant bitrate increase.  Furthermore, suddenly
>>    increasing the bitrate to deal with packet losses may actually make
>>    things worse.
>>
>>    Because of the lower packet rate of audio encodings, usually a single
>>    packet per frame, use of a separate FEC stream comes with a higher
>>    overhead than other mechanisms, and therefore is NOT RECOMMENDED.
>>
>>
>> So what is recommended for PCMU?
>>
>> Jozsef
>>
>>
>> Jozsef Vass
>> SDK Engineering
>> [image: twilio] <http://www.twilio.com/?utm_source=email_signature>
>> MOBILE(415) 216-6603EMAILjvass@twilio.com
>> SIGNAL: The modern communications conference
>> <http://signal.twilio.com/?utm_source=employee_signature&utm_medium=email&utm_content=signal>
>> May 19/20 2015, Fort Mason Festival Pavilion, SF, CA
>> <http://signal.twilio.com/?utm_source=employee_signature&utm_medium=email&utm_content=signal>
>> <https://www.twilio.com/> <https://www.facebook.com/TeamTwilio>
>> <https://plus.google.com/u/0/+twilio/posts> <https://twitter.com/twilio>
>> <https://www.linkedin.com/company/twilio-inc.>
>> <http://www.glassdoor.com/Overview/Working-at-Twilio-EI_IE410790.11,17.htm>
>>
>> On Wed, Feb 11, 2015 at 11:39 AM, Simon Perreault <sperreault@jive.com>
>> wrote:
>>
>>>
>>> On Wed, Feb 11, 2015 at 12:01 PM, Jozsef Vass <jvass@twilio.com> wrote:
>>>
>>>> I just wondering what kind of FEC is recommended (or will be supported)
>>>> for G.711. The draft does not recommend neither redundant encoding nor a
>>>> separate FEC stream.
>>>
>>>
>>> Isn't this what you're looking for?
>>>
>>> 3.1 <http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00#section-3.1>.  Separate FEC Stream
>>>
>>>    This approach, as described in [RFC5956], Section 4.3 <http://tools.ietf.org/html/rfc5956#section-4.3>, sends FEC
>>>    packets as an independent SSRC-multiplexed stream, with its own SSRC
>>>    and payload type.  While by far the most flexible, each FEC packet
>>>    will have its own IP+UDP+RTP+FEC header, leading to additional
>>>    overhead of the FEC stream.
>>>
>>>
>>> Simon
>>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

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<div dir=3D"ltr">Yes, that would work for us. Our main use case is audio on=
ly. In addition, implementers may add PLC (Appendix I) to G.711.<div><br></=
div><div>Jozsef</div></div><div class=3D"gmail_extra"><br clear=3D"all"><di=
v><div class=3D"gmail_signature"><div dir=3D"ltr"><div style=3D"color:rgb(4=
5,44,42);font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;fo=
nt-size:1em;line-height:20px;padding-bottom:2px">Jozsef Vass</div><div styl=
e=3D"font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;font-s=
ize:0.95em;padding-bottom:15px;color:rgb(178,173,162)">SDK Engineering</div=
><div style=3D"color:rgb(45,44,42);font-family:&#39;Helvetica Neue&#39;,Hel=
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background:0px 0px" target=3D"_blank"><span style=3D"color:rgb(235,100,97);=
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00">I</span><span style=3D"color:rgb(214,67,117);font-weight:900">G</span><=
span style=3D"color:rgb(122,99,126);font-weight:900">N</span><span style=3D=
"color:rgb(66,99,113);font-weight:900">A</span><span style=3D"color:rgb(43,=
154,137);font-weight:900">L</span>: The modern communications conference</a=
></div><div style=3D"font-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,s=
ans-serif;color:rgb(178,173,162);font-size:12px;padding-bottom:20px"><a hre=
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um=3Demail&amp;utm_content=3Dsignal" style=3D"color:rgb(178,173,162);text-d=
ecoration:none!important;background:0px 0px" target=3D"_blank">May 19/20 20=
15, Fort Mason Festival Pavilion, SF, CA</a></div><div style=3D"color:rgb(4=
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." style=3D"color:rgb(66,139,202);text-decoration:none;background:0px 0px" =
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x 16px;border-style:solid;border-color:rgb(216,214,208);border-width:2px 0p=
x"><a href=3D"http://www.glassdoor.com/Overview/Working-at-Twilio-EI_IE4107=
90.11,17.htm" style=3D"color:rgb(66,139,202);text-decoration:none;backgroun=
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/div></div></div></div>
<br><div class=3D"gmail_quote">On Thu, Feb 12, 2015 at 4:01 PM, Justin Uber=
ti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_b=
lank">juberti@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><div dir=3D"ltr">If we do any FEC with PCMU, it would probably be 219=
8-style for simplicity (counter to what the current text says). Is this som=
ething that you would use?</div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Wed, Feb 11, 2015 at 11:44 AM, Jozsef Vass <span dir=3D=
"ltr">&lt;<a href=3D"mailto:jvass@twilio.com" target=3D"_blank">jvass@twili=
o.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"l=
tr">Later, in 4.1 Recommended Mechanisms:<div><br></div><div><pre style=3D"=
font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   When us=
ing constant-bitrate codecs, e.g.  PCMU, use of [<a href=3D"http://tools.ie=
tf.org/html/rfc2198" title=3D"&quot;RTP Payload for Redundant Audio Data&qu=
ot;" target=3D"_blank">RFC2198</a>]
   redundant encoding is NOT RECOMMENDED, as this will result in a
   potentially significant bitrate increase.  Furthermore, suddenly
   increasing the bitrate to deal with packet losses may actually make
   things worse.

   Because of the lower packet rate of audio encodings, usually a single
   packet per frame, use of a separate FEC stream comes with a higher
   overhead than other mechanisms, and therefore is NOT RECOMMENDED.</pre><=
div><br></div><div>So what is recommended for PCMU?</div><span><font color=
=3D"#888888"><div><br></div><div>Jozsef</div><div><br></div></font></span><=
/div></div><div class=3D"gmail_extra"><span><br clear=3D"all"><div><div><di=
v dir=3D"ltr"><div style=3D"color:rgb(45,44,42);font-family:&#39;Helvetica =
Neue&#39;,Helvetica,Arial,sans-serif;font-size:1em;line-height:20px;padding=
-bottom:2px">Jozsef Vass</div><div style=3D"font-family:&#39;Helvetica Neue=
&#39;,Helvetica,Arial,sans-serif;font-size:0.95em;padding-bottom:15px;color=
:rgb(178,173,162)">SDK Engineering</div><div style=3D"color:rgb(45,44,42);f=
ont-family:&#39;Helvetica Neue&#39;,Helvetica,Arial,sans-serif;font-size:14=
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0">A</span><span style=3D"color:rgb(43,154,137);font-weight:900">L</span>: =
The modern communications conference</a></div><div style=3D"font-family:&#3=
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, CA</a></div><div style=3D"color:rgb(45,44,42);font-family:&#39;Helvetica =
Neue&#39;,Helvetica,Arial,sans-serif;font-size:14px;line-height:20px;paddin=
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3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signature/img/icon-tw=
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<td style=3D"padding:12px 16px;border-style:solid;border-color:rgb(216,214,=
208);border-width:2px 0px"><a href=3D"https://www.facebook.com/TeamTwilio" =
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le:solid;border-color:rgb(216,214,208);border-width:2px 0px"><a href=3D"htt=
ps://www.linkedin.com/company/twilio-inc." style=3D"color:rgb(66,139,202);t=
ext-decoration:none;background:0px 0px" target=3D"_blank"><img src=3D"https=
://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signature/img/ico=
n-linkedin.png" width=3D"20" style=3D"border:0px;vertical-align:middle"></a=
></td><td style=3D"padding:12px 12px 12px 16px;border-style:solid;border-co=
lor:rgb(216,214,208);border-width:2px 0px"><a href=3D"http://www.glassdoor.=
com/Overview/Working-at-Twilio-EI_IE410790.11,17.htm" style=3D"color:rgb(66=
,139,202);text-decoration:none;background:0px 0px" target=3D"_blank"><img s=
rc=3D"https://s3.amazonaws.com/ahoy-assets.twilio.com/Emails/Company-Signat=
ure/img/icon-glassdoor.png" width=3D"20" style=3D"border:0px;vertical-align=
:middle"></a></td></tr></tbody></table></div></div></div></div>
<br></span><div><div><div class=3D"gmail_quote">On Wed, Feb 11, 2015 at 11:=
39 AM, Simon Perreault <span dir=3D"ltr">&lt;<a href=3D"mailto:sperreault@j=
ive.com" target=3D"_blank">sperreault@jive.com</a>&gt;</span> wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br=
><div class=3D"gmail_quote">On Wed, Feb 11, 2015 at 12:01 PM, Jozsef Vass <=
span dir=3D"ltr">&lt;<a href=3D"mailto:jvass@twilio.com" target=3D"_blank">=
jvass@twilio.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:=
rgb(204,204,204);border-left-style:solid;padding-left:1ex">I just wondering=
 what kind of FEC is recommended (or will be supported) for G.711. The draf=
t does not recommend neither redundant encoding nor a separate FEC stream.<=
/blockquote><div><br></div><div>Isn&#39;t this what you&#39;re looking for?=
=C2=A0</div></div><br><pre style=3D"font-size:1em;margin-top:0px;margin-bot=
tom:0px;color:rgb(0,0,0)"><span style=3D"line-height:0pt;display:inline;fon=
t-size:1em;font-weight:bold"><h3 style=3D"line-height:0pt;display:inline;fo=
nt-size:1em"><a name=3D"14b803debb942a60_14b7a2c5f2019384_14b7a271b939f06e_=
section-3.1" href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-fec-00#se=
ction-3.1" style=3D"color:black;text-decoration:none" target=3D"_blank">3.1=
</a>.  Separate FEC Stream</h3></span>

   This approach, as described in <a href=3D"http://tools.ietf.org/html/rfc=
5956#section-4.3" target=3D"_blank">[RFC5956], Section=C2=A04.3</a>, sends =
FEC
   packets as an independent SSRC-multiplexed stream, with its own SSRC
   and payload type.  While by far the most flexible, each FEC packet
   will have its own IP+UDP+RTP+FEC header, leading to additional
   overhead of the FEC stream.</pre><span><font color=3D"#888888"><pre styl=
e=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br><=
/pre>Simon</font></span></div></div>
</blockquote></div><br></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Simon Perreault" <sperreault@jive.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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Subject: Re: [rtcweb] SDP-SCTP: Does re-establishment of DTLS connection affect underlying SCTP or TCP connection?
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: SCTP-SDP: Mux category for SDP sctp-port and max-message-size attributes
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Date: Sun, 22 Feb 2015 16:20:02 +0000
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Subject: [rtcweb] FW: SCTP-SDP: Mux category for SDP sctp-port and max-message-size attributes
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FYI,

I sent this to MMUSIC, but it may be of interest for RTCWEB folks too.

If you have any comments, please post them on the MMUSIC list.

Regards,

Christer


From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christer Holmber=
g
Sent: 22 February 2015 18:19
To: mmusic@ietf.org
Subject: [MMUSIC] SCTP-SDP: Mux category for SDP sctp-port and max-message-=
size attributes

Hi,

I've been asked to include the MUX category (as defined in draft-ietf-mmusi=
c-sdp-mux-attributes) for the SDP attributes defined in the SCTP-SDP draft:=
 sctp-port and max-message-size.

For 'SCTP' and 'SCTP/DTLS' I assume it is rather straight forward. The sctp=
-port attribute is not used to begin with, and the max-message-size applies=
 to the transport layer SCTP association.

'(UDP | TCP)/DTLS/SCTP' is a little different.  The draft says (or, *will* =
say, based on the agreement in Honolulu) that multiple SCTP associations ov=
er a single '(UDP | TCP)/DTLS' connection is outside the scope of the docum=
ent, which means it is not possible to have multiple '(UDP | TCP)/DTLS/SCTP=
' m- lines within a BUNDLE group.

Currently, unless I've missed something, there is however no category for a=
ttributes that can't be bundled, eventhough one could perhaps claim that th=
e NOT RECOMMENDED category fits. So, I see three options:


1)      Allocate SPECIAL category, with a description saying that they can'=
t be bundled

2)      Allocate NOT RECOMMENDED category

3)      Define a new mux category, for attributes that can't be bundled

Comments? My personal preference would be either 2) or 3).

SCTP:
=3D=3D=3D=3D=3D

Attribute:                          Mux category:

sctp-port                           Not applicable
max-message-size          TRANSPORT


SCTP/DTLS:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

sctp-port                           Not applicable
max-message-size          TRANSPORT

(UDP | TCP)/DTLS/SCTP:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Attribute:                          Mux category:

sctp-port                           SPECIAL | New category
max-message-size          SPECIAL | New category

Regards,

Christer


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<body lang=3D"EN-GB" link=3D"#0563C1" vlink=3D"#954F72">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">FYI,<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">I sent this to MMUSIC,=
 but it may be of interest for RTCWEB folks too.<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">If you have any commen=
ts, please post them on the MMUSIC list.<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">Regards,<o:p></o:p></s=
pan></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">Christer<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><a name=3D"_MailEndCompose"><span style=3D"color:#1F=
497D"><o:p>&nbsp;</o:p></span></a></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"mso-fareast-languag=
e:EN-GB">From:</span></b><span lang=3D"EN-US" style=3D"mso-fareast-language=
:EN-GB"> mmusic [mailto:mmusic-bounces@ietf.org]
<b>On Behalf Of </b>Christer Holmberg<br>
<b>Sent:</b> 22 February 2015 18:19<br>
<b>To:</b> mmusic@ietf.org<br>
<b>Subject:</b> [MMUSIC] SCTP-SDP: Mux category for SDP sctp-port and max-m=
essage-size attributes<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I&#8217;ve been asked to include the MUX category (a=
s defined in draft-ietf-mmusic-sdp-mux-attributes) for the SDP attributes d=
efined in the SCTP-SDP draft: sctp-port and max-message-size.<o:p></o:p></p=
>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">For &#8216;SCTP&#8217; and &#8216;SCTP/DTLS&#8217; I=
 assume it is rather straight forward. The sctp-port attribute is not used =
to begin with, and the max-message-size applies to the transport layer SCTP=
 association.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&#8216;(UDP | TCP)/DTLS/SCTP&#8217; is a little diff=
erent. &nbsp;The draft says (or, *<b>will</b>* say, based on the agreement =
in Honolulu) that multiple SCTP associations over a single &#8216;(UDP | TC=
P)/DTLS&#8217; connection is outside the scope of the document,
 which means it is not possible to have multiple &#8216;(UDP | TCP)/DTLS/SC=
TP&#8217; m- lines within a BUNDLE group.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Currently, unless I&#8217;ve missed something, there=
 is however no category for attributes that can&#8217;t be bundled, eventho=
ugh one could perhaps claim that the NOT RECOMMENDED category fits. So, I s=
ee three options:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">1)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Allocate SPECIAL category, with a description sayin=
g that they can&#8217;t be bundled<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">2)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Allocate NOT RECOMMENDED category<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">3)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Define a new mux category, for attributes that can&=
#8217;t be bundled
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Comments? My personal preference would be either 2) =
or 3).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">SCTP:<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Attribute:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Mux category:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">sctp-port&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Not applicable<o:p></o:p></p>
<p class=3D"MsoNormal">max-message-size&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; TRANSPORT<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">SCTP/DTLS:<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">sctp-port&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Not applicable<o:p></o:p></p>
<p class=3D"MsoNormal">max-message-size&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; TRANSPORT<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">(UDP | TCP)/DTLS/SCTP:<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Attribute:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Mux category:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">sctp-port&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SPECIAL | New category&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;
<o:p></o:p></p>
<p class=3D"MsoNormal">max-message-size&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; SPECIAL | New category<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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Date: Sun, 22 Feb 2015 12:43:41 -0500
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: [rtcweb] Offer/Answer and DTLS
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Let me try to explain what my issue is with offer/answer, DTLS and multiple
sessions over the same transport. First of all, as far as I understand,
based on the specifications, an end point is supposed to start a new DTLS
session (not re-key) if the setup role or fingerprint changes.

Imagine the following situation: An end point has no intent to create a new
DTLS session, it send an offer with the existing fingerprint and actpass
setup role. Since it does not intend to change anything DTLS related,
existing DTLS session continues to run. Remote party maintains the
transport parameters but changes either role or fingerprint. This implies
new DTLS session. So remote party closes the existing DTLS session (sends
an alert) and starts negotiating a new session (potentially sends Client
Hello). All of this happens at the same time as the answer is being
generated and sent out to the offering pary. Offering party might not get
this answer before new DTLS session handshake is received, so this
handshake is most likely to be ignored, potentially delaying the session
setup by DTLS timeout.

Another problem: Offering party desires to change the DTLS session settings
without changing the transport parameters and sends a new offer with new
fingerprint and/or setup role. It cannot simply close the existing DTLS
session, since offer might still fail for unrelated reason. If setup role
is passive or actpass, offering party can still get the Client Hello before
the successful answer was received, which means it should be able to
process DTLS client offers while still processing data for the existing
DTLS session.

based on this, it looks like anything that runs offer/answer in combination
with DTLS, should be able to handle a new DTLS session setup while an
existing session is still running. At the very least, it should queue the
Client Hello packets to be processed when signaling level decides to start
a new session. Alternative would be loss of data or media for the duration
of DTLS timeout.

Finally, data and alerts from the previous session can be received after
new session is setup. These two DTLS sessions can be demultiplexed based on
their signatures or packets from an old session can simply be ignored since
they failed the signature verification, which would be much less graceful.
I think the better solution would be for an end point to maintain the list
of recent DTLS sessions and try to dispatch a DTLS packet to the right
session based on the packet signature. Having some sort of session
indicator (similar to MKI) would make this process more transparent and
less resource intensive but I do not think this is available for DTLS.

So, my questions are:
1. Am I missing something in regard to DTLS session mulitplexing?
2. Is DTLS session multiplexing defined somewhere?
3. Have anybody implemented something similar to this?

I am primarily interested in this in regard to WebRTC implementations. In
particular I am interested in what is going to be supported or implemented
by web browsers. As far as I know, current browser implementations do not
support such renegotiation at all. If this is the intent for the
foreseeable future, then prohibiting changes (at least with a SHOULD) to
DTLS session without change to the underlying transport might not be a bad
option.
_____________
Roman Shpount

On Mon, Feb 16, 2015 at 3:01 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>
>
> On Mon, Feb 16, 2015 at 12:51 PM, Roman Shpount <roman@telurix.com> wrote=
:
>
> What is unclear to me is how DTLS is dealing with multiple DTLS sessions
> (i.e. DTLS data and alerts) on the same transport connection. I cannot fi=
nd
> any documentation or implementations of such feature.
>
>
> *[Raju] **DTLS is a protocol which is typically runs in application-space=
. You typically run just one DTLS session on top of underlying transport la=
yer (TCP or UDP). As explained by Simon, DTLS rekeying does not result in a=
 new DTLS association, but rather new handshake sequence on top of existing=
 association. *
>
> *However, during subsequent SDP O/A role change may trigger a new DTLS as=
sociation to be setup. In such a case previous association should be gone a=
nd any transport layer packets (TCP or UDP) from the previous DTLS associat=
ion may arrive on new DTLS association, but I think they will and should be=
 discarded due to MAC failure (with bad_record_mac alert?).*
>
> *Please note that this is not unique to TCP; though unreliable, UDP trans=
port could also receive delayed pkts from previous DTLS association. Or som=
eone could simply inject UPD pkts into mix. So, re-establishing TCP connect=
ion is not a solution and won=E2=80=99t help when UDP is used.*
>
> *So, IMO a DTLS implementation should handle such out-of-band (OOB) pkts =
in a graceful way (i.e. without dropping the DTLS association).*
>
>
>
> *Hope this helps!*
>
> *Raju*
>
>
>
>

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<div dir=3D"ltr">Let me try to explain what my issue is with offer/answer, =
DTLS and multiple sessions over the same transport. First of all, as far as=
 I understand, based on the specifications, an end point is supposed to sta=
rt a new DTLS session (not re-key) if the setup role or fingerprint changes=
.=C2=A0<div><br></div><div>Imagine the following situation: An end point ha=
s no intent to create a new DTLS session, it send an offer with the existin=
g fingerprint and actpass setup role. Since it does not intend to change an=
ything DTLS related, existing DTLS session continues to run. Remote party m=
aintains the transport parameters but changes either role or fingerprint. T=
his implies new DTLS session. So remote party closes the existing DTLS sess=
ion (sends an alert) and starts negotiating a new session (potentially send=
s Client Hello). All of this happens at the same time as the answer is bein=
g generated and sent out to the offering pary. Offering party might not get=
 this answer before new DTLS session handshake is received, so this handsha=
ke is most likely to be ignored, potentially delaying the session setup by =
DTLS timeout.<div><br></div><div>Another problem: Offering party desires to=
 change the DTLS session settings without changing the transport parameters=
 and sends a new offer with new fingerprint and/or setup role. It cannot si=
mply close the existing DTLS session, since offer might still fail for unre=
lated reason. If setup role is passive or actpass, offering party can still=
 get the Client Hello before the successful answer was received, which mean=
s it should be able to process DTLS client offers while still processing da=
ta for the existing DTLS session.<br><div class=3D"gmail_extra"><br></div><=
div class=3D"gmail_extra">based on this, it looks like anything that runs o=
ffer/answer in combination with DTLS, should be able to handle a new DTLS s=
ession setup while an existing session is still running. At the very least,=
 it should queue the Client Hello packets to be processed when signaling le=
vel decides to start a new session. Alternative would be loss of data or me=
dia for the duration of DTLS timeout.<br></div><div class=3D"gmail_extra"><=
br></div><div class=3D"gmail_extra">Finally, data and alerts from the previ=
ous session can be received after new session is setup. These two DTLS sess=
ions can be demultiplexed based on their signatures or packets from an old =
session can simply be ignored since they failed the signature verification,=
 which would be much less graceful. I think the better solution would be fo=
r an end point to maintain the list of recent DTLS sessions and try to disp=
atch a DTLS packet to the right session based on the packet signature. Havi=
ng some sort of session indicator (similar to MKI) would make this process =
more transparent and less resource intensive but I do not think this is ava=
ilable for DTLS.</div><div class=3D"gmail_extra"><br></div><div class=3D"gm=
ail_extra">So, my questions are:<br></div><div class=3D"gmail_extra">1. Am =
I missing something in regard to DTLS session mulitplexing?</div><div class=
=3D"gmail_extra">2. Is DTLS session multiplexing defined somewhere?</div><d=
iv class=3D"gmail_extra">3. Have anybody implemented something similar to t=
his?</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">I=
 am primarily interested in this in regard to WebRTC implementations. In pa=
rticular I am interested in what is going to be supported or implemented by=
 web browsers. As far as I know, current browser implementations do not sup=
port such renegotiation at all. If this is the intent for the foreseeable f=
uture, then prohibiting changes (at least with a SHOULD) to DTLS session wi=
thout change to the underlying transport might not be a bad option.</div><d=
iv class=3D"gmail_extra"><div><div class=3D"gmail_signature">_____________<=
br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Mon, Feb 16, 2015 at 3:01 PM, Makaraju, M=
aridi Raju (Raju) <span dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alc=
atel-lucent.com" target=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-=
left-style:solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<div style=3D"border-style:none none none solid;border-left-color:blue;bord=
er-left-width:1.5pt;padding:0in 0in 0in 4pt">
<div>
<div><span class=3D"">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Mon, Feb 16, 2015 at 12:51 PM, Roman Shpount &lt;=
<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a=
>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">What is unclear to me is how DTLS is dealing with mu=
ltiple DTLS sessions (i.e. DTLS data and alerts) on the same transport conn=
ection. I cannot find any documentation or implementations of such feature.=
<u></u><u></u></p>
</div>
</span><pre><br><b><i><span style=3D"font-size:11pt;font-family:Calibri,san=
s-serif;color:rgb(31,73,125)">[Raju] </span></i></b><b><i><span style=3D"fo=
nt-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">DTLS is a=
 protocol which is typically runs in application-space. You typically run j=
ust one DTLS session on top of underlying transport layer (TCP or UDP). As =
explained by Simon, DTLS rekeying does not result in a new DTLS association=
, but rather new handshake sequence on top of existing association. <u></u>=
<u></u></span></i></b></pre>
<pre><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)">However, during subsequent SDP O/A role change may trigg=
er a new DTLS association to be setup. In such a case previous association =
should be gone and any transport layer packets (TCP or UDP) from the previo=
us DTLS association may arrive on new DTLS association, but I think they wi=
ll and should be discarded due to MAC failure (with bad_record_mac alert?).=
<u></u><u></u></span></i></b></pre>
<pre><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)">Please note that this is not unique to TCP; though unrel=
iable, UDP transport could also receive delayed pkts from previous DTLS ass=
ociation. Or someone could simply inject UPD pkts into mix. So, re-establis=
hing TCP connection is not a solution and won=E2=80=99t help when UDP is us=
ed.<u></u><u></u></span></i></b></pre>
<pre><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)">So, IMO a DTLS implementation should handle such out-of-=
band (OOB) pkts in a graceful way (i.e. without dropping the DTLS associati=
on).<u></u><u></u></span></i></b></pre>
<pre><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></i></b></pre>
<pre><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)">Hope this helps!<span class=3D""><font color=3D"#888888"=
><u></u><u></u></font></span></span></i></b></pre><span class=3D""><font co=
lor=3D"#888888">
<pre><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)">Raju<u></u><u></u></span></i></b></pre>
<pre><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></i></b></pre>
</font></span></div>
</div>
</div>
</div>
</div>

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

--089e01229aaa93efd9050fb0d37c--


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Subject: Re: [rtcweb] Offer/Answer and DTLS
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On 23 February 2015 at 06:43, Roman Shpount <roman@telurix.com> wrote:
> Let me try to explain what my issue is with offer/answer, DTLS and multiple
> sessions over the same transport. First of all, as far as I understand,
> based on the specifications, an end point is supposed to start a new DTLS
> session (not re-key) if the setup role or fingerprint changes.

Or you could not do that.  The feature exists to support a change to a
new communication endpoint.  A new DTLS association should not be
needed if the two endpoints are the same.  A different endpoint will
have different a transport addresses and different ICE credentials,
meaning that the whole demux issue is addressed at lower layers of the
stack.

The only thing we would need to do in that case is say that you MUST
NOT change a=setup during renegotiation unless the endpoint (and
therefore the transport parameters) needs to change.


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--20cf304352c819e71b050fb1cad4
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On Sun, Feb 22, 2015 at 1:39 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 23 February 2015 at 06:43, Roman Shpount <roman@telurix.com> wrote:
> > Let me try to explain what my issue is with offer/answer, DTLS and
> multiple
> > sessions over the same transport. First of all, as far as I understand,
> > based on the specifications, an end point is supposed to start a new DTLS
> > session (not re-key) if the setup role or fingerprint changes.
>
> Or you could not do that.  The feature exists to support a change to a
> new communication endpoint.  A new DTLS association should not be
> needed if the two endpoints are the same.  A different endpoint will
> have different a transport addresses and different ICE credentials,
> meaning that the whole demux issue is addressed at lower layers of the
> stack.
>
> The only thing we would need to do in that case is say that you MUST
> NOT change a=setup during renegotiation unless the endpoint (and
> therefore the transport parameters) needs to change.
>

I actually agree. I would really like to see that specified somewhere that
*fingerprint* and *setup role* SHOULD NOT, or better yet, MUST NOT change
if end points and transport parameters did not change as well. This will
address all necessary use cases and simplify the implementations. In fact I
have suggested this on multiple occasions. Other people keep insisting that
these things can change independently.

Alternatively, which I do like a lot less, it is possible to define how
DTLS sessions can be setup and demultiplexed over the same connection.
Unless someone provides a good reason why this is needed and comes up with
a plausible implementation I would prefer not to have multiple DTLS
sessions over the same connection.
_____________
Roman Shpount

--20cf304352c819e71b050fb1cad4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Sun, Feb 22, 2015 at 1:39 PM, Martin Thomson <span dir=3D"ltr">&lt;=
<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomso=
n@gmail.com</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;=
padding-left:1ex"><span class=3D"">On 23 February 2015 at 06:43, Roman Shpo=
unt &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt; wrot=
e:<br>
&gt; Let me try to explain what my issue is with offer/answer, DTLS and mul=
tiple<br>
&gt; sessions over the same transport. First of all, as far as I understand=
,<br>
&gt; based on the specifications, an end point is supposed to start a new D=
TLS<br>
&gt; session (not re-key) if the setup role or fingerprint changes.<br>
<br>
</span>Or you could not do that.=C2=A0 The feature exists to support a chan=
ge to a<br>
new communication endpoint.=C2=A0 A new DTLS association should not be<br>
needed if the two endpoints are the same.=C2=A0 A different endpoint will<b=
r>
have different a transport addresses and different ICE credentials,<br>
meaning that the whole demux issue is addressed at lower layers of the<br>
stack.<br>
<br>
The only thing we would need to do in that case is say that you MUST<br>
NOT change a=3Dsetup during renegotiation unless the endpoint (and<br>
therefore the transport parameters) needs to change.<br>
</blockquote></div><br></div><div class=3D"gmail_extra">I actually agree. I=
 would really like to see that specified somewhere that <b>fingerprint</b> =
and <b>setup role</b> SHOULD NOT, or better yet, MUST NOT change if end poi=
nts and transport parameters did not change as well. This will address all =
necessary use cases and simplify the implementations. In fact I have sugges=
ted this on multiple occasions. Other people keep insisting that these thin=
gs can change independently.</div><div class=3D"gmail_extra"><br></div><div=
 class=3D"gmail_extra">Alternatively, which I do like a lot less, it is pos=
sible to define how DTLS sessions can be setup and demultiplexed over the s=
ame connection. Unless someone provides a good reason why this is needed an=
d comes up with a plausible implementation I would prefer not to have multi=
ple DTLS sessions over the same connection.</div><div class=3D"gmail_extra"=
><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></=
div><br></div></div>

--20cf304352c819e71b050fb1cad4--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Offer/Answer and DTLS
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Date: Mon, 23 Feb 2015 08:54:07 -0800
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Roman said:

"I actually agree. I would really like to see that specified somewhere that
*fingerprint* and *setup role* SHOULD NOT, or better yet, MUST NOT change
if end points and transport parameters did not change as well. This will
address all necessary use cases and simplify the implementations. In fact I
have suggested this on multiple occasions. Other people keep insisting that
these things can change independently."

[BA] I'm generally supportive of this direction.  Existing implementations
don't support re-negotiation, so in general it won't work anyway.


"Alternatively, which I do like a lot less, it is possible to define how
DTLS sessions can be setup and demultiplexed over the same connection."

[BA] This is much less desirable - and even if the standards work were
done, who would implement it?

On Sun, Feb 22, 2015 at 9:06 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>> Let me try to explain what my issue is with offer/answer, DTLS and
> multiple
> >>> sessions over the same transport. First of all, as far as I understand,
> >>> based on the specifications, an end point is supposed to start a new
> DTLS
> >>> session (not re-key) if the setup role or fingerprint changes.
> >>
> >> Or you could not do that.  The feature exists to support a change to a
> >> new communication endpoint.  A new DTLS association should not be
> >> needed if the two endpoints are the same.  A different endpoint will
> >> have different a transport addresses and different ICE credentials,
> >> meaning that the whole demux issue is addressed at lower layers of the
> >> stack.
> >>
> >> The only thing we would need to do in that case is say that you MUST
> >> NOT change a=setup during renegotiation unless the endpoint (and
> >> therefore the transport parameters) needs to change.
> >
> > I actually agree. I would really like to see that specified somewhere
> that fingerprint and
> > setup role SHOULD NOT, or better yet, MUST NOT change if end points and
> transport
> > parameters did not change as well. This will address all necessary use
> cases and simplify
> > the implementations. In fact I have suggested this on multiple
> occasions. Other people
> > keep insisting that these things can change independently.
>
> The SCTP-SDP draft currently contains the following sentence:
>
>         "Therefore, endpoints SHOULD NOT change the 'active/
>         passive' status during a session, unless they want to establish a
> new
>         DTLS connection."
>
> And, even if a new DTLS connection is established, it of course doesn't
> mean that the active/passive status automatically will be changed - but the
> status will be re-negotiated.
>
>         "If the transport parameters or the key fingerprints change, the
>         endpoints MUST establish a new DTLS connection.  In such case the
>         'active/passive' status of the endpoints will again be determined
>         following the procedures in [RFC4145], and the new status will be
>         used to determine the TLS roles of the endpoints associated with
> the
>         new DTLS connection."
>
> > Alternatively, which I do like a lot less, it is possible to define how
> DTLS sessions can be setup and
> > demultiplexed over the same connection. Unless someone provides a good
> reason why this is
> > needed and comes up with a plausible implementation I would prefer not
> to have multiple
> > DTLS sessions over the same connection.
>
> My understanding is that, when BUNDLE is used, a single DTLS connection
> will be used for all bundled media/data within that BUNDLE group.
>
> I am not even sure it would be allowed to have multiple DTLS connections
> on a single 5-tuple, but I guess some DTLS expert can answer that.
>
> Regards,
>
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--e89a8f3ba43b54dfda050fc441b8
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div>Roman said: </div><div><br></div><div><div class=3D"g=
mail_extra">&quot;I actually agree. I would really like to see that specifi=
ed somewhere that <b>fingerprint</b> and <b>setup role</b> SHOULD NOT, or b=
etter yet, MUST NOT change if end points and transport parameters did not c=
hange as well. This will address all necessary use cases and simplify the i=
mplementations. In fact I have suggested this on multiple occasions. Other =
people keep insisting that these things can change independently.&quot;</di=
v><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">[BA] I&#3=
9;m generally supportive of this direction.=C2=A0 Existing implementations =
don&#39;t support re-negotiation, so in general it won&#39;t work anyway. <=
/div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra"><br></=
div><div class=3D"gmail_extra">&quot;Alternatively, which I do like a lot l=
ess, it is possible to define how DTLS sessions can be setup and demultiple=
xed over the same connection.&quot;</div><div class=3D"gmail_extra"><br></d=
iv><div class=3D"gmail_extra">[BA] This is much less desirable - and even i=
f the standards work were done, who would implement it? <br></div></div></d=
iv><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Feb 22=
, 2015 at 9:06 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@erics=
son.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<span><br>
&gt;&gt;&gt; Let me try to explain what my issue is with offer/answer, DTLS=
 and multiple<br>
&gt;&gt;&gt; sessions over the same transport. First of all, as far as I un=
derstand,<br>
&gt;&gt;&gt; based on the specifications, an end point is supposed to start=
 a new DTLS<br>
&gt;&gt;&gt; session (not re-key) if the setup role or fingerprint changes.=
<br>
&gt;&gt;<br>
&gt;&gt; Or you could not do that.=C2=A0 The feature exists to support a ch=
ange to a<br>
&gt;&gt; new communication endpoint.=C2=A0 A new DTLS association should no=
t be<br>
&gt;&gt; needed if the two endpoints are the same.=C2=A0 A different endpoi=
nt will<br>
&gt;&gt; have different a transport addresses and different ICE credentials=
,<br>
&gt;&gt; meaning that the whole demux issue is addressed at lower layers of=
 the<br>
&gt;&gt; stack.<br>
&gt;&gt;<br>
&gt;&gt; The only thing we would need to do in that case is say that you MU=
ST<br>
&gt;&gt; NOT change a=3Dsetup during renegotiation unless the endpoint (and=
<br>
&gt;&gt; therefore the transport parameters) needs to change.<br>
&gt;<br>
&gt; I actually agree. I would really like to see that specified somewhere =
that fingerprint and<br>
&gt; setup role SHOULD NOT, or better yet, MUST NOT change if end points an=
d transport<br>
&gt; parameters did not change as well. This will address all necessary use=
 cases and simplify<br>
&gt; the implementations. In fact I have suggested this on multiple occasio=
ns. Other people<br>
&gt; keep insisting that these things can change independently.<br>
<br>
</span>The SCTP-SDP draft currently contains the following sentence:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;Therefore, endpoints SHOULD NOT change th=
e &#39;active/<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 passive&#39; status during a session, unless th=
ey want to establish a new<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 DTLS connection.&quot;<br>
<br>
And, even if a new DTLS connection is established, it of course doesn&#39;t=
 mean that the active/passive status automatically will be changed - but th=
e status will be re-negotiated.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;If the transport parameters or the key fi=
ngerprints change, the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 endpoints MUST establish a new DTLS connection.=
=C2=A0 In such case the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &#39;active/passive&#39; status of the endpoint=
s will again be determined<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 following the procedures in [RFC4145], and the =
new status will be<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 used to determine the TLS roles of the endpoint=
s associated with the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 new DTLS connection.&quot;<br>
<span><br>
&gt; Alternatively, which I do like a lot less, it is possible to define ho=
w DTLS sessions can be setup and<br>
&gt; demultiplexed over the same connection. Unless someone provides a good=
 reason why this is<br>
&gt; needed and comes up with a plausible implementation I would prefer not=
 to have multiple<br>
&gt; DTLS sessions over the same connection.<br>
<br>
</span>My understanding is that, when BUNDLE is used, a single DTLS connect=
ion will be used for all bundled media/data within that BUNDLE group.<br>
<br>
I am not even sure it would be allowed to have multiple DTLS connections on=
 a single 5-tuple, but I guess some DTLS expert can answer that.<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5">___________________________________=
____________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--e89a8f3ba43b54dfda050fc441b8--


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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Offer/Answer and DTLS
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--001a1141ca9ea9f56f050fc48020
Content-Type: text/plain; charset=UTF-8

On Mon, Feb 23, 2015 at 12:06 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> The SCTP-SDP draft currently contains the following sentence:
>
>         "Therefore, endpoints SHOULD NOT change the 'active/
>         passive' status during a session, unless they want to establish a
> new
>         DTLS connection."
>
> And, even if a new DTLS connection is established, it of course doesn't
> mean that the active/passive status automatically will be changed - but the
> status will be re-negotiated.
>
>         "If the transport parameters or the key fingerprints change, the
>         endpoints MUST establish a new DTLS connection.  In such case the
>         'active/passive' status of the endpoints will again be determined
>         following the procedures in [RFC4145], and the new status will be
>         used to determine the TLS roles of the endpoints associated with
> the
>         new DTLS connection."
>
>
My understand is that the current draft allows to establish new DTLS
connection over the same transport connection if an endpoint wants to do
this. I would like to make sure that the specification explicitly prohibits
this and states somewhere that fingerprint and setup role MUST NOT change
unless the transport parameters change as well.

Unless this is prohibited, some sort of specification language needs to be
provided stating how multiple DTLS connections can run over a single
5-tuple. This is not currently allowed or defined anywhere and it is
required for DTLS to operate with offer/answer end transport connection
reuse. As I have stated on several occasions, I do not like the option of
allowing multiple DTLS connections over the same transport since it
complicates implementations for no reason. Unless someone provides a
compelling use case why this is needed I strongly believe this should not
be allowed.
_____________
Roman Shpount

--001a1141ca9ea9f56f050fc48020
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Feb 23, 2015 at 12:06 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.hol=
mberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-colo=
r:rgb(204,204,204);border-left-style:solid;padding-left:1ex">The SCTP-SDP d=
raft currently contains the following sentence:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;Therefore, endpoints SHOULD NOT change th=
e &#39;active/<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 passive&#39; status during a session, unless th=
ey want to establish a new<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 DTLS connection.&quot;<br>
<br>
And, even if a new DTLS connection is established, it of course doesn&#39;t=
 mean that the active/passive status automatically will be changed - but th=
e status will be re-negotiated.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;If the transport parameters or the key fi=
ngerprints change, the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 endpoints MUST establish a new DTLS connection.=
=C2=A0 In such case the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &#39;active/passive&#39; status of the endpoint=
s will again be determined<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 following the procedures in [RFC4145], and the =
new status will be<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 used to determine the TLS roles of the endpoint=
s associated with the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 new DTLS connection.&quot;<br>
<span class=3D""><br></span></blockquote><div><br></div><div>My understand =
is that the current draft allows to establish new DTLS connection over the =
same transport connection if an endpoint wants to do this. I would like to =
make sure that the specification explicitly prohibits this and states somew=
here that fingerprint and setup role MUST NOT change unless the transport p=
arameters change as well.</div><div><br></div><div>Unless this is prohibite=
d, some sort of specification language needs to be provided stating how mul=
tiple DTLS connections can run over a single 5-tuple. This is not currently=
 allowed or defined anywhere and it is required for DTLS to operate with of=
fer/answer end transport connection reuse. As I have stated on several occa=
sions, I do not like the option of allowing multiple DTLS connections over =
the same transport since it complicates implementations for no reason. Unle=
ss someone provides a compelling use case why this is needed I strongly bel=
ieve this should not be allowed.</div><div><div class=3D"gmail_signature">_=
____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div=
>

--001a1141ca9ea9f56f050fc48020--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Offer/Answer and DTLS
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 23 Feb 2015 23:38:04 +0100
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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2015-02-23 21:37 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> Doesn't the DTLS spec say that, when a new DTLS connection has been estab=
lished, you need to maintain the keys for the previous connection for a whi=
le, in order to handle late packets?

AFAIK the spec states that just in the case of DTLS rekeying.


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


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--20cf304352c87bc357050fc9ac3c
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On Mon, Feb 23, 2015 at 3:37 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> > I would like to make sure that the specification explicitly prohibits
> this and states
> > somewhere that fingerprint and setup role MUST NOT change unless the
> transport
> > parameters change as well.
>
> As far as SCTP-SDP is concerned, personally I don't (at least not at this
> moment) have a problem with such restriction.
>
> However, when BUNDLE is used, SRTP and SCTP may share a single DTLS
> connection. It could cause problems if SRTP and SCTP define different rules
> when a DTLS connection can be re-established. And, SRTP-DTLS does not
> forbid that fingerprint and setup role change, unless the transport
> parameters also change.
>
> >Unless this is prohibited, some sort of specification language needs to
> be provided stating how
> >multiple DTLS connections can run over a single 5-tuple. This is not
> currently allowed or
> >defined anywhere and it is required for DTLS to operate with offer/answer
> end transport connection reuse.
>
> Doesn't the DTLS spec say that, when a new DTLS connection has been
> established, you need to maintain the keys for the previous connection for
> a while, in order to handle late packets?
>

This is specific to DTLS re-key (not a complete new session establishment)
in combination with SRTP. In case of DTLS re-key with DTLS data frames the
epoch number can be used to reference the correct cipher settings. This
would be used for SCTP or BFCP which run on top of DTLS data.

Unlike re-key, creating a new session over the same transport at the same
time as the existing session continues to run is not defined for DTLS.

>As I have stated on several occasions, I do not like the option of
> allowing multiple DTLS connections
> >over the same transport since it complicates implementations for no
> reason. Unless someone provides
> >a compelling use case why this is needed I strongly believe this should
> not be allowed.
>
> I don't have a use-case, but because of BUNDLE I'd like to have the same
> rules for SRTP and SCTP.
>

I agree that common rules for DTLS offer/answer would be beneficial for
everybody. It is currently either undefined or broken in a similar manner
for both DTLS-SRTP and SCTP draft. New BFCP drafts actually reference
DTLS-SRTP as far as DTLS session setup is concerned, so they will suffer
from the same problem. Furthermore,  I do not think anybody has any intent
to implement DTLS changes without transport changes for any protocol. I
think it would be better to define this correctly at some point vs to keep
repeating the same uncertainty.


> Of course, we could add your suggested restriction also to SRTP...
>
> But, would we then need add the restriction also to other protocols using
> DTLS? E.g. BFCP.
>
> The best thing would be to have generic Offer/Answer rules for DTLS, which
> all usages (SRTP, SCTP, BFCP etc) then reference, but it's too late for
> that...
>

It might not be a bad idea to create such a document and use it to update
DTLS-SRTP and other protocols which use DTLS with offer/answer. I am
actually not sure what protocols are out there, since DTLS for BFCP is only
mentioned in the new RFC 4583 bis drafts. RFC 4583 bis draft would actually
benefit from a document that describes DTLS offer/answer setup, which it
can reference instead of DTLS-SRTP. All the new work, such as SCTP, can
reference this new document as well. I would guess this will need to go to
MMUSIC.

--20cf304352c87bc357050fc9ac3c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Feb 23, 2015 at 3:37 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holm=
berg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color=
:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span class=3D"=
">&gt; I would like to make sure that the specification explicitly prohibit=
s this and states<br>
&gt; somewhere that fingerprint and setup role MUST NOT change unless the t=
ransport<br>
&gt; parameters change as well.<br>
<br>
</span>As far as SCTP-SDP is concerned, personally I don&#39;t (at least no=
t at this moment) have a problem with such restriction.<br>
<br>
However, when BUNDLE is used, SRTP and SCTP may share a single DTLS connect=
ion. It could cause problems if SRTP and SCTP define different rules when a=
 DTLS connection can be re-established. And, SRTP-DTLS does not forbid that=
 fingerprint and setup role change, unless the transport parameters also ch=
ange.<br>
<span class=3D""><br>
&gt;Unless this is prohibited, some sort of specification language needs to=
 be provided stating how<br>
&gt;multiple DTLS connections can run over a single 5-tuple. This is not cu=
rrently allowed or<br>
&gt;defined anywhere and it is required for DTLS to operate with offer/answ=
er end transport connection reuse.<br>
<br>
</span>Doesn&#39;t the DTLS spec say that, when a new DTLS connection has b=
een established, you need to maintain the keys for the previous connection =
for a while, in order to handle late packets?<br></blockquote><div><br></di=
v><div>This is specific to DTLS re-key (not a complete new session establis=
hment) in combination with SRTP. In case of DTLS re-key with DTLS data fram=
es the epoch number can be used to reference the correct cipher settings. T=
his would be used for SCTP or BFCP which run on top of DTLS data.</div><div=
><br></div><div>Unlike re-key, creating a new session over the same transpo=
rt at the same time as the existing session continues to run is not defined=
 for DTLS.</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204=
,204);border-left-style:solid;padding-left:1ex"><span class=3D"">&gt;As I h=
ave stated on several occasions, I do not like the option of allowing multi=
ple DTLS connections<br>
&gt;over the same transport since it complicates implementations for no rea=
son. Unless someone provides<br>
&gt;a compelling use case why this is needed I strongly believe this should=
 not be allowed.<br>
<br>
</span>I don&#39;t have a use-case, but because of BUNDLE I&#39;d like to h=
ave the same rules for SRTP and SCTP.<br></blockquote><div><br></div><div>I=
 agree that common rules for DTLS offer/answer would be beneficial for ever=
ybody. It is currently either undefined or broken in a similar manner for b=
oth DTLS-SRTP and SCTP draft. New BFCP drafts actually reference DTLS-SRTP =
as far as DTLS session setup is concerned, so they will suffer from the sam=
e problem. Furthermore, =C2=A0I do not think anybody has any intent to impl=
ement DTLS changes without transport changes for any protocol. I think it w=
ould be better to define this correctly at some point vs to keep repeating =
the same uncertainty.</div><div>=C2=A0</div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-colo=
r:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Of course, we =
could add your suggested restriction also to SRTP...<br>
<br>
But, would we then need add the restriction also to other protocols using D=
TLS? E.g. BFCP.<br>
<br>
The best thing would be to have generic Offer/Answer rules for DTLS, which =
all usages (SRTP, SCTP, BFCP etc) then reference, but it&#39;s too late for=
 that...<br></blockquote><div><br></div><div>It might not be a bad idea to =
create such a document and use it to update DTLS-SRTP and other protocols w=
hich use DTLS with offer/answer. I am actually not sure what protocols are =
out there, since DTLS for BFCP is only mentioned in the new RFC 4583 bis dr=
afts.=C2=A0RFC 4583 bis draft would actually benefit from a document=C2=A0t=
hat describes DTLS offer/answer setup, which it can reference instead of DT=
LS-SRTP. All the new work, such as SCTP, can reference this new document as=
 well. I would guess this will need to go to MMUSIC.</div></div></div></div=
>

--20cf304352c87bc357050fc9ac3c--


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Date: Tue, 24 Feb 2015 10:30:52 -0500
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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--001a11343b92431840050fd73463
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On Tue, Feb 24, 2015 at 10:20 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  >This is specific to DTLS re-key (not a complete new session
> establishment) in combination with SRTP. In case of >DTLS re-key with DTLS
> data frames the epoch number can be used to reference the correct cipher
> settings. This >would be used for SCTP or BFCP which run on top of DTLS
> data.
>
>   >
>
> >Unlike re-key, creating a new session over the same transport at the same
> time as the existing session continues to run >is not defined for DTLS.
>
>
>
> Ok. Would it be useful to mention that re-keying can be done without
> re-negotiation the underlying transport? Or, is it obvious?
>
>
>
> Because, I assume we still want to allow re-keying, or?
>

I think re-key is something that has a real value for long duration
connections and should be allowed. Furthermore, DTLS re-key can be done at
any time at the DTLS session layer without the need for the offer/answer
exchange. Re-key is an integral part of DTLS protocol and it is well
defined for both DTLS data (used by SCTP and BFCP) and DTLS-SRTP. Since it
is an integral part of DTLS protocol, I would assume we do not need to
explicitly state that it is allowed.

_____________
Roman Shpount

--001a11343b92431840050fd73463
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
ue, Feb 24, 2015 at 10:20 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.hol=
mberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-colo=
r:rgb(204,204,204);border-left-style:solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">&gt;This is specific to DTLS re-key (not a complete =
new session establishment) in combination with SRTP. In case of &gt;DTLS re=
-key with DTLS data frames the epoch number can be used to reference the co=
rrect cipher settings. This &gt;would be used
 for SCTP or BFCP which run on top of DTLS data.<br></p><div><div><div><spa=
n class=3D""><div><p class=3D"MsoNormal"><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&gt;<u></u>=C2=A0<u></u></p>
</div>
</span><div><span class=3D"">
<p class=3D"MsoNormal">&gt;Unlike re-key, creating a new session over the s=
ame transport at the same time as the existing session continues to run &gt=
;is not defined for DTLS.<u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif"><u></u>=C2=A0<u></u></span></p>
</span><p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Cal=
ibri,sans-serif">Ok. Would it be useful to mention that re-keying can be do=
ne without re-negotiation the underlying transport? Or, is it obvious?<u></=
u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif">Because, I assume we still want to allow re-keying, or?</span></p=
></div></div></div></div></div></div></blockquote><div><br></div><div>I thi=
nk re-key is something that has a real value for long duration connections =
and should be allowed. Furthermore, DTLS re-key can be done at any time at =
the DTLS session layer without the need for the offer/answer exchange. Re-k=
ey is an integral part of DTLS protocol and it is well defined for both DTL=
S data (used by SCTP and BFCP) and DTLS-SRTP. Since it is an integral part =
of DTLS protocol, I would assume we do not need to explicitly state that it=
 is allowed.</div><div><br></div><div><div><div class=3D"gmail_signature">_=
____________<br>Roman Shpount</div></div></div><div><br></div></div></div><=
/div>

--001a11343b92431840050fd73463--


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Date: Tue, 24 Feb 2015 10:34:44 -0500
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From: Simon Perreault <sperreault@jive.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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On Tue, Feb 24, 2015 at 10:20 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> >As I have stated on several occasions, I do not like the option of
> allowing multiple DTLS connections
> >over the same transport since it complicates implementations for no
> reason. Unless someone provides
> >a compelling use case why this is needed I strongly believe this should
> not be allowed.
>
> I don't have a use-case, but because of BUNDLE I'd like to have the same
> rules for SRTP and SCTP.
>
>
>
> >I agree that common rules for DTLS offer/answer would be beneficial for
> everybody. It is currently either undefined >or broken in a similar manner
> for both DTLS-SRTP and SCTP draft. New BFCP drafts actually reference
> DTLS-SRTP >as far as DTLS session setup is concerned, so they will suffer
> from the same problem. Furthermore,  I do not think >anybody has any intent
> to implement DTLS changes without transport changes for any protocol. I
> think it would be >better to define this correctly at some point vs to keep
> repeating the same uncertainty.
>
>
>
> So, would anyone **object** to such restriction? I have seen no
> objections so far.
>

Are you referring specifically to *simultaneous* DTLS connections over the
same transport? I wouldn't object to prohibiting those, but objecting would
be pointless anyway because it's just impossible to have multiple
simultaneous DTLS connections over the same transport. It's like saying
"packets MUST NOT travel faster than the speed of light."

If you're instead referring to multiple *non-simultaneous* DTLS connections
over the same transport, then yes I would object. There's no valid reason
to prohibit those.

Simon

--001a11c3645014ccdb050fd742d6
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Feb 24, 2015 at 10:20 AM, Christer Holmberg <span dir=3D"ltr">&lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer=
.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><span class=3D""><blockquote style=3D"border:none;border-left:solid #cc=
cccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm"><p=
 class=3D"MsoNormal">&gt;As I have stated on several occasions, I do not li=
ke the option of allowing multiple DTLS connections<br>
&gt;over the same transport since it complicates implementations for no rea=
son. Unless someone provides<br>
&gt;a compelling use case why this is needed I strongly believe this should=
 not be allowed.<br>
<br>
I don&#39;t have a use-case, but because of BUNDLE I&#39;d like to have the=
 same rules for SRTP and SCTP.<u></u><u></u></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&gt;I agree that common rules for DTLS offer/answer =
would be beneficial for everybody. It is currently either undefined &gt;or =
broken in a similar manner for both DTLS-SRTP and SCTP draft. New BFCP draf=
ts actually reference DTLS-SRTP &gt;as far
 as DTLS session setup is concerned, so they will suffer from the same prob=
lem. Furthermore, =C2=A0I do not think &gt;anybody has any intent to implem=
ent DTLS changes without transport changes for any protocol. I think it wou=
ld be &gt;better to define this correctly at
 some point vs to keep repeating the same uncertainty.<u></u><u></u></p>
</div>
</span><div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif">So, would anyone *<b>object</b>* to such restrictio=
n? I have seen no objections so far.</span></p></div></blockquote></div><br=
>Are you referring specifically to *simultaneous* DTLS connections over the=
 same transport? I wouldn&#39;t object to prohibiting those, but objecting =
would be pointless anyway because it&#39;s just impossible to have multiple=
 simultaneous DTLS connections over the same transport. It&#39;s like sayin=
g &quot;packets MUST NOT travel faster than the speed of light.&quot;</div>=
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">If you&#39;=
re instead referring to multiple *non-simultaneous* DTLS connections over t=
he same transport, then yes I would object. There&#39;s no valid reason to =
prohibit those.</div><div class=3D"gmail_extra"><br></div><div class=3D"gma=
il_extra">Simon</div></div>

--001a11c3645014ccdb050fd742d6--


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Date: Tue, 24 Feb 2015 10:36:34 -0500
Message-ID: <CANO7kWAZofUabxYxiuFZ28+UWTwNtYYXrh1arCRMCgXKaOTdow@mail.gmail.com>
From: Simon Perreault <sperreault@jive.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Offer/Answer and DTLS
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--047d7b3a8192a788c6050fd748fe
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On Tue, Feb 24, 2015 at 10:34 AM, Simon Perreault <sperreault@jive.com>
wrote:

> On Tue, Feb 24, 2015 at 10:20 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> >As I have stated on several occasions, I do not like the option of
>> allowing multiple DTLS connections
>> >over the same transport since it complicates implementations for no
>> reason. Unless someone provides
>> >a compelling use case why this is needed I strongly believe this should
>> not be allowed.
>>
>> I don't have a use-case, but because of BUNDLE I'd like to have the same
>> rules for SRTP and SCTP.
>>
>>
>>
>> >I agree that common rules for DTLS offer/answer would be beneficial for
>> everybody. It is currently either undefined >or broken in a similar manner
>> for both DTLS-SRTP and SCTP draft. New BFCP drafts actually reference
>> DTLS-SRTP >as far as DTLS session setup is concerned, so they will suffer
>> from the same problem. Furthermore,  I do not think >anybody has any intent
>> to implement DTLS changes without transport changes for any protocol. I
>> think it would be >better to define this correctly at some point vs to keep
>> repeating the same uncertainty.
>>
>>
>>
>> So, would anyone **object** to such restriction? I have seen no
>> objections so far.
>>
>
> Are you referring specifically to *simultaneous* DTLS connections over the
> same transport? I wouldn't object to prohibiting those, but objecting would
> be pointless anyway because it's just impossible to have multiple
> simultaneous DTLS connections over the same transport. It's like saying
> "packets MUST NOT travel faster than the speed of light."
>
> If you're instead referring to multiple *non-simultaneous* DTLS
> connections over the same transport, then yes I would object. There's no
> valid reason to prohibit those.
>

Oh, and by "transport" I'm assuming you mean "5-tuple". If that's not
correct then... I object, your honour! ;)

Simon

--047d7b3a8192a788c6050fd748fe
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Feb 24, 2015 at 10:34 AM, Simon Perreault <span dir=3D"ltr">&lt;<a =
href=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</=
a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=3D"gmail_=
extra"><span class=3D""><div class=3D"gmail_quote">On Tue, Feb 24, 2015 at =
10:20 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christe=
r.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com</=
a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span><blockquote st=
yle=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0p=
t;margin-left:4.8pt;margin-right:0cm"><p class=3D"MsoNormal">&gt;As I have =
stated on several occasions, I do not like the option of allowing multiple =
DTLS connections<br>
&gt;over the same transport since it complicates implementations for no rea=
son. Unless someone provides<br>
&gt;a compelling use case why this is needed I strongly believe this should=
 not be allowed.<br>
<br>
I don&#39;t have a use-case, but because of BUNDLE I&#39;d like to have the=
 same rules for SRTP and SCTP.<u></u><u></u></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&gt;I agree that common rules for DTLS offer/answer =
would be beneficial for everybody. It is currently either undefined &gt;or =
broken in a similar manner for both DTLS-SRTP and SCTP draft. New BFCP draf=
ts actually reference DTLS-SRTP &gt;as far
 as DTLS session setup is concerned, so they will suffer from the same prob=
lem. Furthermore, =C2=A0I do not think &gt;anybody has any intent to implem=
ent DTLS changes without transport changes for any protocol. I think it wou=
ld be &gt;better to define this correctly at
 some point vs to keep repeating the same uncertainty.<u></u><u></u></p>
</div>
</span><div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif">So, would anyone *<b>object</b>* to such restrictio=
n? I have seen no objections so far.</span></p></div></blockquote></div><br=
></span>Are you referring specifically to *simultaneous* DTLS connections o=
ver the same transport? I wouldn&#39;t object to prohibiting those, but obj=
ecting would be pointless anyway because it&#39;s just impossible to have m=
ultiple simultaneous DTLS connections over the same transport. It&#39;s lik=
e saying &quot;packets MUST NOT travel faster than the speed of light.&quot=
;</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">If y=
ou&#39;re instead referring to multiple *non-simultaneous* DTLS connections=
 over the same transport, then yes I would object. There&#39;s no valid rea=
son to prohibit those.</div></blockquote></div><br>Oh, and by &quot;transpo=
rt&quot; I&#39;m assuming you mean &quot;5-tuple&quot;. If that&#39;s not c=
orrect then... I object, your honour! ;)</div><div class=3D"gmail_extra"><b=
r></div><div class=3D"gmail_extra">Simon</div></div>

--047d7b3a8192a788c6050fd748fe--


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Date: Tue, 24 Feb 2015 10:44:42 -0500
Message-ID: <CAD5OKxs_FD+ojWVBZ7nAFTusNtJY7PzXHFpBa8c3D65ZpfH80w@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Archived-At: <http://mailarchive.ietf.org/arch/msg/rtcweb/WjVZ0ejqMXOZh6lu-T1VOmK0MpU>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Offer/Answer and DTLS
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Content-Type: text/plain; charset=UTF-8

On Tue, Feb 24, 2015 at 10:36 AM, Simon Perreault <sperreault@jive.com>
wrote:

> Are you referring specifically to *simultaneous* DTLS connections over the
>> same transport? I wouldn't object to prohibiting those, but objecting would
>> be pointless anyway because it's just impossible to have multiple
>> simultaneous DTLS connections over the same transport. It's like saying
>> "packets MUST NOT travel faster than the speed of light."
>>
>> If you're instead referring to multiple *non-simultaneous* DTLS
>> connections over the same transport, then yes I would object. There's no
>> valid reason to prohibit those.
>>
>
> Oh, and by "transport" I'm assuming you mean "5-tuple". If that's not
> correct then... I object, your honour! ;)
>
>
So, do you want to allow setting up new DTLS session over the same
underlying transport ("5-tuple" or ICE connection)? If you do, can explain
how can you ensure that only one DTLS connection would be running over this
transport at any given time during the offer/answer re-negotiation? For the
description of the problems please see my original email that started this
thread. By very nature of offer/answer multiple simultaneous DTLS sessions
must be supported during the re-negotiation, and as you said yourself,
those are not possible.

Also, what is the use case for which you would need to setup new DTLS
session when underlying transport stays the same?
_____________
Roman Shpount

--20cf304352c8b817ca050fd765a1
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
ue, Feb 24, 2015 at 10:36 AM, Simon Perreault <span dir=3D"ltr">&lt;<a href=
=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_e=
xtra"><div><div class=3D"h5"><div class=3D"gmail_quote"><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">=
<div class=3D"gmail_extra">Are you referring specifically to *simultaneous*=
 DTLS connections over the same transport? I wouldn&#39;t object to prohibi=
ting those, but objecting would be pointless anyway because it&#39;s just i=
mpossible to have multiple simultaneous DTLS connections over the same tran=
sport. It&#39;s like saying &quot;packets MUST NOT travel faster than the s=
peed of light.&quot;</div><div class=3D"gmail_extra"><br></div><div class=
=3D"gmail_extra">If you&#39;re instead referring to multiple *non-simultane=
ous* DTLS connections over the same transport, then yes I would object. The=
re&#39;s no valid reason to prohibit those.</div></blockquote></div><br></d=
iv></div>Oh, and by &quot;transport&quot; I&#39;m assuming you mean &quot;5=
-tuple&quot;. If that&#39;s not correct then... I object, your honour! ;)</=
div><span class=3D""><font color=3D"#888888"><div class=3D"gmail_extra"><br=
></div></font></span></div></blockquote><div><br></div><div>So, do you want=
 to allow setting up new DTLS session over the same underlying transport (&=
quot;5-tuple&quot; or ICE connection)? If you do, can explain how can you e=
nsure that only one DTLS connection would be running over this transport at=
 any given time during the offer/answer re-negotiation? For the description=
 of the problems please see my original email that started this thread. By =
very nature of offer/answer multiple simultaneous DTLS sessions must be sup=
ported during the re-negotiation, and as you said yourself, those are not p=
ossible.=C2=A0</div><div><br></div><div>Also, what is the use case for whic=
h you would need to setup new DTLS session when underlying transport stays =
the same?</div><div><div class=3D"gmail_signature">_____________<br>Roman S=
hpount</div></div><div>=C2=A0</div></div></div></div>

--20cf304352c8b817ca050fd765a1--


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Date: Tue, 24 Feb 2015 10:57:40 -0500
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From: Simon Perreault <sperreault@jive.com>
To: Roman Shpount <roman@telurix.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Offer/Answer and DTLS
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--089e0160a9d21a37cc050fd794e7
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On Tue, Feb 24, 2015 at 10:44 AM, Roman Shpount <roman@telurix.com> wrote:
>
> So, do you want to allow setting up new DTLS session over the same
> underlying transport ("5-tuple" or ICE connection)?
>

I don't, but that's besides the point. The point is that when there's no
reason to prohibit doing something, then we must not prohibit it.


> If you do, can explain how can you ensure that only one DTLS connection
> would be running over this transport at any given time during the
> offer/answer re-negotiation? For the description of the problems please see
> my original email that started this thread. By very nature of offer/answer
> multiple simultaneous DTLS sessions must be supported during the
> re-negotiation, and as you said yourself, those are not possible.
>

This is not specific to DTLS. All UDP-based protocols are affected by this.
When you want to set-up a new session over the same transport, either you
implement something like TCP's TIME_WAIT state, or you define a way to
unambiguously differentiate sessions. Various protocols do it in various
ways. For DTLS, packets from to the old session would be automatically
rejected. This can be acceptable depending on the use case.

Simon

--089e0160a9d21a37cc050fd794e7
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Feb 24, 2015 at 10:44 AM, Roman Shpount <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;=
</span> wrote:<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div>So, do you want to allow =
setting up new DTLS session over the same underlying transport (&quot;5-tup=
le&quot; or ICE connection)?</div></blockquote><div><br></div><div>I don&#3=
9;t, but that&#39;s besides the point. The point is that when there&#39;s n=
o reason to prohibit doing something, then we must not prohibit it.</div><d=
iv>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex"><div> If you do, can explain =
how can you ensure that only one DTLS connection would be running over this=
 transport at any given time during the offer/answer re-negotiation? For th=
e description of the problems please see my original email that started thi=
s thread. By very nature of offer/answer multiple simultaneous DTLS session=
s must be supported during the re-negotiation, and as you said yourself, th=
ose are not possible.=C2=A0</div></blockquote><div><br></div><div>This is n=
ot specific to DTLS. All UDP-based protocols are affected by this. When you=
 want to set-up a new session over the same transport, either you implement=
 something like TCP&#39;s TIME_WAIT state, or you define a way to unambiguo=
usly differentiate sessions. Various protocols do it in various ways. For D=
TLS, packets from to the old session would be automatically rejected. This =
can be acceptable depending on the use case.</div><div><br></div><div>Simon=
</div></div></div></div>

--089e0160a9d21a37cc050fd794e7--


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From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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--001a11c396ce48546e050fd7bcf3
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On Tue, Feb 24, 2015 at 10:57 AM, Simon Perreault <sperreault@jive.com>
wrote:

>
> On Tue, Feb 24, 2015 at 10:44 AM, Roman Shpount <roman@telurix.com> wrote:
>>
>> So, do you want to allow setting up new DTLS session over the same
>> underlying transport ("5-tuple" or ICE connection)?
>>
>
> I don't, but that's besides the point. The point is that when there's no
> reason to prohibit doing something, then we must not prohibit it.
>

At this point none of the webrtc implementations plan to support this. Why
specify something that no one will support?


> This is not specific to DTLS. All UDP-based protocols are affected by
> this. When you want to set-up a new session over the same transport, either
> you implement something like TCP's TIME_WAIT state, or you define a way to
> unambiguously differentiate sessions. Various protocols do it in various
> ways. For DTLS, packets from to the old session would be automatically
> rejected. This can be acceptable depending on the use case.
>
>

Can you please read the problem description from my original email and
reply to it? I think you are missing the point that I am raising. This is
not about late packets, even though late packets are also somewhat of an
issue. This is about the fact that the new session would need to be
negotiated at the same time as the old session is running. Furthermore,
this can very often happen when an end point is not expecting it.

_____________
Roman Shpount

--001a11c396ce48546e050fd7bcf3
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Feb 24, 2015 at 10:57 AM, Simon Perreault <span dir=3D"ltr">&l=
t;<a href=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.=
com</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote"><span class=3D"">On Tue, Feb 24, 2015 at 10:44 AM, Roman Shpount =
<span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank=
">roman@telurix.com</a>&gt;</span> wrote:<blockquote class=3D"gmail_quote" =
style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:r=
gb(204,204,204);border-left-style:solid;padding-left:1ex"><div>So, do you w=
ant to allow setting up new DTLS session over the same underlying transport=
 (&quot;5-tuple&quot; or ICE connection)?</div></blockquote><div><br></div>=
</span><div>I don&#39;t, but that&#39;s besides the point. The point is tha=
t when there&#39;s no reason to prohibit doing something, then we must not =
prohibit it.</div></div></div></div></blockquote><div><br></div><div>At thi=
s point none of the webrtc implementations plan to support this. Why specif=
y something that no one will support?</div><div>=C2=A0</div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;=
border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex=
"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><s=
pan class=3D""><div>This is not specific to DTLS. All UDP-based protocols a=
re affected by this. When you want to set-up a new session over the same tr=
ansport, either you implement something like TCP&#39;s TIME_WAIT state, or =
you define a way to unambiguously differentiate sessions. Various protocols=
 do it in various ways. For DTLS, packets from to the old session would be =
automatically rejected. This can be acceptable depending on the use case.<b=
r></div></span><span class=3D""><font color=3D"#888888"><div><br></div><div=
></div></font></span></div></div></div></blockquote></div><br></div><div cl=
ass=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Can you please rea=
d the problem description from my original email and reply to it? I think y=
ou are missing the point that I am raising. This is not about late packets,=
 even though late packets are also somewhat of an issue. This is about the =
fact that the new session would need to be negotiated at the same time as t=
he old session is running. Furthermore, this can very often happen when an =
end point is not expecting it.<br></div><div class=3D"gmail_extra"><br></di=
v><div class=3D"gmail_extra"><div><div class=3D"gmail_signature">__________=
___<br>Roman Shpount</div></div><div><br></div></div></div>

--001a11c396ce48546e050fd7bcf3--


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From: Simon Perreault <sperreault@jive.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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On Tue, Feb 24, 2015 at 11:08 AM, Roman Shpount <roman@telurix.com> wrote:

> On Tue, Feb 24, 2015 at 10:57 AM, Simon Perreault <sperreault@jive.com>
> wrote:
>
>>
>> On Tue, Feb 24, 2015 at 10:44 AM, Roman Shpount <roman@telurix.com>
>> wrote:
>>>
>>> So, do you want to allow setting up new DTLS session over the same
>>> underlying transport ("5-tuple" or ICE connection)?
>>>
>>
>> I don't, but that's besides the point. The point is that when there's no
>> reason to prohibit doing something, then we must not prohibit it.
>>
>
> At this point none of the webrtc implementations plan to support this. Why
> specify something that no one will support?
>

That's exactly what I'm saying! Let's not "specify" it. Let's just say
nothing at all about this topic.

This is not specific to DTLS. All UDP-based protocols are affected by this.
>> When you want to set-up a new session over the same transport, either you
>> implement something like TCP's TIME_WAIT state, or you define a way to
>> unambiguously differentiate sessions. Various protocols do it in various
>> ways. For DTLS, packets from to the old session would be automatically
>> rejected. This can be acceptable depending on the use case.
>>
>
> Can you please read the problem description from my original email and
> reply to it? I think you are missing the point that I am raising. This is
> not about late packets, even though late packets are also somewhat of an
> issue. This is about the fact that the new session would need to be
> negotiated at the same time as the old session is running. Furthermore,
> this can very often happen when an end point is not expecting it.
>

Look, I think we're agreeing on maybe 95% of the things. I think what we
need at this point is a verbatim text proposal. Then we can wordsmith it as
we like.

Simon

--001a113464ac82d117050fd7dad9
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Feb 24, 2015 at 11:08 AM, Roman Shpount <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=3D"gmail_extra"=
><span class=3D""><div><div>On Tue, Feb 24, 2015 at 10:57 AM, Simon Perreau=
lt <span dir=3D"ltr">&lt;<a href=3D"mailto:sperreault@jive.com" target=3D"_=
blank">sperreault@jive.com</a>&gt;</span> wrote:<br></div></div></span><div=
 class=3D"gmail_quote"><span class=3D""><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rg=
b(204,204,204);border-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><span>On Tue, Feb =
24, 2015 at 10:44 AM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto=
:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrot=
e:<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid=
;padding-left:1ex"><div>So, do you want to allow setting up new DTLS sessio=
n over the same underlying transport (&quot;5-tuple&quot; or ICE connection=
)?</div></blockquote><div><br></div></span><div>I don&#39;t, but that&#39;s=
 besides the point. The point is that when there&#39;s no reason to prohibi=
t doing something, then we must not prohibit it.</div></div></div></div></b=
lockquote><div><br></div></span><div>At this point none of the webrtc imple=
mentations plan to support this. Why specify something that no one will sup=
port?</div></div></div></blockquote><div><br></div><div>That&#39;s exactly =
what I&#39;m saying! Let&#39;s not &quot;specify&quot; it. Let&#39;s just s=
ay nothing at all about this topic.</div><div><br></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span c=
lass=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-sty=
le:solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div=
 class=3D"gmail_quote"><span><div>This is not specific to DTLS. All UDP-bas=
ed protocols are affected by this. When you want to set-up a new session ov=
er the same transport, either you implement something like TCP&#39;s TIME_W=
AIT state, or you define a way to unambiguously differentiate sessions. Var=
ious protocols do it in various ways. For DTLS, packets from to the old ses=
sion would be automatically rejected. This can be acceptable depending on t=
he use case.</div></span></div></div></div></blockquote></span></div></div>=
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Can you ple=
ase read the problem description from my original email and reply to it? I =
think you are missing the point that I am raising. This is not about late p=
ackets, even though late packets are also somewhat of an issue. This is abo=
ut the fact that the new session would need to be negotiated at the same ti=
me as the old session is running. Furthermore, this can very often happen w=
hen an end point is not expecting it.</div></blockquote></div><br>Look, I t=
hink we&#39;re agreeing on maybe 95% of the things. I think what we need at=
 this point is a verbatim text proposal. Then we can wordsmith it as we lik=
e.</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Sim=
on</div></div>

--001a113464ac82d117050fd7dad9--


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From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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On Tue, Feb 24, 2015 at 11:17 AM, Simon Perreault <sperreault@jive.com>
wrote:

>
> Look, I think we're agreeing on maybe 95% of the things. I think what we
> need at this point is a verbatim text proposal. Then we can wordsmith it as
> we like.
>
>
My text proposal is: "DTLS fingerprint and setup role MUST NOT change
unless underlying transport did not change as well." You have just said
that you disagree with this.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
ue, Feb 24, 2015 at 11:17 AM, Simon Perreault <span dir=3D"ltr">&lt;<a href=
=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_e=
xtra"><br><div class=3D"gmail_quote">Look, I think we&#39;re agreeing on ma=
ybe 95% of the things. I think what we need at this point is a verbatim tex=
t proposal. Then we can wordsmith it as we like.<br></div></div><span class=
=3D""><font color=3D"#888888"><div class=3D"gmail_extra"><br></div></font><=
/span></div></blockquote><div><br></div><div>My text proposal is: &quot;DTL=
S fingerprint and setup role MUST NOT change unless underlying transport di=
d not change as well.&quot; You have just said that you disagree with this.=
</div><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</d=
iv></div><div>=C2=A0</div></div></div></div>

--bcaec50e5f1dde7e26050fd7eb9c--


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From: Simon Perreault <sperreault@jive.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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On Tue, Feb 24, 2015 at 11:22 AM, Roman Shpount <roman@telurix.com> wrote:

> My text proposal is: "DTLS fingerprint and setup role MUST NOT change
> unless underlying transport did not change as well." You have just said
> that you disagree with this.


I don't disagree, as I said I think I agree with about 95% of it.

One negation needs to be reversed: "DTLS fingerprint and setup role MUST
NOT change unless underlying transport did change as well."

In what draft would this go? Which section? I'm sorry but after re-reading
the whole thread I still can't find this.

Simon

--001a11c36450e7ab1c050fd81fc7
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Feb 24, 2015 at 11:22 AM, Roman Shpount <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-=
left-style:solid;padding-left:1ex">My text proposal is: &quot;DTLS fingerpr=
int and setup role MUST NOT change unless underlying transport did not chan=
ge as well.&quot; You have just said that you disagree with this.</blockquo=
te></div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">I =
don&#39;t disagree, as I said I think I agree with about 95% of it.</div><d=
iv class=3D"gmail_extra"><br></div>One negation needs to be reversed: &quot=
;DTLS fingerprint and setup role MUST NOT change unless underlying transpor=
t did change as well.&quot;</div><div class=3D"gmail_extra"><br></div><div =
class=3D"gmail_extra">In what draft would this go? Which section? I&#39;m s=
orry but after re-reading the whole thread I still can&#39;t find this.</di=
v><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Simon</di=
v></div>

--001a11c36450e7ab1c050fd81fc7--


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From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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--089e01229aaa7917f3050fd86172
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On Tue, Feb 24, 2015 at 11:36 AM, Simon Perreault <sperreault@jive.com>
wrote:

>
> On Tue, Feb 24, 2015 at 11:22 AM, Roman Shpount <roman@telurix.com> wrote:
>
>> My text proposal is: "DTLS fingerprint and setup role MUST NOT change
>> unless underlying transport did not change as well." You have just said
>> that you disagree with this.
>
>
> I don't disagree, as I said I think I agree with about 95% of it.
>
> One negation needs to be reversed: "DTLS fingerprint and setup role MUST
> NOT change unless underlying transport did change as well."
>

This is not I do not write drafts -- I can be dyslexic at times.

In what draft would this go? Which section? I'm sorry but after re-reading
> the whole thread I still can't find this.
>
>
I think this particular discussion started because of
draft-ietf-mmusic-sctp-sdp. The best alternative would be to have a
separate new draft in MMUSIC which will define common offer/answer rules
for DTLS. This draft can update RFC 5763 and it can be referenced by
draft-ietf-mmusic-sctp-sdp and RFC 4583 bis. Another alternative would be to
update DTLS-SRTP (update RFC 5763) and include this language in
draft-ietf-mmusic-sctp-sdp.
_____________
Roman Shpount

--089e01229aaa7917f3050fd86172
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
ue, Feb 24, 2015 at 11:36 AM, Simon Perreault <span dir=3D"ltr">&lt;<a href=
=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_e=
xtra"><span class=3D""><br><div class=3D"gmail_quote">On Tue, Feb 24, 2015 =
at 11:22 AM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@te=
lurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-lef=
t-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padd=
ing-left:1ex">My text proposal is: &quot;DTLS fingerprint and setup role MU=
ST NOT change unless underlying transport did not change as well.&quot; You=
 have just said that you disagree with this.</blockquote></div><div class=
=3D"gmail_extra"><br></div></span><div class=3D"gmail_extra">I don&#39;t di=
sagree, as I said I think I agree with about 95% of it.</div><div class=3D"=
gmail_extra"><br></div>One negation needs to be reversed: &quot;DTLS finger=
print and setup role MUST NOT change unless underlying transport did change=
 as well.&quot;</div></div></blockquote><div><br></div><div>This is not I d=
o not write drafts -- I can be dyslexic at times.</div><div><br></div><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-=
width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;paddin=
g-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">In what draft would=
 this go? Which section? I&#39;m sorry but after re-reading the whole threa=
d I still can&#39;t find this.<br></div><span class=3D""><font color=3D"#88=
8888"><div class=3D"gmail_extra"><br></div></font></span></div></blockquote=
><div><br></div>I think this particular discussion started because of draft=
-ietf-mmusic-sctp-sdp. The best alternative would be to have a separate new=
 draft in MMUSIC which will define common offer/answer rules for DTLS. This=
 draft can update RFC 5763 and it can be referenced by draft-ietf-mmusic-sc=
tp-sdp and=C2=A0<span style=3D"color:rgb(0,0,0);font-size:12.8000001907349p=
x">RFC 4583 bis. Another alternative would be=C2=A0</span>to update DTLS-SR=
TP (update RFC 5763) and include this language in draft-ietf-mmusic-sctp-sd=
p.</div><div class=3D"gmail_quote"><div><div class=3D"gmail_signature">____=
_________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--089e01229aaa7917f3050fd86172--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Simon Perreault <sperreault@jive.com>
Thread-Topic: [rtcweb] Offer/Answer and DTLS
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Subject: Re: [rtcweb] Offer/Answer and DTLS
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--_000_7594FB04B1934943A5C02806D1A2204B1D7172A0ESESSMB209erics_--


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Subject: Re: [rtcweb] Offer/Answer and DTLS
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Den 24. feb. 2015 18:36, skrev Christer Holmberg:
> Hi,
> 
>  
> 
> The actual documentation would be discussed in MMUSIC, and obviously
> they may have opinions on the suggested restriction too. Keep in mind
> that this is not an RTCWEB specific issue.
> 
>  
> 
> But, again, before I move the discussion to MMUSIC Id like to verify
> whether we, from an RTCWEB perspective, are ok with the suggestion.

Speaking as an RTCWEB participant, I'm happy with this resolution.
It says that I can't do something I don't want to do, and don't have to
write code for someone else doing it either.


> 
>  
> 
> Regards,
> 
>  
> 
> Christer
> 
>  
> 
> *From:*Roman Shpount [mailto:roman@telurix.com]
> *Sent:* 24 February 2015 18:55
> *To:* Simon Perreault
> *Cc:* Christer Holmberg; rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Offer/Answer and DTLS
> 
>  
> 
> On Tue, Feb 24, 2015 at 11:36 AM, Simon Perreault <sperreault@jive.com
> <mailto:sperreault@jive.com>> wrote:
> 
>      
> 
>     On Tue, Feb 24, 2015 at 11:22 AM, Roman Shpount <roman@telurix.com
>     <mailto:roman@telurix.com>> wrote:
> 
>         My text proposal is: "DTLS fingerprint and setup role MUST NOT
>         change unless underlying transport did not change as well." You
>         have just said that you disagree with this.
> 
>      
> 
>     I don't disagree, as I said I think I agree with about 95% of it.
> 
>      
> 
>     One negation needs to be reversed: "DTLS fingerprint and setup role
>     MUST NOT change unless underlying transport did change as well."
> 
>  
> 
> This is not I do not write drafts -- I can be dyslexic at times.
> 
>  
> 
>     In what draft would this go? Which section? I'm sorry but after
>     re-reading the whole thread I still can't find this.
> 
>      
> 
>  
> 
> I think this particular discussion started because of
> draft-ietf-mmusic-sctp-sdp. The best alternative would be to have a
> separate new draft in MMUSIC which will define common offer/answer rules
> for DTLS. This draft can update RFC 5763 and it can be referenced by
> draft-ietf-mmusic-sctp-sdp and RFC 4583 bis. Another alternative would
> be to update DTLS-SRTP (update RFC 5763) and include this language in
> draft-ietf-mmusic-sctp-sdp.
> 
> _____________
> Roman Shpount
> 
>  
> 
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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Subject: [rtcweb] JSEP: Question relating to ICE candidate pool
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--089e01493f60008be7050fd9da63
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A question has arisen with respect to draft-ietf-rtcweb-jsep-08 Sections
3.4.4 and 4.1.2.  Exerpts from these sections are enclosed below.

While Section 3.4.4 refers to "pre-gathering" of ICE candidates prior to
calling of setLocalDescription(), Section 4.1.2 states that createOffer()
does not result in candidate gathering.

One way to interpret this is that while createOffer() does not result in
firing of the onicecandidate callback (which needs to wait until
setLocalDescription() is called), it is ok (or even encouraged) for the
implementation to begin "pre-gathering" ICE candidates into the ICE
candidate pool once createOffer() is called.

Is this a reasonable interpretation or is there another way this
could/should be read?


Section 4.1.2 createOffer:

   Calling this method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.


Section 3.4.4 ICE candidate pool:


   JSEP applications typically inform the browser to begin ICE gathering
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams, and thereby ICE
   components, for which to gather candidates.  However, to accelerate
   cases where the application knows the number of ICE components to use
   ahead of time, it may ask the browser to gather a pool of potential
   ICE candidates to help ensure rapid media setup.

   When setLocalDescription is eventually called, and the browser goes
   to gather the needed ICE candidates, it SHOULD start by checking if
   any candidates are available in the pool.  If there are candidates in
   the pool, they SHOULD be handed to the application immediately via
   the ICE candidate callback.  If the pool becomes depleted, either
   because a larger-than-expected number of ICE components is used, or
   because the pool has not had enough time to gather candidates, the
   remaining candidates are gathered as usual.

   One example of where this concept is useful is an application that
   expects an incoming call at some point in the future, and wants to
   minimize the time it takes to establish connectivity, to avoid
   clipping of initial media.  By pre-gathering candidates into the
   pool, it can exchange and start sending connectivity checks from
   these candidates almost immediately upon receipt of a call.  Note
   though that by holding on to these pre-gathered candidates, which
   will be kept alive as long as they may be needed, the application
   will consume resources on the STUN/TURN servers it is using.

--089e01493f60008be7050fd9da63
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">A question has arisen with respect to draft-ietf-rtcweb-js=
ep-08 Sections 3.4.4 and 4.1.2.=C2=A0 Exerpts from these sections are enclo=
sed below.=C2=A0<div><br></div><div>While Section 3.4.4 refers to &quot;pre=
-gathering&quot; of ICE candidates prior to calling of setLocalDescription(=
), Section 4.1.2 states that createOffer() does not result in candidate gat=
hering.=C2=A0</div><div><br></div><div>One way to interpret this is that wh=
ile createOffer() does not result in firing of the onicecandidate callback =
(which needs to wait until setLocalDescription() is called), it is ok (or e=
ven encouraged) for the implementation to begin &quot;pre-gathering&quot; I=
CE candidates into the ICE candidate pool once createOffer() is called.=C2=
=A0</div><div><br></div><div>Is this a reasonable interpretation or is ther=
e another way this could/should be read? =C2=A0<br><div><br></div><div><br>=
</div><div>Section 4.1.2 createOffer:=C2=A0<div><br></div><div><pre class=
=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0=
,0)">   Calling this method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.</pre><pre class=3D"" style=3D"font-size:=
1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=
=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0=
,0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-s=
ize:small;white-space:normal">Section 3.4.4 ICE candidate pool:=C2=A0</span=
><br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bot=
tom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:ar=
ial,sans-serif;font-size:small;white-space:normal"><br></span></pre><pre cl=
ass=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(=
0,0,0)"><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom=
:0px">   JSEP applications typically inform the browser to begin ICE gather=
ing
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams, and thereby ICE
   components, for which to gather candidates.  However, to accelerate
   cases where the application knows the number of ICE components to use
   ahead of time, it may ask the browser to gather a pool of potential
   ICE candidates to help ensure rapid media setup.

   When setLocalDescription is eventually called, and the browser goes
   to gather the needed ICE candidates, it SHOULD start by checking if
   any candidates are available in the pool.  If there are candidates in
   the pool, they SHOULD be handed to the application immediately via
   the ICE candidate callback.  If the pool becomes depleted, either
   because a larger-than-expected number of ICE components is used, or
   because the pool has not had enough time to gather candidates, the
   remaining candidates are gathered as usual.

   One example of where this concept is useful is an application that
   expects an incoming call at some point in the future, and wants to
   minimize the time it takes to establish connectivity, to avoid
   clipping of initial media.  By pre-gathering candidates into the
   pool, it can exchange and start sending connectivity checks from
   these candidates almost immediately upon receipt of a call.  Note
   though that by holding on to these pre-gathered candidates, which
   will be kept alive as long as they may be needed, the application
   will consume resources on the STUN/TURN servers it is using.</pre></pre>=
<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-se=
rif;font-size:small;white-space:normal"><br></span></pre><pre class=3D"" st=
yle=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><sp=
an style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:smal=
l;white-space:normal"><br></span></pre></div></div></div></div>

--089e01493f60008be7050fd9da63--


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Date: Wed, 25 Feb 2015 09:44:55 +1300
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Subject: Re: [rtcweb] JSEP: Question relating to ICE candidate pool
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--089e013a2b2c5dd8d5050fdb97cb
Content-Type: text/plain; charset=UTF-8

I recall that the gathering could start at any point. At one point, Firefox
was gathering immediately after construction. I think that the intent was
to have pre-gathering start when the setting was applied.
On Feb 25, 2015 7:40 AM, "Bernard Aboba" <bernard.aboba@gmail.com> wrote:

> A question has arisen with respect to draft-ietf-rtcweb-jsep-08 Sections
> 3.4.4 and 4.1.2.  Exerpts from these sections are enclosed below.
>
> While Section 3.4.4 refers to "pre-gathering" of ICE candidates prior to
> calling of setLocalDescription(), Section 4.1.2 states that createOffer()
> does not result in candidate gathering.
>
> One way to interpret this is that while createOffer() does not result in
> firing of the onicecandidate callback (which needs to wait until
> setLocalDescription() is called), it is ok (or even encouraged) for the
> implementation to begin "pre-gathering" ICE candidates into the ICE
> candidate pool once createOffer() is called.
>
> Is this a reasonable interpretation or is there another way this
> could/should be read?
>
>
> Section 4.1.2 createOffer:
>
>    Calling this method may do things such as generate new ICE
>    credentials, but does not result in candidate gathering, or cause
>    media to start or stop flowing.
>
>
> Section 3.4.4 ICE candidate pool:
>
>
>    JSEP applications typically inform the browser to begin ICE gathering
>    via the information supplied to setLocalDescription, as this is where
>    the app specifies the number of media streams, and thereby ICE
>    components, for which to gather candidates.  However, to accelerate
>    cases where the application knows the number of ICE components to use
>    ahead of time, it may ask the browser to gather a pool of potential
>    ICE candidates to help ensure rapid media setup.
>
>    When setLocalDescription is eventually called, and the browser goes
>    to gather the needed ICE candidates, it SHOULD start by checking if
>    any candidates are available in the pool.  If there are candidates in
>    the pool, they SHOULD be handed to the application immediately via
>    the ICE candidate callback.  If the pool becomes depleted, either
>    because a larger-than-expected number of ICE components is used, or
>    because the pool has not had enough time to gather candidates, the
>    remaining candidates are gathered as usual.
>
>    One example of where this concept is useful is an application that
>    expects an incoming call at some point in the future, and wants to
>    minimize the time it takes to establish connectivity, to avoid
>    clipping of initial media.  By pre-gathering candidates into the
>    pool, it can exchange and start sending connectivity checks from
>    these candidates almost immediately upon receipt of a call.  Note
>    though that by holding on to these pre-gathered candidates, which
>    will be kept alive as long as they may be needed, the application
>    will consume resources on the STUN/TURN servers it is using.
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e013a2b2c5dd8d5050fdb97cb
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">I recall that the gathering could start at any point. At one=
 point, Firefox was gathering immediately after construction. I think that =
the intent was to have pre-gathering start when the setting was applied.</p=
>
<div class=3D"gmail_quote">On Feb 25, 2015 7:40 AM, &quot;Bernard Aboba&quo=
t; &lt;<a href=3D"mailto:bernard.aboba@gmail.com">bernard.aboba@gmail.com</=
a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div d=
ir=3D"ltr">A question has arisen with respect to draft-ietf-rtcweb-jsep-08 =
Sections 3.4.4 and 4.1.2.=C2=A0 Exerpts from these sections are enclosed be=
low.=C2=A0<div><br></div><div>While Section 3.4.4 refers to &quot;pre-gathe=
ring&quot; of ICE candidates prior to calling of setLocalDescription(), Sec=
tion 4.1.2 states that createOffer() does not result in candidate gathering=
.=C2=A0</div><div><br></div><div>One way to interpret this is that while cr=
eateOffer() does not result in firing of the onicecandidate callback (which=
 needs to wait until setLocalDescription() is called), it is ok (or even en=
couraged) for the implementation to begin &quot;pre-gathering&quot; ICE can=
didates into the ICE candidate pool once createOffer() is called.=C2=A0</di=
v><div><br></div><div>Is this a reasonable interpretation or is there anoth=
er way this could/should be read? =C2=A0<br><div><br></div><div><br></div><=
div>Section 4.1.2 createOffer:=C2=A0<div><br></div><div><pre style=3D"font-=
size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   Calling this=
 method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.</pre><pre style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre style=3D"font-si=
ze:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"co=
lor:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;white-space:=
normal">Section 3.4.4 ICE candidate pool:=C2=A0</span><br></pre><pre style=
=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;w=
hite-space:normal"><br></span></pre><pre style=3D"font-size:1em;margin-top:=
0px;margin-bottom:0px;color:rgb(0,0,0)"><pre style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px">   JSEP applications typically inform the browse=
r to begin ICE gathering
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams, and thereby ICE
   components, for which to gather candidates.  However, to accelerate
   cases where the application knows the number of ICE components to use
   ahead of time, it may ask the browser to gather a pool of potential
   ICE candidates to help ensure rapid media setup.

   When setLocalDescription is eventually called, and the browser goes
   to gather the needed ICE candidates, it SHOULD start by checking if
   any candidates are available in the pool.  If there are candidates in
   the pool, they SHOULD be handed to the application immediately via
   the ICE candidate callback.  If the pool becomes depleted, either
   because a larger-than-expected number of ICE components is used, or
   because the pool has not had enough time to gather candidates, the
   remaining candidates are gathered as usual.

   One example of where this concept is useful is an application that
   expects an incoming call at some point in the future, and wants to
   minimize the time it takes to establish connectivity, to avoid
   clipping of initial media.  By pre-gathering candidates into the
   pool, it can exchange and start sending connectivity checks from
   these candidates almost immediately upon receipt of a call.  Note
   though that by holding on to these pre-gathered candidates, which
   will be kept alive as long as they may be needed, the application
   will consume resources on the STUN/TURN servers it is using.</pre></pre>=
<pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,=
0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-si=
ze:small;white-space:normal"><br></span></pre><pre style=3D"font-size:1em;m=
argin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(=
34,34,34);font-family:arial,sans-serif;font-size:small;white-space:normal">=
<br></span></pre></div></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div>

--089e013a2b2c5dd8d5050fdb97cb--


From nobody Tue Feb 24 14:41:00 2015
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Subject: Re: [rtcweb] JSEP: Question relating to ICE candidate pool
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On Tue, Feb 24, 2015 at 10:40 AM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> A question has arisen with respect to draft-ietf-rtcweb-jsep-08 Sections
> 3.4.4 and 4.1.2.  Exerpts from these sections are enclosed below.
>
> While Section 3.4.4 refers to "pre-gathering" of ICE candidates prior to
> calling of setLocalDescription(), Section 4.1.2 states that createOffer()
> does not result in candidate gathering.
>
> One way to interpret this is that while createOffer() does not result in
> firing of the onicecandidate callback (which needs to wait until
> setLocalDescription() is called), it is ok (or even encouraged) for the
> implementation to begin "pre-gathering" ICE candidates into the ICE
> candidate pool once createOffer() is called.
>

The way I understood this was that you would start pre-gathering as soon as
the pool
size was increased to > 0. I.e., potentially before CreateOffer

-Ekr


> Is this a reasonable interpretation or is there another way this
> could/should be read?
>
>
> Section 4.1.2 createOffer:
>
>    Calling this method may do things such as generate new ICE
>    credentials, but does not result in candidate gathering, or cause
>    media to start or stop flowing.
>
>
> Section 3.4.4 ICE candidate pool:
>
>
>    JSEP applications typically inform the browser to begin ICE gathering
>    via the information supplied to setLocalDescription, as this is where
>    the app specifies the number of media streams, and thereby ICE
>    components, for which to gather candidates.  However, to accelerate
>    cases where the application knows the number of ICE components to use
>    ahead of time, it may ask the browser to gather a pool of potential
>    ICE candidates to help ensure rapid media setup.
>
>    When setLocalDescription is eventually called, and the browser goes
>    to gather the needed ICE candidates, it SHOULD start by checking if
>    any candidates are available in the pool.  If there are candidates in
>    the pool, they SHOULD be handed to the application immediately via
>    the ICE candidate callback.  If the pool becomes depleted, either
>    because a larger-than-expected number of ICE components is used, or
>    because the pool has not had enough time to gather candidates, the
>    remaining candidates are gathered as usual.
>
>    One example of where this concept is useful is an application that
>    expects an incoming call at some point in the future, and wants to
>    minimize the time it takes to establish connectivity, to avoid
>    clipping of initial media.  By pre-gathering candidates into the
>    pool, it can exchange and start sending connectivity checks from
>    these candidates almost immediately upon receipt of a call.  Note
>    though that by holding on to these pre-gathered candidates, which
>    will be kept alive as long as they may be needed, the application
>    will consume resources on the STUN/TURN servers it is using.
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--f46d043c059e3b3685050fdd36ea
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Feb 24, 2015 at 10:40 AM, Bernard Aboba <span dir=3D"ltr">&lt;<=
a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@g=
mail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr">A question has arisen with respect to draft-ietf-rtcweb-jsep-08 Se=
ctions 3.4.4 and 4.1.2.=C2=A0 Exerpts from these sections are enclosed belo=
w.=C2=A0<div><br></div><div>While Section 3.4.4 refers to &quot;pre-gatheri=
ng&quot; of ICE candidates prior to calling of setLocalDescription(), Secti=
on 4.1.2 states that createOffer() does not result in candidate gathering.=
=C2=A0</div><div><br></div><div>One way to interpret this is that while cre=
ateOffer() does not result in firing of the onicecandidate callback (which =
needs to wait until setLocalDescription() is called), it is ok (or even enc=
ouraged) for the implementation to begin &quot;pre-gathering&quot; ICE cand=
idates into the ICE candidate pool once createOffer() is called.=C2=A0</div=
></div></blockquote><div><br></div><div>The way I understood this was that =
you would start pre-gathering as soon as the pool</div><div>size was increa=
sed to &gt; 0. I.e., potentially before CreateOffer</div><div><br></div><di=
v>-Ekr</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr=
"><div>Is this a reasonable interpretation or is there another way this cou=
ld/should be read? =C2=A0<br><div><br></div><div><br></div><div>Section 4.1=
.2 createOffer:=C2=A0<div><br></div><div><pre style=3D"font-size:1em;margin=
-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   Calling this method may do =
things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.</pre><pre style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre style=3D"font-si=
ze:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"co=
lor:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;white-space:=
normal">Section 3.4.4 ICE candidate pool:=C2=A0</span><br></pre><pre style=
=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;w=
hite-space:normal"><br></span></pre><pre style=3D"font-size:1em;margin-top:=
0px;margin-bottom:0px;color:rgb(0,0,0)"><pre style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px">   JSEP applications typically inform the browse=
r to begin ICE gathering
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams, and thereby ICE
   components, for which to gather candidates.  However, to accelerate
   cases where the application knows the number of ICE components to use
   ahead of time, it may ask the browser to gather a pool of potential
   ICE candidates to help ensure rapid media setup.

   When setLocalDescription is eventually called, and the browser goes
   to gather the needed ICE candidates, it SHOULD start by checking if
   any candidates are available in the pool.  If there are candidates in
   the pool, they SHOULD be handed to the application immediately via
   the ICE candidate callback.  If the pool becomes depleted, either
   because a larger-than-expected number of ICE components is used, or
   because the pool has not had enough time to gather candidates, the
   remaining candidates are gathered as usual.

   One example of where this concept is useful is an application that
   expects an incoming call at some point in the future, and wants to
   minimize the time it takes to establish connectivity, to avoid
   clipping of initial media.  By pre-gathering candidates into the
   pool, it can exchange and start sending connectivity checks from
   these candidates almost immediately upon receipt of a call.  Note
   though that by holding on to these pre-gathered candidates, which
   will be kept alive as long as they may be needed, the application
   will consume resources on the STUN/TURN servers it is using.</pre></pre>=
<pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,=
0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-si=
ze:small;white-space:normal"><br></span></pre><pre style=3D"font-size:1em;m=
argin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(=
34,34,34);font-family:arial,sans-serif;font-size:small;white-space:normal">=
<br></span></pre></div></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--f46d043c059e3b3685050fdd36ea--


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From: Justin Uberti <juberti@google.com>
Date: Tue, 24 Feb 2015 18:45:16 -0800
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Subject: Re: [rtcweb] JSEP: Question relating to ICE candidate pool
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--001a1132fb6e4af98e050fe0a16d
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Exactly what Eric said. createOffer is probably too late to be useful in
many cases.

On Tue, Feb 24, 2015 at 2:40 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
> On Tue, Feb 24, 2015 at 10:40 AM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>
>> A question has arisen with respect to draft-ietf-rtcweb-jsep-08 Sections
>> 3.4.4 and 4.1.2.  Exerpts from these sections are enclosed below.
>>
>> While Section 3.4.4 refers to "pre-gathering" of ICE candidates prior to
>> calling of setLocalDescription(), Section 4.1.2 states that createOffer()
>> does not result in candidate gathering.
>>
>> One way to interpret this is that while createOffer() does not result in
>> firing of the onicecandidate callback (which needs to wait until
>> setLocalDescription() is called), it is ok (or even encouraged) for the
>> implementation to begin "pre-gathering" ICE candidates into the ICE
>> candidate pool once createOffer() is called.
>>
>
> The way I understood this was that you would start pre-gathering as soon
> as the pool
> size was increased to > 0. I.e., potentially before CreateOffer
>
> -Ekr
>
>
>> Is this a reasonable interpretation or is there another way this
>> could/should be read?
>>
>>
>> Section 4.1.2 createOffer:
>>
>>    Calling this method may do things such as generate new ICE
>>    credentials, but does not result in candidate gathering, or cause
>>    media to start or stop flowing.
>>
>>
>> Section 3.4.4 ICE candidate pool:
>>
>>
>>    JSEP applications typically inform the browser to begin ICE gathering
>>    via the information supplied to setLocalDescription, as this is where
>>    the app specifies the number of media streams, and thereby ICE
>>    components, for which to gather candidates.  However, to accelerate
>>    cases where the application knows the number of ICE components to use
>>    ahead of time, it may ask the browser to gather a pool of potential
>>    ICE candidates to help ensure rapid media setup.
>>
>>    When setLocalDescription is eventually called, and the browser goes
>>    to gather the needed ICE candidates, it SHOULD start by checking if
>>    any candidates are available in the pool.  If there are candidates in
>>    the pool, they SHOULD be handed to the application immediately via
>>    the ICE candidate callback.  If the pool becomes depleted, either
>>    because a larger-than-expected number of ICE components is used, or
>>    because the pool has not had enough time to gather candidates, the
>>    remaining candidates are gathered as usual.
>>
>>    One example of where this concept is useful is an application that
>>    expects an incoming call at some point in the future, and wants to
>>    minimize the time it takes to establish connectivity, to avoid
>>    clipping of initial media.  By pre-gathering candidates into the
>>    pool, it can exchange and start sending connectivity checks from
>>    these candidates almost immediately upon receipt of a call.  Note
>>    though that by holding on to these pre-gathered candidates, which
>>    will be kept alive as long as they may be needed, the application
>>    will consume resources on the STUN/TURN servers it is using.
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a1132fb6e4af98e050fe0a16d
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Exactly what Eric said. createOffer is probably too late t=
o be useful in many cases.</div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Tue, Feb 24, 2015 at 2:40 PM, Eric Rescorla <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com=
</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><=
br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><span class=3D=
"">On Tue, Feb 24, 2015 at 10:40 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gm=
ail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D=
"ltr">A question has arisen with respect to draft-ietf-rtcweb-jsep-08 Secti=
ons 3.4.4 and 4.1.2.=C2=A0 Exerpts from these sections are enclosed below.=
=C2=A0<div><br></div><div>While Section 3.4.4 refers to &quot;pre-gathering=
&quot; of ICE candidates prior to calling of setLocalDescription(), Section=
 4.1.2 states that createOffer() does not result in candidate gathering.=C2=
=A0</div><div><br></div><div>One way to interpret this is that while create=
Offer() does not result in firing of the onicecandidate callback (which nee=
ds to wait until setLocalDescription() is called), it is ok (or even encour=
aged) for the implementation to begin &quot;pre-gathering&quot; ICE candida=
tes into the ICE candidate pool once createOffer() is called.=C2=A0</div></=
div></blockquote><div><br></div></span><div>The way I understood this was t=
hat you would start pre-gathering as soon as the pool</div><div>size was in=
creased to &gt; 0. I.e., potentially before CreateOffer</div><div><br></div=
><div>-Ekr</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div c=
lass=3D"h5"><div dir=3D"ltr"><div>Is this a reasonable interpretation or is=
 there another way this could/should be read? =C2=A0<br><div><br></div><div=
><br></div><div>Section 4.1.2 createOffer:=C2=A0<div><br></div><div><pre st=
yle=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   =
Calling this method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.</pre><pre style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre style=3D"font-si=
ze:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"co=
lor:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;white-space:=
normal">Section 3.4.4 ICE candidate pool:=C2=A0</span><br></pre><pre style=
=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span =
style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;w=
hite-space:normal"><br></span></pre><pre style=3D"font-size:1em;margin-top:=
0px;margin-bottom:0px;color:rgb(0,0,0)"><pre style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px">   JSEP applications typically inform the browse=
r to begin ICE gathering
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams, and thereby ICE
   components, for which to gather candidates.  However, to accelerate
   cases where the application knows the number of ICE components to use
   ahead of time, it may ask the browser to gather a pool of potential
   ICE candidates to help ensure rapid media setup.

   When setLocalDescription is eventually called, and the browser goes
   to gather the needed ICE candidates, it SHOULD start by checking if
   any candidates are available in the pool.  If there are candidates in
   the pool, they SHOULD be handed to the application immediately via
   the ICE candidate callback.  If the pool becomes depleted, either
   because a larger-than-expected number of ICE components is used, or
   because the pool has not had enough time to gather candidates, the
   remaining candidates are gathered as usual.

   One example of where this concept is useful is an application that
   expects an incoming call at some point in the future, and wants to
   minimize the time it takes to establish connectivity, to avoid
   clipping of initial media.  By pre-gathering candidates into the
   pool, it can exchange and start sending connectivity checks from
   these candidates almost immediately upon receipt of a call.  Note
   though that by holding on to these pre-gathered candidates, which
   will be kept alive as long as they may be needed, the application
   will consume resources on the STUN/TURN servers it is using.</pre></pre>=
<pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,=
0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-si=
ze:small;white-space:normal"><br></span></pre><pre style=3D"font-size:1em;m=
argin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(=
34,34,34);font-family:arial,sans-serif;font-size:small;white-space:normal">=
<br></span></pre></div></div></div></div>
<br></div></div><span class=3D"">__________________________________________=
_____<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></span></blockquote></div><br></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a1132fb6e4af98e050fe0a16d--


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Subject: [rtcweb] Fwd: [video-codec] Come hack on Daala @ IETF 92
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Distributing more broadly on Charles's suggestion.

-------- Forwarded Message --------
Subject: 	[video-codec] Come hack on Daala @ IETF 92
Date: 	Wed, 25 Feb 2015 12:38:00 -0700
From: 	Jack Moffitt <jack@metajack.im>
To: 	video-codec@ietf.org <video-codec@ietf.org>, daala@xiph.org



Charles Eckel has organized an IETF hackathon the weekend before IETF.
Daala is one of the groups participating and we'd love to have some
others come hack with us on video codecs. There are Daala projects
appropriate for many skills levels, and several of the Daala team will
be there to mentor people who need some.

Be sure to register early as there are only so many spots available.

Here are the details from Charles:

The Internet Engineering Task Force (IETF) is holding a Hackathon to
encourage developers to discuss, collaborate and develop utilities,
ideas, sample code and solutions that show practical implementations of
IETF standards. It takes place just before IETF 92, and it¹s free and
open to everyone.

All you need is a knowledge of networking technologies, and you¹ll be
guaranteed to learn something new and have fun!

Event details: https://developer.cisco.com/site/ietf-hackathon
Link to register: https://www.ietf.org/meeting/92/hackathon-signup.html

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_______________________________________________
video-codec mailing list
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From nobody Thu Feb 26 07:49:23 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Security Considerations for WebRTC
        Author          : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-08.txt
	Pages           : 24
	Date            : 2015-02-26

Abstract:
   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers, generally called "WebRTC".  The major use cases
   for WebRTC technology are real-time audio and/or video calls, Web
   conferencing, and direct data transfer.  Unlike most conventional
   real-time systems (e.g., SIP-based soft phones) WebRTC communications
   are directly controlled by a Web server, which poses new security
   challenges.  For instance, a Web browser might expose a JavaScript
   API which allows a server to place a video call.  Unrestricted access
   to such an API would allow any site which a user visited to "bug" a
   user's computer, capturing any activity which passed in front of
   their camera.  This document defines the WebRTC threat model and
   analyzes the security threats of WebRTC in that model.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-security-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-security-08


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Thu Feb 26 13:22:59 2015
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From: Ted Hardie <ted.ietf@gmail.com>
To: btdingle@gmail.com, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "rtcweb-chairs@tools.ietf.org" <rtcweb-chairs@tools.ietf.org>,  "rfc-design@rfc-editor.org" <rfc-design@rfc-editor.org>
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--001a11429990d9b7a905100459bd
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Howdy,

Barry recently raised a question about the Abstracts of WebRTC-related
RFCs:  should there be a consistent single sentence in the Abstract of all
related RFCs?  The opinion of the chairs is that any Abstract text approved
by the working group represents its consensus, and that there is no
particular need to require uniformity.  Authors and editors are, of course,
welcome to re-use the existing text.  Folks who prefer the existing text
may also, of course, suggest edits for the Abstract as for other parts of
the document.

regards,

Ted, Sean, and Cullen

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">Howdy,<br><br></div><div class=3D"gmail_default"=
 style=3D"font-family:tahoma,sans-serif;font-size:small">Barry recently rai=
sed a question about the Abstracts of WebRTC-related RFCs:=C2=A0 should the=
re be a consistent single sentence in the Abstract of all related RFCs?=C2=
=A0 The opinion of the chairs is that any Abstract text approved by the wor=
king group represents its consensus, and that there is no particular need t=
o require uniformity.=C2=A0 Authors and editors are, of course, welcome to =
re-use the existing text.=C2=A0 Folks who prefer the existing text may also=
, of course, suggest edits for the Abstract as for other parts of the docum=
ent.=C2=A0 <br><br></div><div class=3D"gmail_default" style=3D"font-family:=
tahoma,sans-serif;font-size:small">regards,<br><br></div><div class=3D"gmai=
l_default" style=3D"font-family:tahoma,sans-serif;font-size:small">Ted, Sea=
n, and Cullen<br></div></div>

--001a11429990d9b7a905100459bd--


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On 27 February 2015 at 10:22, Ted Hardie <ted.ietf@gmail.com> wrote:
> Barry recently raised a question about the Abstracts of WebRTC-related RFCs:
> should there be a consistent single sentence in the Abstract of all related
> RFCs?


What would that sentence say?


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On Thu, Feb 26, 2015 at 1:28 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 27 February 2015 at 10:22, Ted Hardie <ted.ietf@gmail.com> wrote:
> > Barry recently raised a question about the Abstracts of WebRTC-related
> RFCs:
> > should there be a consistent single sentence in the Abstract of all
> related
> > RFCs?
>
>
> What would that sentence say?
>


=E2=80=8BBarry raised the point because the Data Channel drafts used this p=
hrasing:

The WebRTC framework specifies protocol support for direct interactive
rich communication using audio,
video, and data between two peers' web-browsers.

where other drafts use other phrases. The Security architecture draft,
for example, says:

The Real-Time Communications on the Web (RTCWEB) working group is tasked wi=
th
standardizing protocols for real-time communications between Web
browsers, generally
called "WebRTC".

=E2=80=8B
and overview says:

This document gives an overview and context of a protocol suite intended
for use with real-time applications that can be deployed in browsers -
"real time communication on the Web".


=E2=80=8BTed=E2=80=8B

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Feb 26, 2015 at 1:28 PM=
, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gma=
il.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br=
><div class=3D"gmail_quote"><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"><span>On 27 February 2015 at 10:22, Ted Hardie &lt;<a href=3D"mailto:ted=
.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt; wrote:<br>
&gt; Barry recently raised a question about the Abstracts of WebRTC-related=
 RFCs:<br>
&gt; should there be a consistent single sentence in the Abstract of all re=
lated<br>
&gt; RFCs?<br>
<br>
<br>
</span>What would that sentence say?<br>
</blockquote></div><br><br><div class=3D"gmail_default" style=3D"font-famil=
y:tahoma,sans-serif;font-size:small">=E2=80=8BBarry raised the point becaus=
e the Data Channel drafts used this phrasing:<br><br><pre>The WebRTC framew=
ork specifies protocol support for direct interactive rich communication us=
ing audio, <br>video, and data between two peers&#39; web-browsers.<br><br>=
</pre><pre>where other drafts use other phrases. The Security architecture =
draft, for example, says:<br>=C2=A0<br>The Real-Time Communications on the =
Web (RTCWEB) working group is tasked with <br>standardizing protocols for r=
eal-time communications between Web browsers, generally <br>called &quot;We=
bRTC&quot;.<br></pre>=E2=80=8B<br></div><div class=3D"gmail_default" style=
=3D"font-family:tahoma,sans-serif;font-size:small">and overview says:<br><b=
r>This document gives an overview and context of a protocol suite intended =
for use with real-time applications that can be deployed in browsers - &quo=
t;real time communication on the Web&quot;.<br></div><br><br><div class=3D"=
gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">=E2=
=80=8BTed=E2=80=8B</div><br></div></div>

--089e011825c0181c90051004888e--


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From: Ted Hardie <ted.ietf@gmail.com>
To: Barry Dingle <btdingle@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "rtcweb-chairs@tools.ietf.org" <rtcweb-chairs@tools.ietf.org>
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Subject: Re: [rtcweb] Consistent description of WebRTC in abstracts
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--001a11427bbc5544f50510048d8c
Content-Type: text/plain; charset=UTF-8

My original message was cc'ed to rfc-design because I didn't catch an
auto-complete error; please follow up without them cc'ed.

thanks,

Ted

On Thu, Feb 26, 2015 at 1:22 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> Howdy,
>
> Barry recently raised a question about the Abstracts of WebRTC-related
> RFCs:  should there be a consistent single sentence in the Abstract of all
> related RFCs?  The opinion of the chairs is that any Abstract text approved
> by the working group represents its consensus, and that there is no
> particular need to require uniformity.  Authors and editors are, of course,
> welcome to re-use the existing text.  Folks who prefer the existing text
> may also, of course, suggest edits for the Abstract as for other parts of
> the document.
>
> regards,
>
> Ted, Sean, and Cullen
>

--001a11427bbc5544f50510048d8c
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">My original message was cc&#39;ed to rfc-design =
because I didn&#39;t catch an auto-complete error; please follow up without=
 them cc&#39;ed.<br><br></div><div class=3D"gmail_default" style=3D"font-fa=
mily:tahoma,sans-serif;font-size:small">thanks,<br><br></div><div class=3D"=
gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">Ted<=
br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Thu, Feb 26, 2015 at 1:22 PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"=
mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</sp=
an> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D=
"gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">How=
dy,<br><br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">Barry recently raised a question about the Abstr=
acts of WebRTC-related RFCs:=C2=A0 should there be a consistent single sent=
ence in the Abstract of all related RFCs?=C2=A0 The opinion of the chairs i=
s that any Abstract text approved by the working group represents its conse=
nsus, and that there is no particular need to require uniformity.=C2=A0 Aut=
hors and editors are, of course, welcome to re-use the existing text.=C2=A0=
 Folks who prefer the existing text may also, of course, suggest edits for =
the Abstract as for other parts of the document.=C2=A0 <br><br></div><div c=
lass=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:sma=
ll">regards,<br><br></div><div class=3D"gmail_default" style=3D"font-family=
:tahoma,sans-serif;font-size:small">Ted, Sean, and Cullen<br></div></div>
</blockquote></div><br></div>

--001a11427bbc5544f50510048d8c--


From nobody Thu Feb 26 14:25:34 2015
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Cc: ted.hardie@nominum.com, rtcweb-chairs@tools.ietf.org, rtcweb-chairs@ietf.org, draft-ietf-rtcweb-stun-consent-freshness.shepherd@ietf.org, iesg-secretary@ietf.org, rtcweb@ietf.org, draft-ietf-rtcweb-stun-consent-freshness.ad@ietf.org, draft-ietf-rtcweb-stun-consent-freshness@ietf.org
Subject: [rtcweb] Publication has been requested for draft-ietf-rtcweb-stun-consent-freshness-11
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Ted Hardie has requested publication of draft-ietf-rtcweb-stun-consent-freshness-11 as Proposed Standard on behalf of the RTCWEB working group.

Please verify the document's state at http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/


From nobody Thu Feb 26 15:37:23 2015
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Subject: [rtcweb] WGLC for draft-ietf-rtcweb-video-04
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Dear WG,

This message initiates a WGLC (Working Group Last Call) for the "WebRTC =
Video Processing and Codec Requirements=94 draft:

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-video/

Please review the draft in detail.  This WGLC will end on 13 March, =
2015.

spt=


From nobody Fri Feb 27 06:26:43 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Transports for WebRTC
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-08.txt
	Pages           : 14
	Date            : 2015-02-27

Abstract:
   This document describes the data transport protocols used by WebRTC,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-transports-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-08


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Fri Feb 27 06:31:35 2015
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Subject: [rtcweb] New version of -transport
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I've shipped a new version of -transport, since I'll be away for most of
the time between now and the I-D deadline.

There's only one significant change, which resulted from reading the
JSEP draft and discussing the matter with the JSEP editors about the
meaning of the "balanced" mode.


OLD:

    it may therefore make sense for a sending application to
   choose any of the configurations:

   o  Each media stream carried on its own 5-tuple

   o  Media streams grouped by media type into 5-tuples (such as
      carrying all audio on one 5-tuple)

   o  All media sent over a single 5-tuple, with or without
      differentiation into 6-tuples based on DSCP code points

   In each of the configurations mentioned, data channels may be carried
   in its own 5-tuple, or multiplexed together with one of the media
   flows.

   More complex configurations, such as sending a high priority video
   stream on one 5-tuple and sending all other video streams multiplexed
   together over another 5-tuple, can also be envisioned.  More
   information on mapping media flows to 5-tuples can be found in
   [I-D.ietf-rtcweb-rtp-usage].

   A sending WebRTC endpoint MUST be able to support the following
   configurations:

   o  multiplex all media and data on a single 5-tuple (fully bundled)

   o  send each media stream on its own 5-tuple and data on its own
      5-tuple (fully unbundled)

   o  bundle each media type (audio, video or data) into its own 5-tuple
      (bundling by media type)

   It MAY choose to support other configurations.


NEW:


   it may therefore make sense for a sending application to
   choose any of the configurations:

   o  Each media stream carried on its own 5-tuple

   o  Media streams grouped by media type into 5-tuples (such as
      carrying all audio on one 5-tuple)

   o  All media sent over a single 5-tuple, with or without
      differentiation into 6-tuples based on DSCP code points

   In each of the configurations mentioned, data channels may be carried
   in its own 5-tuple, or multiplexed together with one of the media
   flows.

   More complex configurations, such as sending a high priority video
   stream on one 5-tuple and sending all other video streams multiplexed
   together over another 5-tuple, can also be envisioned.  More
   information on mapping media flows to 5-tuples can be found in
   [I-D.ietf-rtcweb-rtp-usage].

   A sending implementation MUST be able to support the following
   configurations:

   o  multiplex all media and data on a single 5-tuple (fully bundled)

   o  send each media stream on its own 5-tuple and data on its own
      5-tuple (fully unbundled)

   It MAY choose to support other configurations, such as bundling each
   media type (audio, video or data) into its own 5-tuple (bundling by
   media type).

Note that the draft is now on github:

https://github.com/rtcweb-wg/rtcweb-transport

If you wish to make sure I don't forget issues, filing an issue there is
a Good Ideal; if you feel that you can improve the text, don't hesitate
to post pull requests!

Harald


From nobody Fri Feb 27 08:26:47 2015
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Harald,

In the below (both old and new) you identify an alternative that bundles 
all audio on one 5-tuple and all video on another 5-tuple.

AFAIK there is no way to tell JSEP to do that. (I once thought there 
was, but eventually learned it wasn't.) If not, does it make sense to 
list it as an alternative here? Or is that the reason you make that 
configuration optional for senders?

	Thanks,
	Paul

On 2/27/15 9:31 AM, Harald Alvestrand wrote:
> I've shipped a new version of -transport, since I'll be away for most of
> the time between now and the I-D deadline.
>
> There's only one significant change, which resulted from reading the
> JSEP draft and discussing the matter with the JSEP editors about the
> meaning of the "balanced" mode.
>
>
> OLD:
>
>      it may therefore make sense for a sending application to
>     choose any of the configurations:
>
>     o  Each media stream carried on its own 5-tuple
>
>     o  Media streams grouped by media type into 5-tuples (such as
>        carrying all audio on one 5-tuple)
>
>     o  All media sent over a single 5-tuple, with or without
>        differentiation into 6-tuples based on DSCP code points
>
>     In each of the configurations mentioned, data channels may be carried
>     in its own 5-tuple, or multiplexed together with one of the media
>     flows.
>
>     More complex configurations, such as sending a high priority video
>     stream on one 5-tuple and sending all other video streams multiplexed
>     together over another 5-tuple, can also be envisioned.  More
>     information on mapping media flows to 5-tuples can be found in
>     [I-D.ietf-rtcweb-rtp-usage].
>
>     A sending WebRTC endpoint MUST be able to support the following
>     configurations:
>
>     o  multiplex all media and data on a single 5-tuple (fully bundled)
>
>     o  send each media stream on its own 5-tuple and data on its own
>        5-tuple (fully unbundled)
>
>     o  bundle each media type (audio, video or data) into its own 5-tuple
>        (bundling by media type)
>
>     It MAY choose to support other configurations.
>
>
> NEW:
>
>
>     it may therefore make sense for a sending application to
>     choose any of the configurations:
>
>     o  Each media stream carried on its own 5-tuple
>
>     o  Media streams grouped by media type into 5-tuples (such as
>        carrying all audio on one 5-tuple)
>
>     o  All media sent over a single 5-tuple, with or without
>        differentiation into 6-tuples based on DSCP code points
>
>     In each of the configurations mentioned, data channels may be carried
>     in its own 5-tuple, or multiplexed together with one of the media
>     flows.
>
>     More complex configurations, such as sending a high priority video
>     stream on one 5-tuple and sending all other video streams multiplexed
>     together over another 5-tuple, can also be envisioned.  More
>     information on mapping media flows to 5-tuples can be found in
>     [I-D.ietf-rtcweb-rtp-usage].
>
>     A sending implementation MUST be able to support the following
>     configurations:
>
>     o  multiplex all media and data on a single 5-tuple (fully bundled)
>
>     o  send each media stream on its own 5-tuple and data on its own
>        5-tuple (fully unbundled)
>
>     It MAY choose to support other configurations, such as bundling each
>     media type (audio, video or data) into its own 5-tuple (bundling by
>     media type).
>
> Note that the draft is now on github:
>
> https://github.com/rtcweb-wg/rtcweb-transport
>
> If you wish to make sure I don't forget issues, filing an issue there is
> a Good Ideal; if you feel that you can improve the text, don't hesitate
> to post pull requests!
>
> Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From nobody Fri Feb 27 13:33:11 2015
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Den 27. feb. 2015 17:26, skrev Paul Kyzivat:
> Harald,
> 
> In the below (both old and new) you identify an alternative that bundles
> all audio on one 5-tuple and all video on another 5-tuple.
> 
> AFAIK there is no way to tell JSEP to do that. (I once thought there
> was, but eventually learned it wasn't.) If not, does it make sense to
> list it as an alternative here? Or is that the reason you make that
> configuration optional for senders?

That is the reason I make that configuration optional for senders.

There's no sense in mandating it when there is consensus to not have a
control surface to choose it, but there's no sense in forbidding it
either - there will always be endpoints with extensions, and if this
particular extension makes sense, someone will eventually make it.



> 
>     Thanks,
>     Paul
> 
> On 2/27/15 9:31 AM, Harald Alvestrand wrote:
>> I've shipped a new version of -transport, since I'll be away for most of
>> the time between now and the I-D deadline.
>>
>> There's only one significant change, which resulted from reading the
>> JSEP draft and discussing the matter with the JSEP editors about the
>> meaning of the "balanced" mode.
>>
>>
>> OLD:
>>
>>      it may therefore make sense for a sending application to
>>     choose any of the configurations:
>>
>>     o  Each media stream carried on its own 5-tuple
>>
>>     o  Media streams grouped by media type into 5-tuples (such as
>>        carrying all audio on one 5-tuple)
>>
>>     o  All media sent over a single 5-tuple, with or without
>>        differentiation into 6-tuples based on DSCP code points
>>
>>     In each of the configurations mentioned, data channels may be carried
>>     in its own 5-tuple, or multiplexed together with one of the media
>>     flows.
>>
>>     More complex configurations, such as sending a high priority video
>>     stream on one 5-tuple and sending all other video streams multiplexed
>>     together over another 5-tuple, can also be envisioned.  More
>>     information on mapping media flows to 5-tuples can be found in
>>     [I-D.ietf-rtcweb-rtp-usage].
>>
>>     A sending WebRTC endpoint MUST be able to support the following
>>     configurations:
>>
>>     o  multiplex all media and data on a single 5-tuple (fully bundled)
>>
>>     o  send each media stream on its own 5-tuple and data on its own
>>        5-tuple (fully unbundled)
>>
>>     o  bundle each media type (audio, video or data) into its own 5-tuple
>>        (bundling by media type)
>>
>>     It MAY choose to support other configurations.
>>
>>
>> NEW:
>>
>>
>>     it may therefore make sense for a sending application to
>>     choose any of the configurations:
>>
>>     o  Each media stream carried on its own 5-tuple
>>
>>     o  Media streams grouped by media type into 5-tuples (such as
>>        carrying all audio on one 5-tuple)
>>
>>     o  All media sent over a single 5-tuple, with or without
>>        differentiation into 6-tuples based on DSCP code points
>>
>>     In each of the configurations mentioned, data channels may be carried
>>     in its own 5-tuple, or multiplexed together with one of the media
>>     flows.
>>
>>     More complex configurations, such as sending a high priority video
>>     stream on one 5-tuple and sending all other video streams multiplexed
>>     together over another 5-tuple, can also be envisioned.  More
>>     information on mapping media flows to 5-tuples can be found in
>>     [I-D.ietf-rtcweb-rtp-usage].
>>
>>     A sending implementation MUST be able to support the following
>>     configurations:
>>
>>     o  multiplex all media and data on a single 5-tuple (fully bundled)
>>
>>     o  send each media stream on its own 5-tuple and data on its own
>>        5-tuple (fully unbundled)
>>
>>     It MAY choose to support other configurations, such as bundling each
>>     media type (audio, video or data) into its own 5-tuple (bundling by
>>     media type).
>>
>> Note that the draft is now on github:
>>
>> https://github.com/rtcweb-wg/rtcweb-transport
>>
>> If you wish to make sure I don't forget issues, filing an issue there is
>> a Good Ideal; if you feel that you can improve the text, don't hesitate
>> to post pull requests!
>>
>> Harald
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Sat Feb 28 15:22:03 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)
        Author          : Martin Thomson
	Filename        : draft-ietf-rtcweb-alpn-01.txt
	Pages           : 7
	Date            : 2015-02-28

Abstract:
   Application Layer Protocol Negotiation (ALPN) labels are defined for
   use in identifying Web Real-Time Communications (WebRTC) usages of
   Datagram Transport Layer Security (DTLS).  Labels are provided for
   identifying a session that uses a combination of WebRTC compatible
   media and data, and for identifying a session requiring
   confidentiality protection.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-alpn-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-alpn-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Sat Feb 28 15:24:36 2015
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From: Martin Thomson <martin.thomson@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-alpn-01.txt
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On 28 February 2015 at 15:21,  <internet-drafts@ietf.org> wrote:
>         Filename        : draft-ietf-rtcweb-alpn-01.txt

Nothing new, I just changed the year to 2015.

