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In-Reply-To: <D39AC0E1.1A3271%jon.peterson@neustar.biz>
References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <CAGL6epKhbrdNEgpLFDD+JHFCNcAfST1aCKg3votcVe6z4wVa6A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37F97548@ESESSMB209.ericsson.se> <D3562523.18AAA1%jon.peterson@neustar.biz> <CAGL6epJDu1ojsSaPnJqpNoZgCwXvOoFNoEa2ZLNtmRtKMPMyoA@mail.gmail.com> <D35B4CE3.18D9AB%jon.peterson@neustar.biz> <CAGL6epJuByqHi_Fvssyd0V-LE4=_s7HK9DB-9Z3SSVky1Ak4Xw@mail.gmail.com> <D361F10A.18F366%jon.peterson@neustar.biz> <CAGL6epKVaOyVMS_6xmdEXbWsp2UXsbbwjTW2MgM=fJWLE381dg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz> <CAGL6epKJQqprV7z3GA-PMeTWvzD_CE7=q-rcv9Tyw-WSmHndyA@mail.gmail.com> <D392BAC1.1A0530%jon.peterson@neustar.biz> <CAGL6ep++QG3uKt0Rtib_eXQVqJQS1Amr5njz1os3WT6xDy5T-w@mail.gmail.com> <D39AC0E1.1A3271%jon.peterson@neustar.biz>
From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Mon, 4 Jul 2016 09:54:20 -0400
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On Thu, Jun 30, 2016 at 4:03 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
>
>
>
>> But even if we assume a more open environment where the user agent can
>> send INVITEs directly to the conference bridge, all the conference bridge
>> needs is a signature or similar token that indicates the IdP's approval,
>> -not- any specific attribute.
>>
>
> Of course attributes are needed.
> How would otherwise the conference server know which type of conference to
> start (audio, video, etc)? what is the limitation on the number of
> participant? etc...
>
>
> This is exactly what my "deeper level of complexity" in a previous mail
> pulled apart. You need to differentiate a) what kind of conference the user
> wants from b) what kind of conference the IdP is willing to authorize.
> Your token is just an indication of what the IdP is willing to authorize,
> and if that is a superset of what the user wants, then the IdP will issue a
> token. If it is a subset, then it won't.
>
> So how does the conference server know which type of conference to start?
> Well, again, you've focused this discussion on the corner case of an ad hoc
> RFC4579 conference,
>

I disagree that this is a corner case; you could replace that conference
server with any other SIP service and the same mechanism would apply.



> in the ad hoc case the conference bridge has to infer that from the type
> of session that the user agent creates with the conference bridge with its
> initial INVITE.
>

Not sure why you are saying "the conference bridge has to *infer* that";
the information would be explicitly provided in the token.



> It's not really a very good assumption, though, which is why we quickly
> run into trouble when we start trying to understand this corner case.
> However, I think it would be safe to say that for this ad hoc case, if the
> IdP was willing to issue an authorization token, then it authorizes the
> media type(s) that the user specified in their INVITE -  and thus that
> attributes in the token to express what media types the IdP allowed would
> be at best redundant.
>
> But to be clear, a token that conveys what the IdP is willing to authorize
> does not serve the same purpose as a SIP means for a user agent to convey
> to a conference bridge what kind of conference it would like to have - nor
> as a means for a user agent to communicate to an IdP what kind of
> conference it intends to propose. Paul is mentioning mechanisms like caller
> prefs here because it would be new work, through some SIP mechanism along
> those lines, to convey that information with SIP.
>
> Outside of this corner case, though, the conference URI is acquired in an
> out-of-band fashion at the same time as conference policy is set - surely
> through HTTP, surely through a web interface, and that's where all the
> OAuth goodness can (and should) play into conference policy management.
> Trying to inflate this one ad hoc corner case into a murky argument that
> SIP needs OAuth is and has this whole time been a stretch. RFC4353 already
> says conference policy management is not a SIP function. I'm not
> particularly interested in working on overriding that, since it Is
> obviously solved by just using HTTP, without any new work on our part.
>
>
>> The semantics of the INVITE method are "hey, remote resource, I'd like to
>> set up a session with you involving the following media types." The
>> question of whether or not you accept INVITEs, which is to my mind the SIP
>> authorization problem,
>>
>
> This is the main difference between how you and I see this.
> I think that the authorization problem is much bigger than just
> authorizing a specific SIP method. The authorization should be dealing with *access
> to services* provided by SIP servers regardless of the SIP method being
> used.
>
>
> Our real disconnect I think is the degree to which you think SIP offers
> "services" or that application control is inside the scope of SIP. It's not.
>

Here are few RFCs that do just that; define services and control them:
https://tools.ietf.org/html/rfc5359
https://tools.ietf.org/html/rfc3087
https://datatracker.ietf.org/doc/rfc7463/




> SIP is a protocol that allows Internet endpoints to discover one another
> and to agree on the character of a session they'd like to share, according
> to RFC3261. If I may quote from Section 2  of that document "SIP does not
> offer conference control services" and indeed "SIP does not provide
> services."
>

Here is the full quote:

  "SIP does not provide services.  Rather, *SIP provides primitives that*
*   can be used to implement different services*.  For example, *SIP can*
*   locate a user and deliver an opaque object to his current location*.
   If this primitive is used to deliver a session description written in
   SDP, for instance, the endpoints can agree on the parameters of a
   session.  *If the same primitive is used to deliver a photo of the*
*   caller as well as the session description, a "caller ID" service can*
*   be easily implemented*.  As this example shows, *a single primitive is*
*   typically used to provide several different services*."


This means that SIP is not limited to negotiating session characteristics;
it is much more than that and could be used to provide services.

The last sentence in the above quote is really important. An INVITE could
be used to start a conference, access a voice mail, put a call on hold,
etc; these services require different authorizations.

Regards,
 Rifaat





> The only thing SIP has to do with conference services is that "SIP can be
> used to initiate a session that uses some other conference control
> protocol", look ahead to BFCP, and of course that SIP user agents can
> initiate sessions with conference bridges.
>
> Much the same way that HTTP authorizes *access to resources*, regardless
> of the HTTP method being used. You do not need a specific token for HTTP
> GET vs HTTP POST, because the method does not matter; what matters is the
> resource that the method is trying to access.
>
>
> Accessing and changing resources on the Internet is in the scope of HTTP,
> which is why HTTP should be used for the purposes you have in mind here.
> Yes, in addition to initiating sessions, SIP has a number of methods that
> let it convey data, such as INFO, MESSAGE, NOTIFY and so on, and in some
> use cases, they can similarly provide access to resources. Maybe there's a
> SIP authorization problem for those methods that we could find. But for the
> most part, the SIP authorization problem should be aligned with the purpose
> of SIP - it should be about deciding whether or not to accept session
> initiation attempts. That's a problem I'd be willing to work on.
>
> Jon Peterson
> Neustar, Inc.
>
>
> Regards,
>  Rifaat
>
>

--001a114161186801d80536cfac73
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Jun 30, 2016 at 4:03 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@ne=
ustar.biz</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;=
border-left-color:rgb(204,204,204);padding-left:1ex">



<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif"><span class=3D"">
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
<br>
</div>
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>But even if we assume a more open environment where the user agent can=
 send INVITEs directly to the conference bridge, all the conference bridge =
needs is a signature or similar token that indicates the IdP&#39;s approval=
, -not- any specific attribute.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Of course attributes are needed.=C2=A0</div>
<div>How would otherwise the conference server know which type of conferenc=
e to start (audio, video, etc)? what is the limitation on the number of par=
ticipant? etc...</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>This is exactly what my &quot;deeper level of complexity&quot; =
in a previous mail pulled apart. You need to differentiate a) what kind of =
conference the user wants from b) what kind of conference the IdP is willin=
g to authorize.=C2=A0 Your token is just an indication
 of what the IdP is willing to authorize, and if that is a superset of what=
 the user wants, then the IdP will issue a token. If it is a subset, then i=
t won&#39;t.</div>
<div><br>
</div>
<div>So how does the conference server know which type of conference to sta=
rt? Well, again, you&#39;ve focused this discussion on the corner case of a=
n ad hoc RFC4579 conference,</div></div></blockquote><div><br></div><div>I =
disagree that this is a corner case; you could replace that conference serv=
er with any other SIP service and the same mechanism would apply.</div><div=
><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-l=
eft-color:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-=
word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><div> =
in the ad hoc case the conference bridge has to infer that from the type of=
 session
 that the user agent creates with the conference bridge with its initial IN=
VITE. </div></div></blockquote><div><br></div><div>Not sure why you are say=
ing &quot;<span style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;fo=
nt-size:14px">the conference bridge has to <b>infer</b> that</span>&quot;; =
the information would be explicitly provided in the token.</div><div><br></=
div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-col=
or:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-word;co=
lor:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><div>It&#39;s=
 not really a very good assumption, though, which is why we quickly run int=
o trouble when we start trying to understand this corner case. However, I t=
hink it would be safe
 to say that for this ad hoc case, if the IdP was willing to issue an autho=
rization token, then it authorizes the media type(s) that the user specifie=
d in their INVITE - =C2=A0and thus that attributes in the token to express =
what media types the IdP allowed would
 be at best redundant.</div>
<div><br>
</div>
<div>But to be clear, a token that conveys what the IdP is willing to autho=
rize does not serve the same purpose as a SIP means for a user agent to con=
vey to a conference bridge what kind of conference it would like to have - =
nor as a means for a user agent
 to communicate to an IdP what kind of conference it intends to propose. Pa=
ul is mentioning mechanisms like caller prefs here because it would be new =
work, through some SIP mechanism along those lines, to convey that informat=
ion with SIP.</div>
<div><br>
</div>
<div>Outside of this corner case, though, the conference URI is acquired in=
 an out-of-band fashion at the same time as conference policy is set - sure=
ly through HTTP, surely through a web interface, and that&#39;s where all t=
he OAuth goodness can (and should) play
 into conference policy management. Trying to inflate this one ad hoc corne=
r case into a murky argument that SIP needs OAuth is and has this whole tim=
e been a stretch. RFC4353 already says conference policy management is not =
a SIP function. I&#39;m not particularly
 interested in working on overriding that, since it Is obviously solved by =
just using HTTP, without any new work on our part.</div><span class=3D"">
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>The semantics of the INVITE method are &quot;hey, remote resource, I&#=
39;d like to set up a session with you involving the following media types.=
&quot; The question of whether or not you accept INVITEs, which is to my mi=
nd the SIP authorization problem,</div>
</div>
</blockquote>
<div><br>
</div>
<div>This is the main difference between how you and I see this.</div>
<div>I think that the authorization problem is much bigger than just author=
izing a specific SIP method. The authorization should be dealing with
<b>access to services</b> provided by SIP servers regardless of the SIP met=
hod being used.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>Our real disconnect I think is the degree to which you think SI=
P offers &quot;services&quot; or that application control is inside the sco=
pe of SIP. It&#39;s not.</div></div></blockquote><div><br></div><div>Here a=
re few RFCs that do just that; define services and control them:</div><div>=
<a href=3D"https://tools.ietf.org/html/rfc5359">https://tools.ietf.org/html=
/rfc5359</a><br></div><div><a href=3D"https://tools.ietf.org/html/rfc3087">=
https://tools.ietf.org/html/rfc3087</a><br></div><div><a href=3D"https://da=
tatracker.ietf.org/doc/rfc7463/">https://datatracker.ietf.org/doc/rfc7463/<=
/a><br></div><div><br></div><div><br></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-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex=
"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-f=
amily:Calibri,sans-serif"><div> SIP is a protocol that allows Internet endp=
oints to discover one another and to agree on the character
 of a session they&#39;d like to share, according to RFC3261. If I may quot=
e from Section 2 =C2=A0of that document &quot;SIP does not offer conference=
 control services&quot; and indeed &quot;SIP does not provide services.&quo=
t; =C2=A0</div></div></blockquote><div><br></div><div>Here is the full quot=
e:</div><div><br></div><div><div><font face=3D"monospace, monospace">=C2=A0=
 &quot;SIP does not provide services.=C2=A0 Rather, <b><font size=3D"4">SIP=
 provides primitives that</font></b></font></div><div><font face=3D"monospa=
ce, monospace"><b><font size=3D"4">=C2=A0 =C2=A0can be used to implement di=
fferent services</font></b>.=C2=A0 For example, <b>SIP can</b></font></div>=
<div><font face=3D"monospace, monospace"><b>=C2=A0 =C2=A0locate a user and =
deliver an opaque object to his current location</b>.</font></div><div><fon=
t face=3D"monospace, monospace">=C2=A0 =C2=A0If this primitive is used to d=
eliver a session description written in</font></div><div><font face=3D"mono=
space, monospace">=C2=A0 =C2=A0SDP, for instance, the endpoints can agree o=
n the parameters of a</font></div><div><font face=3D"monospace, monospace">=
=C2=A0 =C2=A0session. =C2=A0<b>If the same primitive is used to deliver a p=
hoto of the</b></font></div><div><font face=3D"monospace, monospace"><b>=C2=
=A0 =C2=A0caller as well as the session description, a &quot;caller ID&quot=
; service can</b></font></div><div><font face=3D"monospace, monospace"><b>=
=C2=A0 =C2=A0be easily implemented</b>.=C2=A0 As this example shows, <b><fo=
nt size=3D"4">a single primitive is</font></b></font></div><div><font face=
=3D"monospace, monospace"><b><font size=3D"4">=C2=A0 =C2=A0typically used t=
o provide several different services</font></b>.&quot;</font></div><div><br=
></div></div><div><br></div><div>This means that SIP is not limited to nego=
tiating session characteristics; it is much more than that and could be use=
d to provide services.<br></div><div><br></div><div>The last sentence in th=
e above quote is really important. An INVITE could be used to start a confe=
rence, access a voice mail, put a call on hold, etc; these services require=
 different authorizations.</div><div><br></div><div>Regards,</div><div>=C2=
=A0Rifaat</div><div><br></div><div><br></div><div><br></div><div>=C2=A0</di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204)=
;padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font=
-size:14px;font-family:Calibri,sans-serif"><div>The only thing SIP has to d=
o with conference services is
 that &quot;SIP can be used to initiate a session that uses some other conf=
erence control protocol&quot;, look ahead to BFCP, and of course that SIP u=
ser agents can initiate sessions with conference bridges.</div><span class=
=3D"">
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Much the same way that HTTP authorizes <b>access to resources</b>, reg=
ardless of the HTTP method being used. You do not need a specific token for=
 HTTP GET vs HTTP POST, because the method does not matter; what matters is=
 the resource that the method is
 trying to access.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>Accessing and changing resources on the Internet is in the scop=
e of HTTP, which is why HTTP should be used for the purposes you have in mi=
nd here. Yes, in addition to initiating sessions, SIP has a number of metho=
ds that let it convey data, such as INFO,
 MESSAGE, NOTIFY and so on, and in some use cases, they can similarly provi=
de access to resources. Maybe there&#39;s a SIP authorization problem for t=
hose methods that we could find. But for the most part, the SIP authorizati=
on problem should be aligned with the
 purpose of SIP - it should be about deciding whether or not to accept sess=
ion initiation attempts. That&#39;s a problem I&#39;d be willing to work on=
.</div><span class=3D"">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</span></div>

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

--001a114161186801d80536cfac73--


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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Dual-stack GRUU
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How does dual-stack operation interact with GRUUs?

An essential characteristic of a GRUU is that it's globally accessible.
But if the device only implements one address family, or the intervening
network carries only one protocol, then a URI isn't accessible to a
device that only implements the *other* protocol.

It seems that the theoretical answer is to require a GRUU to be
accessible in practice from the global Internet via either address
family, but it seems like that would de-GRUU-ize probably most of the
GRUUs that are being used in the universe.

Dale


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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Dual-stack GRUU
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On 7/6/16 4:16 PM, Dale R. Worley wrote:
> How does dual-stack operation interact with GRUUs?
>
> An essential characteristic of a GRUU is that it's globally accessible.
> But if the device only implements one address family, or the intervening
> network carries only one protocol, then a URI isn't accessible to a
> device that only implements the *other* protocol.
>
> It seems that the theoretical answer is to require a GRUU to be
> accessible in practice from the global Internet via either address
> family, but it seems like that would de-GRUU-ize probably most of the
> GRUUs that are being used in the universe.

IMO that is a step too far.

ISTM that if the client using the GRUU can't handle the address family 
then it can deal with that itself. For instance it can find a TURN 
server to help.

	Thanks,
	Paul


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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [sipcore] Happy Eyeballs: Targets
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Roman Shpount <roman@telurix.com> writes:
> One more mechanism for checking a UDP flows is STUN keep-alive message
> described in https://tools.ietf.org/html/rfc5626#section-3.5.2. These
> messages are even lower overhead then OPTIONS messages.

Thanks for mentioning that!  I've added that to the rough draft.

Dale


From nobody Wed Jul  6 14:10:31 2016
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Brett Tate <brett@broadsoft.com> writes:
> Although this draft can update it if needed, RFC 3261 section 17.1.1.2
> paragraph four was the text of potential concern since it looks like an
> attempt to limit where T1 can be lowered. [...]

Thanks for the discussion.  I've added it to the rough draft.

Dale


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Roman Shpount <roman@telurix.com> writes:
> This all depends what you mean by the "real systems". For USA hosted PBX
> systems working with office based networks, RTT is 70ms [...]
> Please let me know if you need any more data.

That's great, especially since it covers a wide range of operational
situations.  I've added your discussion to the rough draft.

Dale


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Thanks for the information!  I've added your discussion to the rough
draft.

Roman Shpount <roman@telurix.com> writes:
> Few observations:
> [...]
> 3. For server, the most efficient way to keep the NAT hole open is to send
> OPTIONS or NOTIFY requests to the client. This is much more efficient then
> registrations with small timeouts.
>
> 4. Registration server which has an in-memory database for current
> registrations can be fairly efficient in reducing number of back-end DB
> updates due to registrations and subscriptions with small timeouts. It is
> not going to be stateless, but its state does not need to be replicated and
> would automatically recover on the stand-by server on the next registration
> or subscription request from the client.

I think what you are saying here is that if the server sends
OPTIONS/NOTIFY for keepalive, rather than making the client send
frequent re-REGISTERS, then the server doesn't need to record the timers
for the keepalives in persistent storage -- if the stand-by server is
activated, it simply sends keepalives to all registrations that are
recorded as having keepalive service.  In the long run, this requires no
more processing by the registration server and far less demand on the
persistent storage.

Is that correct?

> 5. One of the big problems with encrypted SIP traffic is that it gets
> modified by ALG. For a lot of hosted PBX providers avoiding the need to
> troubleshoot customer routers is a bigger incentive to deploy TLS then user
> privacy.

If I understand you correctly, hosted PBX providers favor TLS to stop
routers from modifying SIP messages.  But ALGs terminate TLS ... and
then modify the SIP messages.

Dale


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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> then the server doesn't need to record the timers for the keepalives in
> persistent storage
I don't think using OPTIONS/NOTIFY v.s. REGISTER would change anything on t=
his front. Roman probably refers more to the overall processing cost.

Thanks,
Tolga

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Dale R. Worl=
ey
> Sent: Wednesday, July 06, 2016 5:54 PM
> To: Roman Shpount <roman@telurix.com>
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Happy Eyeballs: Outbounde
>=20
> Thanks for the information!  I've added your discussion to the rough draf=
t.
>=20
> Roman Shpount <roman@telurix.com> writes:
> > Few observations:
> > [...]
> > 3. For server, the most efficient way to keep the NAT hole open is to
> > send OPTIONS or NOTIFY requests to the client. This is much more
> > efficient then registrations with small timeouts.
> >
> > 4. Registration server which has an in-memory database for current
> > registrations can be fairly efficient in reducing number of back-end
> > DB updates due to registrations and subscriptions with small timeouts.
> > It is not going to be stateless, but its state does not need to be
> > replicated and would automatically recover on the stand-by server on
> > the next registration or subscription request from the client.
>=20
> I think what you are saying here is that if the server sends OPTIONS/NOTI=
FY
> for keepalive, rather than making the client send frequent re-REGISTERS,
> then the server doesn't need to record the timers for the keepalives in
> persistent storage -- if the stand-by server is activated, it simply send=
s
> keepalives to all registrations that are recorded as having keepalive ser=
vice.
> In the long run, this requires no more processing by the registration ser=
ver
> and far less demand on the persistent storage.
>=20
> Is that correct?
>=20
> > 5. One of the big problems with encrypted SIP traffic is that it gets
> > modified by ALG. For a lot of hosted PBX providers avoiding the need
> > to troubleshoot customer routers is a bigger incentive to deploy TLS
> > then user privacy.
>=20
> If I understand you correctly, hosted PBX providers favor TLS to stop rou=
ters
> from modifying SIP messages.  But ALGs terminate TLS ... and then modify
> the SIP messages.
>=20
> Dale
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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To: sipcore@ietf.org
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Wed, 6 Jul 2016 18:17:44 -0400
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/ldNnVbAdQqdbs6Dw5RuQBcsCLbw>
Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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On 7/6/16 5:54 PM, Dale R. Worley wrote:
> Thanks for the information!  I've added your discussion to the rough
> draft.
>
> Roman Shpount <roman@telurix.com> writes:
>> Few observations:
>> [...]
>> 3. For server, the most efficient way to keep the NAT hole open is to send
>> OPTIONS or NOTIFY requests to the client. This is much more efficient then
>> registrations with small timeouts.

Over TCP you can use <CR><LF>. (And you don't need it very often.)

Over UDP why don't we use STUN?

In any case I see how this is a good substitute for frequent 
registration updates. The timing of both is determined by the server. 
This makes sense if the NAT is on the server end, and so its 
characteristics are known to the server.

But if the the NAT is at the client end and the client knows what is 
need is, then it may not be so good. Are we assuming that clients are 
dumb and won't know this stuff anyway, so a guess by the server is the 
best you are going to get.

	Thanks,
	Paul

>> 4. Registration server which has an in-memory database for current
>> registrations can be fairly efficient in reducing number of back-end DB
>> updates due to registrations and subscriptions with small timeouts. It is
>> not going to be stateless, but its state does not need to be replicated and
>> would automatically recover on the stand-by server on the next registration
>> or subscription request from the client.
>
> I think what you are saying here is that if the server sends
> OPTIONS/NOTIFY for keepalive, rather than making the client send
> frequent re-REGISTERS, then the server doesn't need to record the timers
> for the keepalives in persistent storage -- if the stand-by server is
> activated, it simply sends keepalives to all registrations that are
> recorded as having keepalive service.  In the long run, this requires no
> more processing by the registration server and far less demand on the
> persistent storage.
>
> Is that correct?
>
>> 5. One of the big problems with encrypted SIP traffic is that it gets
>> modified by ALG. For a lot of hosted PBX providers avoiding the need to
>> troubleshoot customer routers is a bigger incentive to deploy TLS then user
>> privacy.
>
> If I understand you correctly, hosted PBX providers favor TLS to stop
> routers from modifying SIP messages.  But ALGs terminate TLS ... and
> then modify the SIP messages.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/saIl_V9_9dyCeEg4miogjWBCyMU>
Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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Inline...

Thanks,
Tolga

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Wednesday, July 06, 2016 6:18 PM
> To: sipcore@ietf.org
> Subject: Re: [sipcore] Happy Eyeballs: Outbounde
>=20
> On 7/6/16 5:54 PM, Dale R. Worley wrote:
> > Thanks for the information!  I've added your discussion to the rough
> > draft.
> >
> > Roman Shpount <roman@telurix.com> writes:
> >> Few observations:
> >> [...]
> >> 3. For server, the most efficient way to keep the NAT hole open is to
> >> send OPTIONS or NOTIFY requests to the client. This is much more
> >> efficient then registrations with small timeouts.
>=20
> Over TCP you can use <CR><LF>. (And you don't need it very often.)
>=20
> Over UDP why don't we use STUN?
>=20
> In any case I see how this is a good substitute for frequent registration
> updates. The timing of both is determined by the server.
> This makes sense if the NAT is on the server end, and so its characterist=
ics are
> known to the server.
>=20
> But if the the NAT is at the client end and the client knows what is need=
 is,
> then it may not be so good. Are we assuming that clients are dumb and won=
't
> know this stuff anyway, so a guess by the server is the best you are goin=
g to
> get.
[TOLGA] In general, it is not common to rely on the client to determine the=
 refresh timing for NAT. It is either statically configured on the server s=
ide or server performs some sort of "probing" to determine a close-to-optim=
um refresh time. And agreed that CRLF/STUN-keepalive are the most preferred=
 approaches, just there are still lots of deployments with no UE support fo=
r those though.
>=20
> 	Thanks,
> 	Paul
>=20
> >> 4. Registration server which has an in-memory database for current
> >> registrations can be fairly efficient in reducing number of back-end
> >> DB updates due to registrations and subscriptions with small
> >> timeouts. It is not going to be stateless, but its state does not
> >> need to be replicated and would automatically recover on the stand-by
> >> server on the next registration or subscription request from the clien=
t.
> >
> > I think what you are saying here is that if the server sends
> > OPTIONS/NOTIFY for keepalive, rather than making the client send
> > frequent re-REGISTERS, then the server doesn't need to record the
> > timers for the keepalives in persistent storage -- if the stand-by
> > server is activated, it simply sends keepalives to all registrations
> > that are recorded as having keepalive service.  In the long run, this
> > requires no more processing by the registration server and far less
> > demand on the persistent storage.
> >
> > Is that correct?
> >
> >> 5. One of the big problems with encrypted SIP traffic is that it gets
> >> modified by ALG. For a lot of hosted PBX providers avoiding the need
> >> to troubleshoot customer routers is a bigger incentive to deploy TLS
> >> then user privacy.
> >
> > If I understand you correctly, hosted PBX providers favor TLS to stop
> > routers from modifying SIP messages.  But ALGs terminate TLS ... and
> > then modify the SIP messages.
> >
> > Dale
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Wed Jul  6 18:10:43 2016
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Date: Thu, 7 Jul 2016 01:10:35 +0000
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References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <CAGL6epKhbrdNEgpLFDD+JHFCNcAfST1aCKg3votcVe6z4wVa6A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37F97548@ESESSMB209.ericsson.se> <D3562523.18AAA1%jon.peterson@neustar.biz> <CAGL6epJDu1ojsSaPnJqpNoZgCwXvOoFNoEa2ZLNtmRtKMPMyoA@mail.gmail.com> <D35B4CE3.18D9AB%jon.peterson@neustar.biz> <CAGL6epJuByqHi_Fvssyd0V-LE4=_s7HK9DB-9Z3SSVky1Ak4Xw@mail.gmail.com> <D361F10A.18F366%jon.peterson@neustar.biz> <CAGL6epKVaOyVMS_6xmdEXbWsp2UXsbbwjTW2MgM=fJWLE381dg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz> <CAGL6epKJQqprV7z3GA-PMeTWvzD_CE7=q-rcv9Tyw-WSmHndyA@mail.gmail.com> <D392BAC1.1A0530%jon.peterson@neustar.biz> <CAGL6ep++QG3uKt0Rtib_eXQVqJQS1Amr5njz1os3WT6xDy5T-w@mail.gmail.com> <D39AC0E1.1A3271%jon.peterson@neustar.biz> <CAGL6epL82MaLHXxt4kO0vwegd_DOAT6ywUifMFciVgbK8328GQ@mail.gmail.com>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--_000_D3A12BC51A4777jonpetersonneustarbiz_
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So how does the conference server know which type of conference to start? W=
ell, again, you've focused this discussion on the corner case of an ad hoc =
RFC4579 conference,

I disagree that this is a corner case; you could replace that conference se=
rver with any other SIP service and the same mechanism would apply.

We've probably delved excessively into this one use case - it perhaps would=
 have been more helpful to see examples of the attributes you propose for o=
thers services.

It is a "corner case" in so far as most of RFC4579 focuses on the "sunny da=
y" case where there is a pre-existing conference URI (probably gotten via m=
eans like an "IVR, web page") that you just dial into or add people to. Onl=
y Section 5.4 deals with this "ad hoc" case. And of course RFC4579 is just =
one way of managing conferences in a landscape full of methods that use pro=
tocols other than SIP, albeit SIP will be used to actually initiate session=
s to the conference.


in the ad hoc case the conference bridge has to infer that from the type of=
 session that the user agent creates with the conference bridge with its in=
itial INVITE.

Not sure why you are saying "the conference bridge has to infer that"; the =
information would be explicitly provided in the token.

The main point of my last message was to try to tease apart a) what kind of=
 conference the user wants from b) what kind of conference the IdP is willi=
ng to authorize.  The fact that you're still just conflating them here sugg=
ests we're making little progress in even communicating, let along persuadi=
ng each other.

Upleveling, if you think a token, rather than the SDP in the SIP INVITE, te=
lls the conference bridge what kind of media the user agent is proposing fo=
r its session, then the scope of the problem you're considering is even mor=
e unwieldy than I thought. The potential difference between a) the media ty=
pes the user (as a matter of personal conference policy) would allow any pa=
rticipants to use in the conference vs. c) which media the user is proposin=
g in the SDP of the INVITE for its own connection to the conference bridge,=
 is the further wrinkle I was attempting to untangle in the line you quote =
above - many elements need to be teased apart to understand the ad hoc case=
.


Our real disconnect I think is the degree to which you think SIP offers "se=
rvices" or that application control is inside the scope of SIP. It's not.

Here are few RFCs that do just that; define services and control them:
https://tools.ietf.org/html/rfc5359
https://tools.ietf.org/html/rfc3087
https://datatracker.ietf.org/doc/rfc7463/

These are pretty much landline replacement "services" for CLASS features an=
d similar functions, so I'll talk more about them in a minute. But I'd be r=
eal interested to hear what kinds of OAuth-style attributes need to be carr=
ied in tokens in order to authorize those sorts of functions, and how the p=
ost-1970s PSTN managed to live without them for decades.


SIP is a protocol that allows Internet endpoints to discover one another an=
d to agree on the character of a session they'd like to share, according to=
 RFC3261. If I may quote from Section 2  of that document "SIP does not off=
er conference control services" and indeed "SIP does not provide services."

Here is the full quote:

  "SIP does not provide services.  Rather, SIP provides primitives that
   can be used to implement different services.  For example, SIP can
   locate a user and deliver an opaque object to his current location.
   If this primitive is used to deliver a session description written in
   SDP, for instance, the endpoints can agree on the parameters of a
   session.  If the same primitive is used to deliver a photo of the
   caller as well as the session description, a "caller ID" service can
   be easily implemented.  As this example shows, a single primitive is
   typically used to provide several different services."


This means that SIP is not limited to negotiating session characteristics; =
it is much more than that and could be used to provide services.

You passed over my RFC3261 quote that "SIP does not offer conference contro=
l services" in silence while bolding lots of other 14 point text there.

To talk about the paragraph you quote we need to open the more philosophica=
l question of "what do we consider a service?" Clearly, if caller-ID is a s=
ervice, or call waiting, or similar CLASS features, then in that sense of c=
ourse SIP provides "services" via its primitives, and I've worked on defini=
ng a few myself. But CLASS features as "services" are not of remotely the s=
ame category as OAuth third-party services that consume tokens from IDPs - =
you don't need any sort of third-party authorization attribute or token in =
an INVITE to put someone on hold to take another call, say, or to send an i=
mage between endpoints. Back in the days of IN, these kinds of features had=
 to be performed by "the network," and it was kind of a big deal that inste=
ad in SIP you just implemented them in primitives under the control of user=
 agents, hence this text. But I think you are more concerned with controlli=
ng third party applications than with features like call waiting, and confl=
ating those two categories under the umbrella of "services" to make it soun=
d like SIP encompasses and legitimizes what you want to do is not going to =
persuade me, in any event.

SIP does provide the primitives necessary to implement CLASS features, and =
a number of RFCs today describe how to implement those features with SIP pr=
imitives. And when you initiate a session with a voicemail bridge, say, we =
could authorize your access to voicemail boxes in a number of different way=
s. But if you wanted to implement a trait-based authorization system for au=
thorizing access to voicemails held on a third-party server, realistically,=
 would (or should) you do it with SIP? Because there are a lot of HTTP and =
IMAP based visual voicemail services out there that support advanced featur=
es like speech-to-text transcription and so on where OAuth-like primitives =
make a lot of sense. Just because you can call a voicemail service with a S=
IP INVITE, that doesn't means that a SIP INVITE is the right place to stage=
 an OAuth "service" like that, nor its security. I feel exactly the same wa=
y about conference policy.


The last sentence in the above quote is really important. An INVITE could b=
e used to start a conference, access a voice mail, put a call on hold, etc;=
 these services require different authorizations.

An INVITE can be used to initiate a session with a conference bridge, becau=
se semantics of an INVITE are "hey, I would like to initiate a session with=
 you with the following media parameters." Setting conference policy requir=
es a different semantics than that, and as I've repeatedly said in this thr=
ead (and as far as I can tell, you have not contradicted), the relevant RFC=
s recommend that you don't set conference policy with SIP. I'm really don't=
 understand why that does not deter you.

I think at this point, if you want to propose your way of setting conferenc=
e policy with SIP as a problem space, feel free - I don't happen to think t=
here's much of a gap there, given the existence of means other than RFC4579=
 ad hoc INVITEs to set up conferences when you have more complicated securi=
ty needs than an ad hoc conference ordinarily would. If the group agrees th=
at there is 1) a need to convey conference policy in SIP, and 2) a need for=
 attribute-based authorization tokens issued by IdPs to support that confer=
ence policy, that could be scoped as its own problem space and put forward =
for dispatch. But I don't think this is a general authorization problem for=
 SIP.

Jon Peterson
Neustar, Inc.

--_000_D3A12BC51A4777jonpetersonneustarbiz_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>So how does the conference server know which type of conference to sta=
rt? Well, again, you've focused this discussion on the corner case of an ad=
 hoc RFC4579 conference,</div>
</div>
</blockquote>
<div><br>
</div>
<div>I disagree that this is a corner case; you could replace that conferen=
ce server with any other SIP service and the same mechanism would apply.</d=
iv>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>We've probably delved excessively into this one use case - it perhaps =
would have been more helpful to see examples of the attributes you propose =
for others services.</div>
<div><br>
</div>
<div>It is a &quot;corner case&quot; in so far as most of RFC4579 focuses o=
n the &quot;sunny day&quot; case where there is a pre-existing conference U=
RI (probably gotten via means like an &quot;IVR, web page&quot;) that you j=
ust dial into or add people to. Only Section 5.4 deals with this
 &quot;ad hoc&quot; case. And of course RFC4579 is just one way of managing=
 conferences in a landscape full of methods that use protocols other than S=
IP, albeit SIP will be used to actually initiate sessions to the conference=
.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>in the ad hoc case the conference bridge has to infer that from the ty=
pe of session that the user agent creates with the conference bridge with i=
ts initial INVITE.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Not sure why you are saying &quot;<span style=3D"color:rgb(0,0,0);font=
-family:Calibri,sans-serif;font-size:14px">the conference bridge has to
<b>infer</b> that</span>&quot;; the information would be explicitly provide=
d in the token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>The main point of my last message was to try to tease apart a) what ki=
nd of conference the user wants from b) what kind of conference the IdP is =
willing to authorize. &nbsp;The fact that you're still just conflating them=
 here suggests we're making little progress
 in even communicating, let along persuading each other. &nbsp;</div>
<div><br>
</div>
<div>Upleveling, if you think a token, rather than the SDP in the SIP INVIT=
E, tells the conference bridge what kind of media the user agent is proposi=
ng for its session, then the scope of the problem you're considering is eve=
n more unwieldy than I thought.
 The potential difference between a) the media types the user (as a matter =
of personal conference policy) would allow any participants to use in the c=
onference vs. c) which media the user is proposing in the SDP of the INVITE=
 for its own connection to the conference
 bridge, is the further wrinkle I was attempting to untangle in the line yo=
u quote above - many elements need to be teased apart to understand the ad =
hoc case.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<span class=3D"">
<div><br>
</div>
</span>
<div>Our real disconnect I think is the degree to which you think SIP offer=
s &quot;services&quot; or that application control is inside the scope of S=
IP. It's not.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Here are few RFCs that do just that; define services and control them:=
</div>
<div><a href=3D"https://tools.ietf.org/html/rfc5359">https://tools.ietf.org=
/html/rfc5359</a><br>
</div>
<div><a href=3D"https://tools.ietf.org/html/rfc3087">https://tools.ietf.org=
/html/rfc3087</a><br>
</div>
<div><a href=3D"https://datatracker.ietf.org/doc/rfc7463/">https://datatrac=
ker.ietf.org/doc/rfc7463/</a></div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>These are pretty much landline replacement &quot;services&quot; for CL=
ASS features and similar functions, so I'll talk more about them in a minut=
e. But I'd be real interested to hear what kinds of OAuth-style attributes =
need to be carried in tokens in order to authorize
 those sorts of functions, and how the post-1970s PSTN managed to live with=
out them for decades.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>SIP is a protocol that allows Internet endpoints to discover one anoth=
er and to agree on the character of a session they'd like to share, accordi=
ng to RFC3261. If I may quote from Section 2 &nbsp;of that document &quot;S=
IP does not offer conference control services&quot;
 and indeed &quot;SIP does not provide services.&quot; &nbsp;</div>
</div>
</blockquote>
<div><br>
</div>
<div>Here is the full quote:</div>
<div><br>
</div>
<div>
<div><font face=3D"monospace,monospace">&nbsp; &quot;SIP does not provide s=
ervices.&nbsp; Rather, <b>
<font size=3D"4">SIP provides primitives that</font></b></font></div>
<div><font face=3D"monospace,monospace"><b><font size=3D"4">&nbsp; &nbsp;ca=
n be used to implement different services</font></b>.&nbsp; For example,
<b>SIP can</b></font></div>
<div><font face=3D"monospace,monospace"><b>&nbsp; &nbsp;locate a user and d=
eliver an opaque object to his current location</b>.</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp;If this primitive is u=
sed to deliver a session description written in</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp;SDP, for instance, the=
 endpoints can agree on the parameters of a</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp;session. &nbsp;<b>If t=
he same primitive is used to deliver a photo of the</b></font></div>
<div><font face=3D"monospace,monospace"><b>&nbsp; &nbsp;caller as well as t=
he session description, a &quot;caller ID&quot; service can</b></font></div=
>
<div><font face=3D"monospace,monospace"><b>&nbsp; &nbsp;be easily implement=
ed</b>.&nbsp; As this example shows,
<b><font size=3D"4">a single primitive is</font></b></font></div>
<div><font face=3D"monospace,monospace"><b><font size=3D"4">&nbsp; &nbsp;ty=
pically used to provide several different services</font></b>.&quot;</font>=
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span><span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<div><br>
</div>
</div>
<div><br>
</div>
<div>This means that SIP is not limited to negotiating session characterist=
ics; it is much more than that and could be used to provide services.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>
<div>You passed over my RFC3261 quote that &quot;SIP does not offer confere=
nce control services&quot; in silence while bolding lots of other 14 point =
text there.</div>
</div>
<div><br>
</div>
<div>To talk about the paragraph you quote we need to open the more philoso=
phical question of &quot;what do we consider a service?&quot; Clearly, if c=
aller-ID is a service, or call waiting, or similar CLASS features, then in =
that sense of course SIP provides &quot;services&quot;
 via its primitives, and I've worked on defining a few myself. But CLASS fe=
atures as &quot;services&quot; are not of remotely the same category as OAu=
th third-party services that consume tokens from IDPs - you don't need any =
sort of third-party authorization attribute
 or token in an INVITE to put someone on hold to take another call, say, or=
 to send an image between endpoints. Back in the days of IN, these kinds of=
 features had to be performed by &quot;the network,&quot; and it was kind o=
f a big deal that instead in SIP you just
 implemented them in primitives under the control of user agents, hence thi=
s text. But I think you are more concerned with controlling third party app=
lications than with features like call waiting, and conflating those two ca=
tegories under the umbrella of &quot;services&quot;
 to make it sound like SIP encompasses and legitimizes what you want to do =
is not going to persuade me, in any event.</div>
<div><br>
</div>
<div>SIP does provide the primitives necessary to implement CLASS features,=
 and a number of RFCs today describe how to implement those features with S=
IP primitives. And when you initiate a session with a voicemail bridge, say=
, we could authorize your access
 to voicemail boxes in a number of different ways. But if you wanted to imp=
lement a trait-based authorization system for authorizing access to voicema=
ils held on a third-party server, realistically, would (or should) you do i=
t with SIP? Because there are a
 lot of HTTP and IMAP based visual voicemail services out there that suppor=
t advanced features like speech-to-text transcription and so on where OAuth=
-like primitives make a lot of sense. Just because you can call a voicemail=
 service with a SIP INVITE, that
 doesn't means that a SIP INVITE is the right place to stage an OAuth &quot=
;service&quot; like that, nor its security. I feel exactly the same way abo=
ut conference policy.&nbsp;</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>The last sentence in the above quote is really important. An INVITE co=
uld be used to start a conference, access a voice mail, put a call on hold,=
 etc; these services require different authorizations.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>An INVITE can be used to initiate a session with a conference bridge, =
because semantics of an INVITE are &quot;hey, I would like to initiate a se=
ssion with you with the following media parameters.&quot; Setting conferenc=
e policy requires a different semantics than
 that, and as I've repeatedly said in this thread (and as far as I can tell=
, you have not contradicted), the relevant RFCs recommend that you don't se=
t conference policy with SIP. I'm really don't understand why that does not=
 deter you.</div>
<div><br>
</div>
<div>I think at this point, if you want to propose your way of setting conf=
erence policy with SIP as a problem space, feel free - I don't happen to th=
ink there's much of a gap there, given the existence of means other than RF=
C4579 ad hoc INVITEs to set up conferences
 when you have more complicated security needs than an ad hoc conference or=
dinarily would. If the group agrees that there is 1) a need to convey confe=
rence policy in SIP, and 2) a need for attribute-based authorization tokens=
 issued by IdPs to support that
 conference policy, that could be scoped as its own problem space and put f=
orward for dispatch. But I don't think this is a general authorization prob=
lem for SIP.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Jon Peterson</div>
</div>
</div>
</div>
</span>
<div>Neustar, Inc.</div>
</body>
</html>

--_000_D3A12BC51A4777jonpetersonneustarbiz_--


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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Very rough draft for remaining Happy Eyeballs work
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I've submitted a very rough draft for the remaining Happy Eyeballs work:
    https://datatracker.ietf.org/doc/draft-worley-sipcore-dual-stack/

At this point, it is an attempt to record the state of the design
discussions for a number of topics which are part of -- or related to --
Happy Eyeballs for SIP.

Dale


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Subject: Re: [sipcore] Dual-stack GRUU
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> On 06 Jul 2016, at 22:53, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
> On 7/6/16 4:16 PM, Dale R. Worley wrote:
>> How does dual-stack operation interact with GRUUs?
>>=20
>> An essential characteristic of a GRUU is that it's globally =
accessible.
>> But if the device only implements one address family, or the =
intervening
>> network carries only one protocol, then a URI isn't accessible to a
>> device that only implements the *other* protocol.
>>=20
>> It seems that the theoretical answer is to require a GRUU to be
>> accessible in practice from the global Internet via either address
>> family, but it seems like that would de-GRUU-ize probably most of the
>> GRUUs that are being used in the universe.
>=20
> IMO that is a step too far.
>=20
> ISTM that if the client using the GRUU can't handle the address family =
then it can deal with that itself. For instance it can find a TURN =
server to help.

Gruu is a good thing (TM) in dual stack environments, but I would not go =
as
far as to require it to be dual stack. I wonder if there are scenarious =
where the
AOR is dual stack, but not the GRUU. That=E2=80=99s the only case I can =
think of
where a GRUU could lead to the problems you describe. The domain in the =
GRUU maps
the AOR and resolves to the same hosts with the same address families
and have the same reachability. If you can reach the AOR you should be
able to always reach the GRUU - or am I missing something?

A gruu puts the burden of finding the proper flow to the
other side on the proxy and ICE solves the media. A dual
stack domain-responsible SIP server enables reachability from any
address family, even if the UA doesn=E2=80=99t support all of them. =
Beautiful
migration path. :-)

I would conclude with the fact that GRUU makes the contact URI address =
family independent.
In most cases this is not that important since we have record routed =
proxys
or simply b2bua=E2=80=99s in the path=E2=80=A6=20

That takes me to the issue of the record-route. In order to have a gruu-
like effect (address family independence) the address in the =
record-route
should be a domain or host name - not an ip address of either family.
This also makes TLS better.

Sorry for the rambling, just brainstorming
/O=


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Subject: Re: [sipcore] Very rough draft for remaining Happy Eyeballs work
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> On 07 Jul 2016, at 03:27, Dale R. Worley <worley@ariadne.com> wrote:
>=20
> I've submitted a very rough draft for the remaining Happy Eyeballs =
work:
>    https://datatracker.ietf.org/doc/draft-worley-sipcore-dual-stack/
>=20
> At this point, it is an attempt to record the state of the design
> discussions for a number of topics which are part of -- or related to =
--
> Happy Eyeballs for SIP.

Comments:

Abstract:
- I think we are targeting both 3263 and 3261. 3263 describes how to set =
up a list of server addresses to reach,
  while my impression is that 3261 describes more of the connection set =
up and connection handling. For server2server
  connections we touch RFC 5923 too. I would rephrase into

=E2=80=9CRFC 3263 describes how to locate a list of addresses to contact =
for a given SIP URI and other document desribes how to
set up and maintain a working IP flow over the SIP transports. These =
procedures doesn=E2=80=99t handle situations where both
the SIP Ua that acts as a client and the server are connected to =
multiple IP address families. As seen in other protocols
this can lead to long delays in flow setup time, which leads to a bad =
user experience. In SIP it is especially critical to have
a short time used from call attempt to first signal - the busy or ring =
tone. This document describes how to implement
=E2=80=98Happy Eyeballs=E2=80=99-like IP flow setups in the current set =
of transports used by the SIP protocol as well as additional client
procedures which improve the SIP user experience in many =
circumstances=E2=80=9D

To requirements I would add:

- Solutions must be documented both for UA=E2=80=99s using SIP Outbound =
extensions, GRUU extensions and those that
  does not support these extensions.

The details about a probe and the note about OPTIONS doesn=E2=80=99t =
really belong in the terminology section. I wonder if
the PROBE also should be used for registration states, not just dialog =
states. We should also document that the probe
is ONLY used for non-connection-oriented protocols (read UDP).

In the solution I think we have to add a flow state somehow. For any =
given IP address/port target found by the DNS lookup
there=E2=80=99s a state - you have it between the lines -=20
 - A. Working flow established and used
 - B. Working flow established and discarded (other flow preferred)
 - C. Not attempted to open flow
 - D. Flow failed, propation mode=20

Seems like state B and D will need a timeout timer (as you point out). =
Yay for new timers! ;-)

I don=E2=80=99t understand why you let devices break the order implied =
by RFC 3263. The order is well defined
there. Do you mean that for a given HOST you may contact the targets in =
any order (or preferred: simultaneously).

Did you quote Roman right in 3.2? "One of the big problems with =
encrypted SIP traffic is that it gets
      modified by ALG. =E2=80=9C

Seems to me that he wanted to say the opposite - non-encrypted SIP =
traffic is modified by ALGs.



/O


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Subject: Re: [sipcore] Dual-stack GRUU
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"Olle E. Johansson" <oej@edvina.net> writes:
> I wonder if there are scenarious where the AOR is dual stack, but not
> the GRUU. That's the only case I can think of where a GRUU could lead
> to the problems you describe. The domain in the GRUU maps the AOR and
> resolves to the same hosts with the same address families and have the
> same reachability. If you can reach the AOR you should be able to
> always reach the GRUU - or am I missing something?

Ah, yes, that's what gets us out of the problem -- the GRUU has the same
hostpart as the AOR (presumably), so if you can contact the AOR to set
up the call, you can contact the GRUU to manipulate it.

That analysis covers all three cases:  an IPv4-only system, an IPv6-only
system, and a fully dual-stack system.

> That takes me to the issue of the record-route. In order to have a
> gruu- like effect (address family independence) the address in the
> record-route should be a domain or host name - not an ip address of
> either family.  This also makes TLS better.

Record-Route URIs are a complicated case.  In many situations, the proxy
that adds the Record-Route is directly aware of the capabilities of the
next-hop proxy, and may choose the Record-Route URI specifically to be
reachable from the next-hop proxy.

E.g., in one situation I was designing, the proxy would
double-record-route so that it could talk IPv4 to the proxy on one side
and IPv6 to the proxy on the other side.  In these situations, it could
use an IPv4 address for one URI and an IPv6 address for the other URI.

But yes, if the proxy wants the R-R URI to be family-independent (or to
be resolvable both outside and inside the corporate private network), it
has to have a suitable hostname to hand.

Oh, yeah, one significant case is where the proxy is dialog-stateless
and has a redundant partner.  Then it wants to R-R with a URI that
resolves to either itself or its partner.

Dale


From nobody Thu Jul  7 14:42:46 2016
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Subject: [sipcore] Registration for SIPIt 32 is open
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The SIP Forum's SIPit 32 will be held at the University of New Hampshire 
Interoperability Laboratory September 12-16, 2016.

In addition to the usual thorough testing, we expect this event to focus 
on new evolutions in SIP Security such as the mechanisms defined in the 
STIR working group. We also hope to inform the new SIPBRANDY effort in 
the IETF.

More information about the SIPit is available at https://www.sipit.net.

Registration is open at https://www.regonline.com/sipit32

RjS



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From: Brett Tate <brett@broadsoft.com>
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Subject: Re: [sipcore] Very rough draft for remaining Happy Eyeballs work
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> I don=E2=80=99t understand why you let devices break the order
> implied by RFC 3263.  The order is well defined there.
> Do you mean that for a given HOST you may contact the
> targets in any order (or preferred: simultaneously).

I'm not sure what is meant by the current text.  However, RFC 3261 does
allow the application of a local policy.  Similarly, some vendors might fin=
d
it more appropriate to move previously non responsive locations to the end
of the list instead of just skipping them.


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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> > then the server doesn't need to record the timers
> > for the keepalives in persistent storage
>
> I don't think using OPTIONS/NOTIFY v.s. REGISTER
> would change anything on this front. Roman probably
> refers more to the overall processing cost.

Because refreshing a registration typically involves updating persistent
information, performing authentication, et cetera, the impacts a REGISTER
can be significantly higher than that of OPTIONS.  This is the reason why
some vendors prefer receiving OPTIONS instead of REGISTER when sent
frequently as a workaround for NAT involvement.

Since OPTIONS is frequently used as a ping instead of the original
purpose, sipcore might want to standardize a mechanism to distinguish
between the two usages.  I don't recall why prior attempts such as
draft-jones-sip-options-ping failed.


From nobody Fri Jul  8 06:42:59 2016
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-dns-dual-stack-07.txt
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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 Session Initiation Protocol Core of the IETF.

        Title           : Locating Session Initiation Protocol (SIP) Servers in a Dual-Stack IP Network
        Authors         : Olle E. Johansson
                          Gonzalo Salgueiro
                          Vijay Gurbani
                          Dale R. Worley
	Filename        : draft-ietf-sipcore-dns-dual-stack-07.txt
	Pages           : 11
	Date            : 2016-07-08

Abstract:
   RFC 3263 defines how a Session Initiation Protocol (SIP)
   implementation, given a SIP Uniform Resource Identifier (URI), should
   locate the next-hop SIP server using Domain Name System (DNS)
   procedures.  As SIP networks increasingly transition from IPv4-only
   to dual-stack, a quality user experience must be ensured for dual-
   stack SIP implementations.  This document updates the DNS procedures
   described in RFC 3263 for dual-stack SIP implementations in
   preparation for forthcoming specifications for applying Happy
   Eyeballs principles to SIP.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-dns-dual-stack/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-sipcore-dns-dual-stack-07

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-sipcore-dns-dual-stack-07


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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On 7/8/16 8:55 AM, Brett Tate wrote:

> Since OPTIONS is frequently used as a ping instead of the original
> purpose, sipcore might want to standardize a mechanism to distinguish
> between the two usages.  I don't recall why prior attempts such as
> draft-jones-sip-options-ping failed.

I just looked back at my private archive. The discussions took place in 
May and July of 2010 on the sipcore list. (I haven't looked for these in 
the archives. These days I can get a URL into the archives from my copy 
of a message, but not with ones from that far back.)

I have attached a couple of messages as hooks into those discussions

Basically we could never agree on the groundrules for when using OPTIONS 
in the way proposed made sense.

	Thanks,
	Paul



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Victor Pascual wrote:
> Would sip-ops be a better place for this?

I don't know - I don't follow that list.
If you think so, feel free to propose that publicly.

	Thanks,
	Paul

> https://www.ietf.org/mailman/listinfo/sip-ops
> 
> Sent from my iPad
> 
> On Jul 1, 2010, at 4:42 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:
> 
>> [as co-chair]
>>
>> Paul J,
>>
>> What are your intentions/expectations with respect to this draft?
>>
>> While it is written using normative language, I find nothing in it that is enhanced by being normative. So I would imagine if this is to go anywhere, it would be as a Informational document.
>>
>> At the moment I don't envision this rising to the level of becoming a milestone for this group.
>>
>> I have no problem with using this list for a modest amount of discussion about this draft - especially about its eventual outcome.
>>
>> 	Thanks,
>> 	Paul
>>
>> Paul E. Jones wrote:
>>> Folks,
>>> I just wanted to bring to your attention the fact that we've submitted a new
>>> revised version of the OPTIONS "ping" draft.  This document contains input
>>> from a number of folks who provided comments to me on various aspects of the
>>> draft.
>>> We had some discussion of moving this work to the SIP Overload Control (SOC)
>>> WG.  Given my current understanding of the scope of the SOC WG, I do not
>>> think that OPTIONS falls within the scope of their charter.  Further, this
>>> document was never produced for the purpose of providing an overload
>>> mechanism, either.
>>> I would like to have some further dialog on this document.  I've received
>>> some positive feedback from several people, as this has been seen as a
>>> useful first step in trying to define use of OPTIONS as a "ping" mechanism
>>> that is interoperable.  As I had mentioned before, there are several
>>> different variants of the same procedure in production networks and I think
>>> it's useful that we reach some agreement on how OPTIONS should be used as an
>>> application-level "ping", as RFC 3261 is a bit light on detail in this area.
>>> I know I might not have captured every comment, but that's because the
>>> discussions went in a variety of directions last time. :-)  So, my edits
>>> were primarily editorial or to add clarity to what was already written.
>>> Comments are most certainly welcome.
>>> Paul
>>>> -----Original Message-----
>>>> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
>>>> On Behalf Of Internet-Drafts@ietf.org
>>>> Sent: Thursday, July 01, 2010 12:30 AM
>>>> To: i-d-announce@ietf.org
>>>> Subject: I-D Action:draft-jones-sip-options-ping-02.txt
>>>>
>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>> directories.
>>>>
>>>> 	Title           : Using OPTIONS to Query for Operational Status in
>>>> the Session Initiation Protocol (SIP)
>>>> 	Author(s)       : V. Bhargava, et al.
>>>> 	Filename        : draft-jones-sip-options-ping-02.txt
>>>> 	Pages           : 8
>>>> 	Date            : 2010-06-30
>>>>
>>>> This document describes a procedure for using the Session Initiation
>>>> Protocol (SIP) OPTIONS method in order to allow one SIP entity to query
>>>> the operational status of another SIP entity.  Through discovery of the
>>>> status of a SIP entity, it is possible to expedite session establishment
>>>> or provide diagnostic information to network administrators.
>>>>
>>>> A URL for this Internet-Draft is:
>>>> http://www.ietf.org/internet-drafts/draft-jones-sip-options-ping-02.txt
>>>>
>>>> Internet-Drafts are also available by anonymous FTP at:
>>>> ftp://ftp.ietf.org/internet-drafts/
>>>>
>>>> Below is the data which will enable a MIME compliant mail reader
>>>> implementation to automatically retrieve the ASCII version of the
>>>> Internet-Draft.
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> 


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        "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] The predictive value of OPTIONS
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This is fragmenting into multiple issues that ought to be discussed 
separately. But I want to go home so I'm not splitting it up.

inline...

Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
>> Of Paul Kyzivat
>> Sent: Tuesday, May 11, 2010 1:51 PM
>>
>>> For me it is.
>>> I do also have a need to actually learn what the next-hop's
>>> supported extensions are (and I do mean "box-level" ones),
>>> but that's not the main driver use-case for me at this instant in time.
>> Well, I would put that into somewhat the same bucket:
>> You are looking to predict which options will be supported for requests
>> you might send in the future.
>>
>> You seem to be making an assumption that extension support is uniquely
>> defined at the "box-level". I don't think its proper to assume that in
>> general, for a variety of reasons. For instance, a B2BUA may choose not
>> to admit to support for an extension in the context of a dialog unless
>> the "other leg" supports that extension.
>> This goes double for other "attributes" that may be reported by OPTIONS,
>> such as: event packages, mime types, media formats, even methods.
> 
> Yup, I know that quite well, and have made that argument before in IETF. :)
> 
> When I said "box-level", I meant capabilities *of* the B2BUA, at least with respect to me.  Not of any other "side", but of "my side".  In practice, there is a set of capabilities that could be indicated, and are.  Some of them would be quite useful, and really only apply on a per-hop basis anyway.  It's not everything, which makes it dangerous (and why I'm not suggesting doing it right now), but that's for a later time.
> 
>  
>> (For the latter, what is a B2BUA supposed to do if it understands some
>> method, but the other "side" doesn't?)
> 
> If it accepts the method on my "side", it accepts the method.  If the other side doesn't, then it will reject it when I send it.  Those two cases are not mutually inconsistent.  The B2BUA can make a routing decision which of many "sides" to send the request to, but it really does accept it and I really should send it to find out - as opposed to when it does not accept it on my side and there's no point ever sending it such a method.  

Lets use MESSAGE for discussion. You would report MESSAGE in an Accept 
header. Then later you receive MESSAGE, forward it to the "other side", 
and get back a 405, including a list of methods that are supported, not 
including MESSAGE. Do you pass that Accept on? Or alter it to indicate 
that you *do* support MESSAGE even though you are rejecting it with a 405?

There is no way to win here. You must be inconsistent at some point. So 
the result of the OPTIONS wrt Accept cannot be trusted to predict future 
behavior. The same is true for most of the other things reported in 
OPTIONS - its really hard to say much about how relevant they are for 
predicting future behavior.

To do better, I think it is necessary to either:
- specify in the query the context of the question, or
- return a complex document that describes what is supported under
   a variety of circumstances.

I doubt we want the latter (at least I don't).

Specifying the context for which you want an answer is also pretty 
difficult to do, given the number of different contexts that are 
possible. Maybe its possible to do a few interesting ones.

One is to send an OPTIONS to a URL and get an answer that pertains to 
*that* URL, while not necessarily pertaining to any other URL. AFAIK 
that is, more or less, what the current definition of OPTIONS does. You 
might choose to extrapolate the results for that URL to other URLs, but 
that may or may not produce correct results. Of course this is also 
limited, in that it only reports the results for OPTIONS. It has no way 
to customize the results for other methods. (For instance, some MIME 
types are likely to be acceptable only in the context of certain 
methods. Application/SDP may not be acceptable for MESSAGE.)

To get beyond the answer for one URL, we need a way to indicate we want 
an answer for a class of URLs, maybe all of them. Some seem to think 
that a URL with no user part is an indicator for this. I think that is 
probably a sensible meaning for some servers, but not all of them. IMO 
there is no good value you can pick to put in the user part of the R-URI 
to indicate this. But perhaps it could be indicated via some new header. 
If this is the desire, and single answers are desired, then some 
accuracy must be given up. Presumably the answer must be the union of 
the answers for all the URIs it applies to. So for any particular URI 
the answer might be less.

Also, there is the distinction between what might be available on a good 
day, vs. what the server is currently capable of, vs. what it is 
*willing* to do. Again, I don't see how any one answer is the right one 
for this. (E.g. the server is heavily loaded. It is capable of accepting 
*some* requests, but is only willing to accept requests from privileged 
callers. Or, it is in shutting-down mode. It has plenty of resources, 
but is only willing to accept in-dialog requests.)

Again, I think to get at this will require an indicator in the request 
of which sort of answer you want. Unless we can agree the answer is 
always and only about *capability*, and never about policy.

Bottom line here: I think there needs to be a selector, or selectors, 
for which kind of answer is desired.

>>> So for me, the main use-case for sending OPTIONS right now is to
>>> learn the capability of the next-hop to process messages.
>> You mean the dynamic, current "capacity" to process messages, not the
>> theoretical "ability" to do so when resources are available? Or do you
>> mean both?
> 
> Current.

Sorry - I botched my message that you are replying to. I wrote the 
above, then the list immediately below to replace that.

>> Do you mean:
>> - the theoretical "ability" to do so when resources are available,
>> - the "capacity" to do so given current load and resource availability,
>> - the "willingness" to do so, at this time, for this requestor?
> 
> If you mean "for this requestor" to mean for the same hop (as opposed to for the same source Contact URI), then I'd argue the second and third are effectively the same thing, in practice.  I believe that has even come up in the overload discussions a long time ago, and I think there was agreement they are effectively the same thing, logically.  Just because a next-hop tells me it can't do any more, but tells someone else something different, doesn't matter to my local processing of future requests to it.

I meant whatever the server would use as input to any policy decisions 
it might make. That might mean the sender of the message (top Via 
value), the originator of the request, whatever.

For instance, administrative shutdown would be a policy that determines 
willingness to handle requests. (At least until the listeners are 
stopped.) It might apply to certain links, which is what you are getting 
at. It might also apply to certain request originators, or to in-dialog 
vs. out-of-dialog requests, etc. It might also apply to certain R-URIs.

Policies to throttle load before overload occurs could also fall into 
this category of "willingness".

The specified expectation that an OPTIONS should return 486 if the 
device (a phone presumably) is on a call also falls into this "policy" 
category, since it could alternatively hang up the existing call or put 
it on hold to accept the new call.

I can see utility in getting theoretical capability as well: it would be 
nice to be able to determine which if any of your devices could support 
video, even if currently busy. I might then choose to wait till the 
video phone is free rather than calling now on the audio phone. (But of 
course this is better done with presence.)

But, if the majority of people can agree on a single context in which 
they want an answer, then perhaps we can simply have a single indicator 
that says we want that, rather than whatever it is that OPTIONS 
otherwise returns.

>> I expect that people have had one, or more, of the meanings I list above
>> in mind, perhaps without realizing that they are different. I'm willing
>> to bet not everyone will agree its the same one of the above. OPTIONS,
>> as currently defined, cannot provide all three answers. Nor will
>> tightening up the specs change that, though doing so might make it do a
>> better job of *one* of those.
> 
> In every case I have seen this used for in deployment, as far as I can tell they have actually meant the same thing.

To be fair, ACME has a particular perspective. SBCs talk to a certain 
class of equipment. I doubt you have need to interface directly to 
phones. But others do use OPTIONS to phones.

>> Then, lets talk about what range of things a given response applies to:
>>
>> Suppose I send OPTIONS to
>> - sip:198.51.100.101:5060
>> Does the response I get apply also to
>> - sip:198.51.100.101:5061,
>> - sip:alice@198.51.100.101:5060,
>> - sip:alice@198.51.100.101:5060,
>> - sips:carol@198.51.100.101:5061
>> - sip:example.com	(maps to 198.51.100.101)
>> - sip:pbx1.example.com	(also maps to 198.51.100.101)
> 
> YES.  If the responder understands the Header, then it's reporting a system-level capability. (luckily you forgot about anycasting ;)

I slipped some ringers in there for you. :-)

There is no guarantee that two ports for the same ip address are served 
by the same "server". They may be served by separate processes with 
independent resources. Statistically there is probably a good chance 
that 5060 and 5061 are served by the same server. But once you get past 
that, then what?

I didn't put in multi-homing situations, or IPv6/IPv6 cases, but they 
apply too. For all of these I can't see a good alternative to assuming 
they are distinct and asking on each, or *configuring* the cases when 
you *know* they are the same.

And I still disagree with assuming they are the same when the user part 
varies, unless we add something to the message to indicate that is what 
is desired. But even an indicator won't deal with the port issue, or 
with multihoming, since you can't know if the device speaks for the 
others. OR, the response could be altered to indicate all the different 
addresses is applies to.

>> But for a "phone" sip:198.51.100.101:5060 might just denote the "primary
>> line", while sip:2@198.51.100.101:5060 may denote the "secondary line".
>> And then the answer for one might have little predictive value over
>> requests to the other. E.g. an OPTIONS to one might return 486 because
>> the phone is busy, while an OPTIONS to the other might return 200.
> 
> Hence the need/desire for a Header.
> 
>  
>> For a gateway, there may well be DSP resources required to establish a
>> call, and how much, or what type may depend on the codecs used in the
>> call. So an answer to OPTIONS won't be valuable in predicting if a
>> particular call can succeed.
> 
> That's OK.  A "box-wide" semantic is a superset.  If you can handle ANY more calls (from that previous hop), then "box-wide" you can handle more calls.  That doesn't mean you can't reject future calls due to unavailable resources such as DSPs.  If you can NOT handle any more "box-wide", for example due to CPU, then the number of DSP resources you have really doesn't matter.

That is a possible interpretation.

If some (expensive) codecs are disabled during peak load, should that be 
reflected in the response?

What if some destinations are currently unreachable? Should OPTIONS for 
a particular address reflect that? (E.g. only +1... calls are currently 
operational because the international trunk is down. Should OPTIONS to 
sip:+44... return 480 or 200?)

	Thanks,
	Paul
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On Wed, Jul 6, 2016 at 9:10 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
>
>> So how does the conference server know which type of conference to start=
?
>> Well, again, you've focused this discussion on the corner case of an ad =
hoc
>> RFC4579 conference,
>>
>
> I disagree that this is a corner case; you could replace that conference
> server with any other SIP service and the same mechanism would apply.
>
>
> We've probably delved excessively into this one use case - it perhaps
> would have been more helpful to see examples of the attributes you propos=
e
> for others services.
>

Sure. I could do that.


>
> It is a "corner case" in so far as most of RFC4579 focuses on the "sunny
> day" case where there is a pre-existing conference URI (probably gotten v=
ia
> means like an "IVR, web page") that you just dial into or add people to.
> Only Section 5.4 deals with this "ad hoc" case. And of course RFC4579 is
> just one way of managing conferences in a landscape full of methods that
> use protocols other than SIP, albeit SIP will be used to actually initiat=
e
> sessions to the conference.
>
> As I mentioned multiple times already, my intention is *not* to
specifically discuss the conference use case, but discuss the idea of
delegating AuthNZ to a separate party other than the server providing the
service.


>
> in the ad hoc case the conference bridge has to infer that from the type
>> of session that the user agent creates with the conference bridge with i=
ts
>> initial INVITE.
>>
>
> Not sure why you are saying "the conference bridge has to *infer* that";
> the information would be explicitly provided in the token.
>
>
> The main point of my last message was to try to tease apart a) what kind
> of conference the user wants from b) what kind of conference the IdP is
> willing to authorize.  The fact that you're still just conflating them he=
re
> suggests we're making little progress in even communicating, let along
> persuading each other.
>
> Upleveling, if you think a token, rather than the SDP in the SIP INVITE,
> tells the conference bridge what kind of media the user agent is proposin=
g
> for its session,
>

No, that is *not* what I am suggesting at all.



> then the scope of the problem you're considering is even more unwieldy
> than I thought. The potential difference between a) the media types the
> user (as a matter of personal conference policy) would allow any
> participants to use in the conference vs. c) which media the user is
> proposing in the SDP of the INVITE for its own connection to the conferen=
ce
> bridge, is the further wrinkle I was attempting to untangle in the line y=
ou
> quote above - many elements need to be teased apart to understand the ad
> hoc case.
>
>
>> Our real disconnect I think is the degree to which you think SIP offers
>> "services" or that application control is inside the scope of SIP. It's =
not.
>>
>
> Here are few RFCs that do just that; define services and control them:
> https://tools.ietf.org/html/rfc5359
> https://tools.ietf.org/html/rfc3087
> https://datatracker.ietf.org/doc/rfc7463/
>
>
> These are pretty much landline replacement "services" for CLASS features
> and similar functions,
>

I have no idea what you mean by "CLASS feature"; can you please elaborate?




> so I'll talk more about them in a minute. But I'd be real interested to
> hear what kinds of OAuth-style attributes need to be carried in tokens in
> order to authorize those sorts of functions, and how the post-1970s PSTN
> managed to live without them for decades.
>
>
> SIP is a protocol that allows Internet endpoints to discover one another
>> and to agree on the character of a session they'd like to share, accordi=
ng
>> to RFC3261. If I may quote from Section 2  of that document "SIP does no=
t
>> offer conference control services" and indeed "SIP does not provide
>> services."
>>
>
> Here is the full quote:
>
>   "SIP does not provide services.  Rather, * SIP provides primitives that=
*
> *   can be used to implement different services*.  For example, *SIP can*
> *   locate a user and deliver an opaque object to his current location*.
>    If this primitive is used to deliver a session description written in
>    SDP, for instance, the endpoints can agree on the parameters of a
>    session.  *If the same primitive is used to deliver a photo of the*
> *   caller as well as the session description, a "caller ID" service can*
> *   be easily implemented*.  As this example shows, *a single primitive
> is*
> *   typically used to provide several different services*."
>
>
>
> This means that SIP is not limited to negotiating session characteristics=
;
> it is much more than that and could be used to provide services.
>
>
> You passed over my RFC3261 quote that "SIP does not offer conference
> control services" in silence while
>

 As I mentioned above, my goal is not to specifically discuss conference
control services.



> bolding lots of other 14 point text there.
>

Sorry about that. I use Gmail which does not provide much control over the
size of the font, I selected the one above Normal, which happened to be
Large =F0=9F=98=80.



> To talk about the paragraph you quote we need to open the more
> philosophical question of "what do we consider a service?" Clearly, if
> caller-ID is a service, or call waiting, or similar CLASS features, then =
in
> that sense of course SIP provides "services" via its primitives, and I've
> worked on defining a few myself. But CLASS features as "services" are not
> of remotely the same category as OAuth third-party services that consume
> tokens from IDPs - you don't need any sort of third-party authorization
> attribute or token in an INVITE to put someone on hold to take another
> call, say, or to send an image between endpoints.
>

While I agree that some of these are basic features that would probably be
provided by default, I do not think that our goal here is to tell the
service provided what and what not to authorize.
The goal should be to define a *framework *that would allow them to choose
what they want to authorize.



> Back in the days of IN, these kinds of features had to be performed by
> "the network," and it was kind of a big deal that instead in SIP you just
> implemented them in primitives under the control of user agents, hence th=
is
> text. But I think you are more concerned with controlling third party
> applications than with features like call waiting, and conflating those t=
wo
> categories under the umbrella of "services" to make it sound like SIP
> encompasses and legitimizes what you want to do is not going to persuade
> me, in any event.
>

Let me make sure I understand what you are saying here,
Are you saying that just because the SIP application is provided by a third
party server you would not consider that to be a SIP service?



>
> SIP does provide the primitives necessary to implement CLASS features, an=
d
> a number of RFCs today describe how to implement those features with SIP
> primitives. And when you initiate a session with a voicemail bridge, say,
> we could authorize your access to voicemail boxes in a number of differen=
t
> ways. But if you wanted to implement a trait-based authorization system f=
or
> authorizing access to voicemails held on a third-party server,
> realistically, would (or should) you do it with SIP?
>

Why not?
There are existing SIP application servers that do that already.



> Because there are a lot of HTTP and IMAP based visual voicemail services
> out there that support advanced features like speech-to-text transcriptio=
n
> and so on where OAuth-like primitives make a lot of sense. Just because y=
ou
> can call a voicemail service with a SIP INVITE, that doesn't means that a
> SIP INVITE is the right place to stage an OAuth "service" like that, nor
> its security. I feel exactly the same way about conference policy.
>

We are obviously in disagreement here, but I am not sure I understand the "=
nor
its security" comment; can you elaborate?



>
>
> The last sentence in the above quote is really important. An INVITE could
> be used to start a conference, access a voice mail, put a call on hold,
> etc; these services require different authorizations.
>
>
> An INVITE can be used to initiate a session with a conference bridge,
> because semantics of an INVITE are "hey, I would like to initiate a sessi=
on
> with you with the following media parameters." Setting conference policy
> requires a different semantics than that, and as I've repeatedly said in
> this thread (and as far as I can tell, you have not contradicted), the
> relevant RFCs recommend that you don't set conference policy with SIP. I'=
m
> really don't understand why that does not deter you.
>

> I think at this point, if you want to propose your way of setting
> conference policy with SIP as a problem space, feel free - I don't happen
> to think there's much of a gap there, given the existence of means other
> than RFC4579 ad hoc INVITEs to set up conferences when you have more
> complicated security needs than an ad hoc conference ordinarily would. If
> the group agrees that there is 1) a need to convey conference policy in
> SIP, and 2) a need for attribute-based authorization tokens issued by IdP=
s
> to support that conference policy,
>

Again, I am *not* suggesting to use SIP to set the policy (1); SIP will
only be used to carry the authorization (2).

Regards,
 Rifaat



> that could be scoped as its own problem space and put forward for
> dispatch. But I don't think this is a general authorization problem for S=
IP.
>
> Jon Peterson
> Neustar, Inc.
>

--94eb2c0c512040d45b0537224f33
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 Wed, Jul 6, 2016 at 9:10 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a =
href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neu=
star.biz</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;=
border-left-color:rgb(204,204,204);padding-left:1ex">



<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif"><span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>So how does the conference server know which type of conference to sta=
rt? Well, again, you&#39;ve focused this discussion on the corner case of a=
n ad hoc RFC4579 conference,</div>
</div>
</blockquote>
<div><br>
</div>
<div>I disagree that this is a corner case; you could replace that conferen=
ce server with any other SIP service and the same mechanism would apply.</d=
iv>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>We&#39;ve probably delved excessively into this one use case - =
it perhaps would have been more helpful to see examples of the attributes y=
ou propose for others services.</div></div></blockquote><div><br></div><div=
>Sure. I could do that.</div><div>=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-st=
yle:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div style=
=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibr=
i,sans-serif">
<div><br>
</div>
<div>It is a &quot;corner case&quot; in so far as most of RFC4579 focuses o=
n the &quot;sunny day&quot; case where there is a pre-existing conference U=
RI (probably gotten via means like an &quot;IVR, web page&quot;) that you j=
ust dial into or add people to. Only Section 5.4 deals with this
 &quot;ad hoc&quot; case. And of course RFC4579 is just one way of managing=
 conferences in a landscape full of methods that use protocols other than S=
IP, albeit SIP will be used to actually initiate sessions to the conference=
.</div><span>
<div><br></div></span></div></blockquote><div>As I mentioned multiple times=
 already, my intention is <b>not</b> to specifically discuss the conference=
 use case, but discuss the idea of delegating AuthNZ to a separate party ot=
her than the server providing the service.</div><div>=C2=A0</div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width=
:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-lef=
t:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;f=
ont-family:Calibri,sans-serif"><span><div>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>in the ad hoc case the conference bridge has to infer that from the ty=
pe of session that the user agent creates with the conference bridge with i=
ts initial INVITE.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Not sure why you are saying &quot;<span style=3D"color:rgb(0,0,0);font=
-family:Calibri,sans-serif;font-size:14px">the conference bridge has to
<b>infer</b> that</span>&quot;; the information would be explicitly provide=
d in the token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>The main point of my last message was to try to tease apart a) =
what kind of conference the user wants from b) what kind of conference the =
IdP is willing to authorize.=C2=A0 The fact that you&#39;re still just conf=
lating them here suggests we&#39;re making little progress
 in even communicating, let along persuading each other. =C2=A0</div>
<div><br>
</div>
<div>Upleveling, if you think a token, rather than the SDP in the SIP INVIT=
E, tells the conference bridge what kind of media the user agent is proposi=
ng for its session, </div></div></blockquote><div><br></div><div>No, that i=
s <b>not</b> what I am suggesting at all.</div><div><br></div><div>=C2=A0</=
div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bor=
der-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,20=
4);padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);fo=
nt-size:14px;font-family:Calibri,sans-serif"><div>then the scope of the pro=
blem you&#39;re considering is even more unwieldy than I thought.
 The potential difference between a) the media types the user (as a matter =
of personal conference policy) would allow any participants to use in the c=
onference vs. c) which media the user is proposing in the SDP of the INVITE=
 for its own connection to the conference
 bridge, is the further wrinkle I was attempting to untangle in the line yo=
u quote above - many elements need to be teased apart to understand the ad =
hoc case.</div><span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<span>
<div><br>
</div>
</span>
<div>Our real disconnect I think is the degree to which you think SIP offer=
s &quot;services&quot; or that application control is inside the scope of S=
IP. It&#39;s not.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Here are few RFCs that do just that; define services and control them:=
</div>
<div><a href=3D"https://tools.ietf.org/html/rfc5359" target=3D"_blank">http=
s://tools.ietf.org/html/rfc5359</a><br>
</div>
<div><a href=3D"https://tools.ietf.org/html/rfc3087" target=3D"_blank">http=
s://tools.ietf.org/html/rfc3087</a><br>
</div>
<div><a href=3D"https://datatracker.ietf.org/doc/rfc7463/" target=3D"_blank=
">https://datatracker.ietf.org/doc/rfc7463/</a></div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>These are pretty much landline replacement &quot;services&quot;=
 for CLASS features and similar functions,</div></div></blockquote><div><br=
></div><div>I have no idea what you mean by &quot;CLASS feature&quot;; can =
you please elaborate?</div><div><br></div><div><br></div><div>=C2=A0</div><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);pa=
dding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-si=
ze:14px;font-family:Calibri,sans-serif"><div> so I&#39;ll talk more about t=
hem in a minute. But I&#39;d be real interested to hear what kinds of OAuth=
-style attributes need to be carried in tokens in order to authorize
 those sorts of functions, and how the post-1970s PSTN managed to live with=
out them for decades.</div><span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>SIP is a protocol that allows Internet endpoints to discover one anoth=
er and to agree on the character of a session they&#39;d like to share, acc=
ording to RFC3261. If I may quote from Section 2 =C2=A0of that document &qu=
ot;SIP does not offer conference control services&quot;
 and indeed &quot;SIP does not provide services.&quot; =C2=A0</div>
</div>
</blockquote>
<div><br>
</div>
<div>Here is the full quote:</div>
<div><br>
</div>
<div>
<div><font face=3D"monospace,monospace">=C2=A0 &quot;SIP does not provide s=
ervices.=C2=A0 Rather, <b>
<font size=3D"4">SIP provides primitives that</font></b></font></div>
<div><font face=3D"monospace,monospace"><b><font size=3D"4">=C2=A0 =C2=A0ca=
n be used to implement different services</font></b>.=C2=A0 For example,
<b>SIP can</b></font></div>
<div><font face=3D"monospace,monospace"><b>=C2=A0 =C2=A0locate a user and d=
eliver an opaque object to his current location</b>.</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0If this primitive is u=
sed to deliver a session description written in</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0SDP, for instance, the=
 endpoints can agree on the parameters of a</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0session. =C2=A0<b>If t=
he same primitive is used to deliver a photo of the</b></font></div>
<div><font face=3D"monospace,monospace"><b>=C2=A0 =C2=A0caller as well as t=
he session description, a &quot;caller ID&quot; service can</b></font></div=
>
<div><font face=3D"monospace,monospace"><b>=C2=A0 =C2=A0be easily implement=
ed</b>.=C2=A0 As this example shows,
<b><font size=3D"4">a single primitive is</font></b></font></div>
<div><font face=3D"monospace,monospace"><b><font size=3D"4">=C2=A0 =C2=A0ty=
pically used to provide several different services</font></b>.&quot;</font>=
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<div><br>
</div>
</div>
<div><br>
</div>
<div>This means that SIP is not limited to negotiating session characterist=
ics; it is much more than that and could be used to provide services.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>
<div>You passed over my RFC3261 quote that &quot;SIP does not offer confere=
nce control services&quot; in silence while </div></div></div></blockquote>=
<div><br></div><div><div>=C2=A0As I mentioned above, my goal is not to spec=
ifically discuss conference control services.</div></div><div><br></div><di=
v>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(=
204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb=
(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><div><div>bolding lo=
ts of other 14 point text there.</div></div></div></blockquote><div><br></d=
iv><div>Sorry about that. I use Gmail which does not provide much control o=
ver the size of the font, I selected the one above Normal, which happened t=
o be Large=C2=A0=F0=9F=98=80.<br></div><div><br></div><div><br></div><block=
quote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-w=
idth:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding=
-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14=
px;font-family:Calibri,sans-serif"><div>
</div>
<div><br>
</div>
<div>To talk about the paragraph you quote we need to open the more philoso=
phical question of &quot;what do we consider a service?&quot; Clearly, if c=
aller-ID is a service, or call waiting, or similar CLASS features, then in =
that sense of course SIP provides &quot;services&quot;
 via its primitives, and I&#39;ve worked on defining a few myself. But CLAS=
S features as &quot;services&quot; are not of remotely the same category as=
 OAuth third-party services that consume tokens from IDPs - you don&#39;t n=
eed any sort of third-party authorization attribute
 or token in an INVITE to put someone on hold to take another call, say, or=
 to send an image between endpoints. </div></div></blockquote><div><br></di=
v><div>While I agree that some of these are basic features that would proba=
bly be provided by default, I do not think that our goal here is to tell th=
e service provided what and what not to authorize.</div><div>The goal shoul=
d be to define a <b>framework </b>that would allow them to choose what they=
 want to authorize.</div><div><br></div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">=
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif"><div>Back in the days of IN, these kinds of feature=
s had to be performed by &quot;the network,&quot; and it was kind of a big =
deal that instead in SIP you just
 implemented them in primitives under the control of user agents, hence thi=
s text. But I think you are more concerned with controlling third party app=
lications than with features like call waiting, and conflating those two ca=
tegories under the umbrella of &quot;services&quot;
 to make it sound like SIP encompasses and legitimizes what you want to do =
is not going to persuade me, in any event.</div></div></blockquote><div><br=
></div><div>Let me make sure I understand what you are saying here,</div><d=
iv>Are you saying that just because the SIP application is provided by a th=
ird party server you would not consider that to be a SIP service?</div><div=
><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-l=
eft-color:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-=
word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif">
<div><br>
</div>
<div>SIP does provide the primitives necessary to implement CLASS features,=
 and a number of RFCs today describe how to implement those features with S=
IP primitives. And when you initiate a session with a voicemail bridge, say=
, we could authorize your access
 to voicemail boxes in a number of different ways. But if you wanted to imp=
lement a trait-based authorization system for authorizing access to voicema=
ils held on a third-party server, realistically, would (or should) you do i=
t with SIP? </div></div></blockquote><div><br></div><div>Why not?</div><div=
>There are existing SIP application servers that do that already.</div><div=
><br></div><div>=C2=A0<br></div><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;bord=
er-left-color:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:br=
eak-word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><d=
iv>Because there are a
 lot of HTTP and IMAP based visual voicemail services out there that suppor=
t advanced features like speech-to-text transcription and so on where OAuth=
-like primitives make a lot of sense. Just because you can call a voicemail=
 service with a SIP INVITE, that
 doesn&#39;t means that a SIP INVITE is the right place to stage an OAuth &=
quot;service&quot; like that, nor its security. I feel exactly the same way=
 about conference policy.=C2=A0<span style=3D"font-family:arial,sans-serif;=
font-size:small;color:rgb(34,34,34)">=C2=A0</span></div></div></blockquote>=
<div><br></div><div>We are obviously in disagreement here, but I am not sur=
e I understand the &quot;<span style=3D"color:rgb(0,0,0);font-family:Calibr=
i,sans-serif;font-size:14px">nor its security</span>&quot; comment; can you=
 elaborate?</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left=
-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div styl=
e=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calib=
ri,sans-serif"><span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>The last sentence in the above quote is really important. An INVITE co=
uld be used to start a conference, access a voice mail, put a call on hold,=
 etc; these services require different authorizations.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>An INVITE can be used to initiate a session with a conference b=
ridge, because semantics of an INVITE are &quot;hey, I would like to initia=
te a session with you with the following media parameters.&quot; Setting co=
nference policy requires a different semantics than
 that, and as I&#39;ve repeatedly said in this thread (and as far as I can =
tell, you have not contradicted), the relevant RFCs recommend that you don&=
#39;t set conference policy with SIP. I&#39;m really don&#39;t understand w=
hy that does not deter you.<span style=3D"font-family:arial,sans-serif;font=
-size:small;color:rgb(34,34,34)">=C2=A0</span></div></div></blockquote><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left=
-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);paddi=
ng-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:=
14px;font-family:Calibri,sans-serif">
<div><br>
</div>
<div>I think at this point, if you want to propose your way of setting conf=
erence policy with SIP as a problem space, feel free - I don&#39;t happen t=
o think there&#39;s much of a gap there, given the existence of means other=
 than RFC4579 ad hoc INVITEs to set up conferences
 when you have more complicated security needs than an ad hoc conference or=
dinarily would. If the group agrees that there is 1) a need to convey confe=
rence policy in SIP, and 2) a need for attribute-based authorization tokens=
 issued by IdPs to support that
 conference policy,</div></div></blockquote><div><br></div><div>Again, I am=
 <b>not</b> suggesting to use SIP to set the policy (1); SIP will only be u=
sed to carry the authorization (2).</div><div><br></div><div>Regards,</div>=
<div>=C2=A0Rifaat</div><div><br></div><div>=C2=A0</div><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><di=
v style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family=
:Calibri,sans-serif"><div> that could be scoped as its own problem space an=
d put forward for dispatch. But I don&#39;t think this is a general authori=
zation problem for SIP.</div><span><font color=3D"#888888">
<div><br>
</div>
<span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Jon Peterson</div>
</div>
</div>
</div>
</span>
<div>Neustar, Inc.</div>
</font></span></div>

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

--94eb2c0c512040d45b0537224f33--


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Subject: [sipcore] Declaring SIP 1.0 historic?
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Friends,

Is it time to declare SIP 1.0 historic? I haven=E2=80=99t met a SIP 1.0 =
implementation in use for many, many years.

https://tools.ietf.org/html/rfc2543

/O=


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From: Roman Shpount <roman@telurix.com>
Date: Wed, 13 Jul 2016 10:23:39 -0400
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Yes, please. I would be happy to remove the RFC 2543 transaction matching.

_____________
Roman Shpount

On Wed, Jul 13, 2016 at 10:15 AM, Olle E. Johansson <oej@edvina.net> wrote:

> Friends,
>
> Is it time to declare SIP 1.0 historic? I haven=E2=80=99t met a SIP 1.0
> implementation in use for many, many years.
>
> https://tools.ietf.org/html/rfc2543
>
> /O
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr">Yes, please. I would be happy to remove the RFC 2543 trans=
action matching.</div><div class=3D"gmail_extra"><br clear=3D"all"><div><di=
v class=3D"gmail_signature" data-smartmail=3D"gmail_signature">____________=
_<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Wed, Jul 13, 2016 at 10:15 AM, Olle E. Jo=
hansson <span dir=3D"ltr">&lt;<a href=3D"mailto:oej@edvina.net" target=3D"_=
blank">oej@edvina.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">Friends,<br>
<br>
Is it time to declare SIP 1.0 historic? I haven=E2=80=99t met a SIP 1.0 imp=
lementation in use for many, many years.<br>
<br>
<a href=3D"https://tools.ietf.org/html/rfc2543" rel=3D"noreferrer" target=
=3D"_blank">https://tools.ietf.org/html/rfc2543</a><br>
<br>
/O<br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</blockquote></div><br></div>

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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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Hi,

RFC 2543 is version 2.0.

Are you wanting to declare RFC 2543 as historic or whatever defined version
1.0 as historic?

What would be the benefit of declaring RFC 2543 (or version 1.0) as
historic?  For instance, are you wanting to remove some of the backward
compatibility related mandates contained within RFC 3261?

Thanks,
Brett

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E.
> Johansson
> Sent: Wednesday, July 13, 2016 10:15 AM
> To: sipcore@ietf.org
> Cc: Olle E Johansson
> Subject: [sipcore] Declaring SIP 1.0 historic?
>
> Friends,
>
> Is it time to declare SIP 1.0 historic? I haven=E2=80=99t met a SIP 1.0
> implementation in use for many, many years.
>
> https://tools.ietf.org/html/rfc2543
>
> /O
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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> On 13 Jul 2016, at 16:28, Brett Tate <brett@broadsoft.com> wrote:
>=20
> Hi,
>=20
> RFC 2543 is version 2.0.
My mistake.
>=20
> Are you wanting to declare RFC 2543 as historic or whatever defined =
version
> 1.0 as historic?
RFC 2543 to historic
>=20
> What would be the benefit of declaring RFC 2543 (or version 1.0) as
> historic?  For instance, are you wanting to remove some of the =
backward
> compatibility related mandates contained within RFC 3261?
I don=E2=80=99t really know - but I suspect there are a lot of legacy =
stuff hanging around
like Roman pointed out. Maybe we can make things more simple.

Just wanting to open a discussion and some housecleaning.

/O
>=20
> Thanks,
> Brett
>=20
>> -----Original Message-----
>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E.
>> Johansson
>> Sent: Wednesday, July 13, 2016 10:15 AM
>> To: sipcore@ietf.org
>> Cc: Olle E Johansson
>> Subject: [sipcore] Declaring SIP 1.0 historic?
>>=20
>> Friends,
>>=20
>> Is it time to declare SIP 1.0 historic? I haven=E2=80=99t met a SIP =
1.0
>> implementation in use for many, many years.
>>=20
>> https://tools.ietf.org/html/rfc2543
>>=20
>> /O
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Wed Jul 13 07:57:28 2016
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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Wed, 13 Jul 2016 16:57:19 +0200
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To: Brett Tate <brett@broadsoft.com>
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Cc: sipcore@ietf.org, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> On 13 Jul 2016, at 16:28, Brett Tate <brett@broadsoft.com> wrote:
>=20
> Hi,
>=20
> RFC 2543 is version 2.0.
I started googling for version 1.0 and it seems half of the Internet =
believes that 2543 is version 1.0.

"Session Initiation Protocol was first introduced in March 1999 by IETF =
RFC 2543. This RFC (Request for Comments) was the original core =
specification and was obsolete by IETF RFC 3261 in June 2002. These =
RFC=E2=80=99s are more commonly known as SIP 1.0 (2543) and SIP 2.0 =
(3261). There are many related RFCs that are summarized at the end of =
this document.=E2=80=9D

http://www.whichvoip.com/voip/protocols/sip.htm


=E2=80=9Cthe protocol was standardized as RFC 2543 in 1999 (SIP 1.0). In =
November 2000, SIP was accepted as a 3GPP signaling protocol and =
permanent element of the IP Multimedia Subsystem (IMS) architecture for =
IP-based streaming multimedia services in cellular systems. As of 2014, =
the latest version (SIP 2.0) of the specification is RFC 3261, published =
in June 2002,[2] with extensions and clarifications since then.[3]=E2=80=9D=


https://en.wikipedia.org/wiki/Session_Initiation_Protocol

As Brett says, RFC 2543 is indeed version 2.0:
"4.3.1 SIP Version

   Both request and response messages include the version of SIP in use,
   and follow [H3.1] (with HTTP replaced by SIP, and HTTP/1.1 replaced
   by SIP/2.0) regarding version ordering, compliance requirements, and
   upgrading of version numbers. To be compliant with this
   specification, applications sending SIP messages MUST include a SIP-
   Version of "SIP/2.0=E2=80=9D.=E2=80=9D

Realising I=E2=80=99ve been fooled by folklore I am still searching for =
version 1.0. Any pointers?

If we have 2.0 according to 2543 and 2.0 according to 3261 it=E2=80=99s =
going to be hard to get
rid of 2543.

I would have no problem hearing that story - why did we keep the 2.0 =
version in 3261 - while gently
SIPping a german beer in Berlin next week :-)

/O

>=20
> Are you wanting to declare RFC 2543 as historic or whatever defined =
version
> 1.0 as historic?
>=20
> What would be the benefit of declaring RFC 2543 (or version 1.0) as
> historic?  For instance, are you wanting to remove some of the =
backward
> compatibility related mandates contained within RFC 3261?
>=20
> Thanks,
> Brett
>=20
>> -----Original Message-----
>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E.
>> Johansson
>> Sent: Wednesday, July 13, 2016 10:15 AM
>> To: sipcore@ietf.org
>> Cc: Olle E Johansson
>> Subject: [sipcore] Declaring SIP 1.0 historic?
>>=20
>> Friends,
>>=20
>> Is it time to declare SIP 1.0 historic? I haven=E2=80=99t met a SIP =
1.0
>> implementation in use for many, many years.
>>=20
>> https://tools.ietf.org/html/rfc2543
>>=20
>> /O
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore


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To: "Olle E. Johansson" <oej@edvina.net>, Brett Tate <brett@broadsoft.com>
Thread-Topic: [sipcore] Declaring SIP 1.0 historic?
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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From nobody Wed Jul 13 08:36:12 2016
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Date: Wed, 13 Jul 2016 11:36:08 -0400
From: Jonathan Lennox <lennox@cs.columbia.edu>
To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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On Wednesday, July 13 2016, "Olle E. Johansson" wrote to "Brett Tate, s=
ipcore@ietf.org, Olle E Johansson" saying:

> Realising I=E2=80=99ve been fooled by folklore I am still searching f=
or version 1.0. Any pointers=3F

There never was a SIP/1.0.  The 2.0 number was chosen to make sure that=
 SIP
wouldn't be confused with HTTP/1.x on the wire, and also in a vague (an=
d it
turned out, vain) hope that it'd be possible to run SIP through HTTP
proxies.

> If we have 2.0 according to 2543 and 2.0 according to 3261 it=E2=80=99=
s going to be hard to get
> rid of 2543.

It'd be possible to deprecate the old 2543-compatible transaction match=
ing
and source routing.  I don't know how much of a gain that is for
implementors, though.
=20
> I would have no problem hearing that story - why did we keep the 2.0 =
version in 3261 - while gently
> SIPping a german beer in Berlin next week :-)

We kept it because SIP routing rules (notably forking) mean that there'=
s no
good way to try to send a request with a newer version, and then if it
fails, fall back to an older version.  This is why all backward
compatibility in SIP has to be single-pass.

--=20
Jonathan Lennox
lennox@cs.columbia.edu


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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> Yes, please. I would be happy to remove the
> RFC 2543 transaction matching.

I find RFC 3261 section 17.2.3 transaction matching to be overly lenient
when using the magic cookie.

For instance, the matching rules don't even include Call-ID and CSeq.  Thus
if you follow RFC 3261 section 17.2.3, you have to allow requests containing
different Call-ID and CSeq to match because of the magic cookie flexibility.


From nobody Wed Jul 13 10:01:28 2016
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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On 7/13/16 10:57 AM, Olle E. Johansson wrote:

> Realising Iâ€™ve been fooled by folklore I am still searching for version 1.0. Any pointers?

I don't know about documents for it. But I bet Henning was involved. So 
he is where I would start.

> If we have 2.0 according to 2543 and 2.0 according to 3261 itâ€™s going to be hard to get
> rid of 2543.
>
> I would have no problem hearing that story - why did we keep the 2.0 version in 3261 - while gently
> SIPping a german beer in Berlin next week :-)

I became involved just a bit before 3261. I don't recall, but I expect 
that it was because there were deployments of 2543 and there is no good 
way to be agile wrt version number.

	Thanks,
	Paul


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From: Brett Tate <brett@broadsoft.com>
References: <1EE2D149-1E1A-49DA-ACB8-625FEC4E4508@edvina.net> <76f2fd5e65bfa393281d682d23e18f11@mail.gmail.com>	<C88324E1-C2D4-4E9A-8B99-1D86663074BD@edvina.net> <22406.24552.369096.256493@compute06.cs.columbia.edu>
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Date: Wed, 13 Jul 2016 13:41:37 -0400
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To: Jonathan Lennox <lennox@cs.columbia.edu>, sipcore@ietf.org
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> > Realising I=E2=80=99ve been fooled by folklore I am still
> > searching for version 1.0. Any pointers?
>
> There never was a SIP/1.0.  The 2.0 number was chosen to
> make sure that SIP wouldn't be confused with HTTP/1.x on
> the wire, and also in a vague (and it turned out, vain)
> hope that it'd be possible to run SIP through HTTP proxies.

Thanks for the information.  I had assumed that there was a version 1.0
somewhere (for instance maybe based upon a Henning Schulzrinne document or
RFC 2543 section F's acknowledgement of [37,38]).


> > If we have 2.0 according to 2543 and 2.0 according to
> > 3261 it=E2=80=99s going to be hard to get rid of 2543.
>
> It'd be possible to deprecate the old 2543-compatible
> transaction matching and source routing.  I don't know
> how much of a gain that is for implementors, though.

I don't think there is much gain concerning transaction matching or strict
routing.  The desire to deprecate occasionally arises concerning To and Fro=
m
URI mandates.

For instance, RFC 3261 section 12.2.1.1 indicates an expectation that the
"mandatory reflection of the original To and From URI in mid-dialog request=
s
will be deprecated in a subsequent revision".  I'm not sure why; however
some vendors and operators want to allow the To/From URI to change without
any restrictions (e.g. they don't want to be restricted by RFC 4916).


From nobody Wed Jul 13 10:42:38 2016
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From: Adam Roach <adam@nostrum.com>
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Date: Wed, 13 Jul 2016 12:42:16 -0500
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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On 7/13/16 10:36, Jonathan Lennox wrote:
> On Wednesday, July 13 2016, "Olle E. Johansson" wrote to "Brett Tate, sipcore@ietf.org, Olle E Johansson" saying:
>
>> Realising Iâ€™ve been fooled by folklore I am still searching for version 1.0. Any pointers?
> There never was a SIP/1.0.

That doesn't quite match my memory.

I beleive there were early implementations of Handly/Schooler's SIP/1.0 
spec 
<http://www.cs.columbia.edu/sip/drafts/mmusic/draft-ietf-mmusic-sip-00.pdf>in 
one or both of the UCL/Caltech labs. This was subsequently merged with 
Schulzrinne's Simple Conference Invitation Protocol (SCIP/1.0) 
<https://tools.ietf.org/html/draft-ietf-mmusic-scip-00>, and the merged 
version was dubbed SIP/2.0.

Poking around, this paper's introduction matches my personal 
recollection of the history:

http://www.jocm.us/uploadfile/2013/0416/20130416114634804.pdf

/a


From nobody Wed Jul 13 12:10:11 2016
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Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
> I just looked back at my private archive. The discussions took place in 
> May and July of 2010 on the sipcore list. (I haven't looked for these in 
> the archives. These days I can get a URL into the archives from my copy 
> of a message, but not with ones from that far back.)
>
> I have attached a couple of messages as hooks into those discussions

The messages with subject "Revised draft-jones-sip-options-ping-02.txt"
are (currently) indexed on
https://www.ietf.org/mail-archive/web/sipcore/current/thrd9.html The
messages with subjects "draft-jones-sip-options-ping-01" and "The
predictive value of OPTIONS" are indexed on
https://www.ietf.org/mail-archive/web/sipcore/current/thrd9.html
https://www.ietf.org/mail-archive/web/sipcore/current/thrd10.html

Dale


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Brett Tate <brett@broadsoft.com> writes:
>> I don't think using OPTIONS/NOTIFY v.s. REGISTER
>> would change anything on this front. Roman probably
>> refers more to the overall processing cost.
>
> Because refreshing a registration typically involves updating persistent
> information, performing authentication, et cetera, the impacts a REGISTER
> can be significantly higher than that of OPTIONS.  This is the reason why
> some vendors prefer receiving OPTIONS instead of REGISTER when sent
> frequently as a workaround for NAT involvement.

Also, knowledge of the expirations of the registrations must be
maintained accurately, while a keepalive only has to be sent frequently
enough.  For instance, consider if the maximum keepalive interval is 60
secs, and the proxy sends an OPTIONS every 30 secs.  The proxy needs to
keep a database of keepalive timers.  But if it goes down, the standby
proxy doesn't need to know when each keepalive expires, it needs to send
an OPTIONS to every target within the next 30 secs.

Dale


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Subject: Re: [sipcore] Very rough draft for remaining Happy Eyeballs work
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Starting to work on this message:

"Olle E. Johansson" <oej@edvina.net> writes:
> I don't understand why you let devices break the order implied by RFC
> 3263. The order is well defined there. Do you mean that for a given
> HOST you may contact the targets in any order (or preferred:
> simultaneously).

The idea is not only to allow devices to contact the targets in a
different order, but to give devices guidance on the best order to
contact targets.  This is useful in at least the case where a particular
target is known to be non-responsive; you want to postpone that target
until after you've tried targets that aren't know to be non-responsive.

And ideal system would be to be able to contact all targets
simultaneously, but there's no general way to prevent that from creating
multiple dialogs.

Dale


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To: "Olle E. Johansson" <oej@edvina.net>, Brett Tate <brett@broadsoft.com>
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From: "Vijay K. Gurbani" <vkg@bell-labs.com>
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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On 07/13/2016 09:57 AM, Olle E. Johansson wrote:
> I started googling for version 1.0 and it seems half of the Internet
> believes that 2543 is version 1.0.

A brief history of SIP (including 1.0) is provided in [1].  The PDF for
this paper is available for free download (open access) at [2].

Have fun spelunking the origin of SIP.

[1] Salman Abdul Baset, Vijay K. Gurbani, Alan B. Johnston,
  Hadriel Kaplan, Brian Rosen and Jonathan D. Rosenberg , "The Session
  Initiation Protocol (SIP): An evolutionary study", In Journal of
  Communications (JCM), pp. 89-105, Vol. 7., No. 2, February 2012

[2] 
http://ojs.academypublisher.com/index.php/jcm/article/download/jcm070289105/4295

Cheers,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Nokia Networks
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
Email: vkg@bell-labs.com / vijay.gurbani@nokia-bell-labs.com
Web: http://ect.bell-labs.com/who/vkg/  | Calendar: http://goo.gl/x3Ogq


From nobody Wed Jul 13 12:55:25 2016
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> On 13 Jul 2016, at 21:50, Vijay K. Gurbani <vkg@bell-labs.com> wrote:
>=20
> On 07/13/2016 09:57 AM, Olle E. Johansson wrote:
>> I started googling for version 1.0 and it seems half of the Internet
>> believes that 2543 is version 1.0.
>=20
> A brief history of SIP (including 1.0) is provided in [1].  The PDF =
for
> this paper is available for free download (open access) at [2].
>=20
> Have fun spelunking the origin of SIP.
>=20
> [1] Salman Abdul Baset, Vijay K. Gurbani, Alan B. Johnston,
> Hadriel Kaplan, Brian Rosen and Jonathan D. Rosenberg , "The Session
> Initiation Protocol (SIP): An evolutionary study", In Journal of
> Communications (JCM), pp. 89-105, Vol. 7., No. 2, February 2012
>=20
> [2] =
http://ojs.academypublisher.com/index.php/jcm/article/download/jcm07028910=
5/4295

Thank you to everyone that have been responding to my spelunking!

I will read up on this. (And update my training materials).
Always great to learn.

/O=


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Subject: Re: [sipcore] Very rough draft for remaining Happy Eyeballs work
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> On 13 Jul 2016, at 21:26, Dale R. Worley <worley@ariadne.com> wrote:
>=20
> Starting to work on this message:
>=20
> "Olle E. Johansson" <oej@edvina.net> writes:
>> I don't understand why you let devices break the order implied by RFC
>> 3263. The order is well defined there. Do you mean that for a given
>> HOST you may contact the targets in any order (or preferred:
>> simultaneously).
>=20
> The idea is not only to allow devices to contact the targets in a
> different order, but to give devices guidance on the best order to
> contact targets.  This is useful in at least the case where a =
particular
> target is known to be non-responsive; you want to postpone that target
> until after you've tried targets that aren't know to be =
non-responsive.
Hmm. That makes sense, but is that not allowed in the RFCs?=20
(need to re-read, as always)
>=20
> And ideal system would be to be able to contact all targets
> simultaneously, but there's no general way to prevent that from =
creating
> multiple dialogs.
You have to be clear here. THere=E2=80=99s no way to preven that from
creating multiple dialogs in UDP. In TCP you can of course prevent
multiple dialogs by not sending anything once a connection is open,
just close it without any SIP message.

/O=


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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> On 13 Jul 2016, at 19:41, Brett Tate <brett@broadsoft.com> wrote:
>=20
>>> Realising I=E2=80=99ve been fooled by folklore I am still
>>> searching for version 1.0. Any pointers?
>>=20
>> There never was a SIP/1.0.  The 2.0 number was chosen to
>> make sure that SIP wouldn't be confused with HTTP/1.x on
>> the wire, and also in a vague (and it turned out, vain)
>> hope that it'd be possible to run SIP through HTTP proxies.
>=20
> Thanks for the information.  I had assumed that there was a version =
1.0
> somewhere (for instance maybe based upon a Henning Schulzrinne =
document or
> RFC 2543 section F's acknowledgement of [37,38]).
>=20
>=20
>>> If we have 2.0 according to 2543 and 2.0 according to
>>> 3261 it=E2=80=99s going to be hard to get rid of 2543.
>>=20
>> It'd be possible to deprecate the old 2543-compatible
>> transaction matching and source routing.  I don't know
>> how much of a gain that is for implementors, though.
>=20
> I don't think there is much gain concerning transaction matching or =
strict
> routing.  The desire to deprecate occasionally arises concerning To =
and From
> URI mandates.
>=20
> For instance, RFC 3261 section 12.2.1.1 indicates an expectation that =
the
> "mandatory reflection of the original To and =46rom URI in mid-dialog =
requests
> will be deprecated in a subsequent revision".  I'm not sure why; =
however
> some vendors and operators want to allow the To/=46rom URI to change =
without
> any restrictions (e.g. they don't want to be restricted by RFC 4916).

There are many call transfer scenarios involving b2bua=E2=80=99s that =
need
to modify/update caller IDs and there are too many implementations
pre-4916 out there.

/O=


From nobody Wed Jul 13 13:12:26 2016
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To: Jonathan Lennox <lennox@cs.columbia.edu>, "Olle E. Johansson" <oej@edvina.net>
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From: "Vijay K. Gurbani" <vkg@bell-labs.com>
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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On 07/13/2016 10:36 AM, Jonathan Lennox wrote:
>> I would have no problem hearing that story - why did we keep the
>> 2.0 version in 3261 - while gently SIPping a german beer in Berlin
>> next week :-)
>
> We kept it because SIP routing rules (notably forking) mean that
> there's no good way to try to send a request with a newer version,
> and then if it fails, fall back to an older version.  This is why all
> backward compatibility in SIP has to be single-pass.

Right, that matches my recollection as well.  When we were designing
what eventually became rfc3261, a prime directive was do no harm and
be backwardly compatible.  Because rfc2543 was deployed and being used,
a forklift upgrade by changing the protocol version was not looked at
kindly.

Almost a decade ago, a brief email exchange surfaced to entertain the
idea of a SIP/3.0 [1].  It was ignominiously short lived :-)

[1] https://www.ietf.org/mail-archive/web/sip/current/thrd19.html#18421

Cheers,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Nokia Networks
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
Email: vkg@bell-labs.com / vijay.gurbani@nokia-bell-labs.com
Web: http://ect.bell-labs.com/who/vkg/  | Calendar: http://goo.gl/x3Ogq


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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> On 13 Jul 2016, at 22:12, Vijay K. Gurbani <vkg@bell-labs.com> wrote:
>=20
> On 07/13/2016 10:36 AM, Jonathan Lennox wrote:
>>> I would have no problem hearing that story - why did we keep the
>>> 2.0 version in 3261 - while gently SIPping a german beer in Berlin
>>> next week :-)
>>=20
>> We kept it because SIP routing rules (notably forking) mean that
>> there's no good way to try to send a request with a newer version,
>> and then if it fails, fall back to an older version.  This is why all
>> backward compatibility in SIP has to be single-pass.
>=20
> Right, that matches my recollection as well.  When we were designing
> what eventually became rfc3261, a prime directive was do no harm and
> be backwardly compatible.  Because rfc2543 was deployed and being =
used,
> a forklift upgrade by changing the protocol version was not looked at
> kindly.
>=20
> Almost a decade ago, a brief email exchange surfaced to entertain the
> idea of a SIP/3.0 [1].  It was ignominiously short lived :-)
>=20
> [1] =
https://www.ietf.org/mail-archive/web/sip/current/thrd19.html#18421

Just love this quote by Dale Worley:

"SIP 3.0 is like fully participatory democracy, the ultimate form of
communism, and the earthly Kingdom of God -- a thorough reform of what
presently exists that will remove all of its flaws.

Similar to those other three projects, there is no WG timeline yet for
SIP 3.0 development.=E2=80=9D

Let=E2=80=99s start SIP 4.0 :-)

/O=


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From: Roman Shpount <roman@telurix.com>
Date: Wed, 13 Jul 2016 16:44:52 -0400
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--001a113ed0d019d8b405378a7589
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On Wed, Jul 6, 2016 at 5:54 PM, Dale R. Worley <worley@ariadne.com> wrote:

> Thanks for the information!  I've added your discussion to the rough
> draft.
>
> Roman Shpount <roman@telurix.com> writes:
> > Few observations:
> > [...]
> > 3. For server, the most efficient way to keep the NAT hole open is to
> send
> > OPTIONS or NOTIFY requests to the client. This is much more efficient
> then
> > registrations with small timeouts.
> >
> > 4. Registration server which has an in-memory database for current
> > registrations can be fairly efficient in reducing number of back-end DB
> > updates due to registrations and subscriptions with small timeouts. It is
> > not going to be stateless, but its state does not need to be replicated
> and
> > would automatically recover on the stand-by server on the next
> registration
> > or subscription request from the client.
>
> I think what you are saying here is that if the server sends
> OPTIONS/NOTIFY for keepalive, rather than making the client send
> frequent re-REGISTERS, then the server doesn't need to record the timers
> for the keepalives in persistent storage -- if the stand-by server is
> activated, it simply sends keepalives to all registrations that are
> recorded as having keepalive service.  In the long run, this requires no
> more processing by the registration server and far less demand on the
> persistent storage.
>
> Is that correct?
>

This is correct. Since OPTIONS/NOTIFY do not update the registration state
they typically require less resources then forcing the client to
re-register at the small interval. Registrations typically require
authentication and cause the shared storage update for the expiration time.
OPTIONS/NOTIFY sent from server require not of this.

>
> > 5. One of the big problems with encrypted SIP traffic is that it gets
> > modified by ALG. For a lot of hosted PBX providers avoiding the need to
> > troubleshoot customer routers is a bigger incentive to deploy TLS then
> user
> > privacy.
>
> If I understand you correctly, hosted PBX providers favor TLS to stop
> routers from modifying SIP messages.  But ALGs terminate TLS ... and
> then modify the SIP messages.
>
>
One of the big problems with deploying SIP on customer premises are SIP ALG
implemented by the consumer routers. Such ALG are typically broken and
cause SIP call setup failures or call drops. If client equipment is using
SIP over TLS, SIP ALG in routers cannot affect the signaling and SIP
message pass the router unmolested. Because of this, SIP calls over TLS are
much more stable which results in big reduction of customer problems and
associated support calls. This reduction in support burden is a much bigger
incentive for service providers to deploy encryption for SIP then customer
privacy (which, in practice, is almost never their concern).
_____________
Roman Shpount

--001a113ed0d019d8b405378a7589
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><div><div class=3D"gmail_si=
gnature" data-smartmail=3D"gmail_signature">On Wed, Jul 6, 2016 at 5:54 PM,=
 Dale R. Worley <span dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com"=
 target=3D"_blank">worley@ariadne.com</a>&gt;</span> wrote:<br></div></div>=
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex=
">Thanks for the information!=C2=A0 I&#39;ve added your discussion to the r=
ough<br>
draft.<br>
<br>
Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.com</a=
>&gt; writes:<br>
&gt; Few observations:<br>
&gt; [...]<br>
<span class=3D"">&gt; 3. For server, the most efficient way to keep the NAT=
 hole open is to send<br>
&gt; OPTIONS or NOTIFY requests to the client. This is much more efficient =
then<br>
&gt; registrations with small timeouts.<br>
&gt;<br>
&gt; 4. Registration server which has an in-memory database for current<br>
&gt; registrations can be fairly efficient in reducing number of back-end D=
B<br>
&gt; updates due to registrations and subscriptions with small timeouts. It=
 is<br>
&gt; not going to be stateless, but its state does not need to be replicate=
d and<br>
&gt; would automatically recover on the stand-by server on the next registr=
ation<br>
&gt; or subscription request from the client.<br>
<br>
</span>I think what you are saying here is that if the server sends<br>
OPTIONS/NOTIFY for keepalive, rather than making the client send<br>
frequent re-REGISTERS, then the server doesn&#39;t need to record the timer=
s<br>
for the keepalives in persistent storage -- if the stand-by server is<br>
activated, it simply sends keepalives to all registrations that are<br>
recorded as having keepalive service.=C2=A0 In the long run, this requires =
no<br>
more processing by the registration server and far less demand on the<br>
persistent storage.<br>
<br>
Is that correct?<br></blockquote><div><br></div><div>This is correct. Since=
 OPTIONS/NOTIFY do not update the registration state they typically require=
 less resources then forcing the client to re-register at the small interva=
l. Registrations typically require authentication and cause the shared stor=
age update for the expiration time. OPTIONS/NOTIFY sent from server require=
 not of this.</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<span class=3D""><br>
&gt; 5. One of the big problems with encrypted SIP traffic is that it gets<=
br>
&gt; modified by ALG. For a lot of hosted PBX providers avoiding the need t=
o<br>
&gt; troubleshoot customer routers is a bigger incentive to deploy TLS then=
 user<br>
&gt; privacy.<br>
<br>
</span>If I understand you correctly, hosted PBX providers favor TLS to sto=
p<br>
routers from modifying SIP messages.=C2=A0 But ALGs terminate TLS ... and<b=
r>
then modify the SIP messages.<br>
<span class=3D""><font color=3D"#888888"><br></font></span></blockquote><di=
v><br></div><div>One of the big problems with deploying SIP on customer pre=
mises are SIP ALG implemented by the consumer routers. Such ALG are typical=
ly broken and cause SIP call setup failures or call drops. If client equipm=
ent is using SIP over TLS, SIP ALG in routers cannot affect the signaling a=
nd SIP message pass the router unmolested. Because of this, SIP calls over =
TLS are much more stable which results in big reduction of customer problem=
s and associated support calls. This reduction in support burden is a much =
bigger incentive for service providers to deploy encryption for SIP then cu=
stomer privacy (which, in practice, is almost never their concern).=C2=A0</=
div><div><div class=3D"gmail_signature" data-smartmail=3D"gmail_signature">=
_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></di=
v>

--001a113ed0d019d8b405378a7589--


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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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On Wed, Jul 6, 2016 at 6:01 PM, Asveren, Tolga <tasveren@sonusnet.com>
wrote:

> > then the server doesn't need to record the timers for the keepalives in
> > persistent storage
> I don't think using OPTIONS/NOTIFY v.s. REGISTER would change anything on
> this front. Roman probably refers more to the overall processing cost.
>
>
There is processing cost since OPTIONS/NOTIFY typically are not
authenticated. There is state storage cost, since registration state
typically needs to be stored in some sort of shared storage and recovered
in case of proxy failure. OPTIONS/NOTIFY can be send and processed based on
in-memory state and do not need to be recovered.

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure" data-smartmail=3D"gmail_signature">On Wed, Jul 6, 2016 at 6:01 PM, Asv=
eren, Tolga <span dir=3D"ltr">&lt;<a href=3D"mailto:tasveren@sonusnet.com" =
target=3D"_blank">tasveren@sonusnet.com</a>&gt;</span> wrote:<br></div></di=
v><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1=
ex"><span class=3D"">&gt; then the server doesn&#39;t need to record the ti=
mers for the keepalives in<br>
&gt; persistent storage<br>
</span>I don&#39;t think using OPTIONS/NOTIFY v.s. REGISTER would change an=
ything on this front. Roman probably refers more to the overall processing =
cost.<br><br></blockquote><div><br></div><div>There is processing cost sinc=
e OPTIONS/NOTIFY typically are not authenticated. There is state storage co=
st, since registration state typically needs to be stored in some sort of s=
hared storage and recovered in case of proxy failure. OPTIONS/NOTIFY can be=
 send and processed based on in-memory state and do not need to be recovere=
d.</div><div><br></div><div>Regards,</div><div><div class=3D"gmail_signatur=
e" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</div><=
/div><div>=C2=A0</div></div><br></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, "Asveren, Tolga" <tasveren@sonusnet.com>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde
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From: Roman Shpount <roman@telurix.com>
Date: Wed, 13 Jul 2016 17:00:36 -0400
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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On Wed, Jul 13, 2016 at 4:53 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>
> REGISTER requests used for keep-alive are not always authenticated,
> because they are not always forwarded to the registrar to begin with. They
> are terminated by an edge/outbound proxy.
>
>
>
When we implemented keep-alive using REGISTER we used an in memory DB in
the edge proxy to store registration and authentication information. We
used this DB to aggregate frequent REGISTER requests on the edge proxy into
a much less frequent requests to the Registration proxy. So, this is
implementable but a lot more work.
_____________
Roman Shpount

--94eb2c1144f85f80b305378aad1b
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" data-smartmail=3D"gmail_signature">On Wed, Jul 13, 2016 at 4:53 PM, Ch=
rister Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@e=
ricsson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span=
> wrote:<br></div></div><div class=3D"gmail_quote"><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><br></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">REGISTER requests used for keep-alive are no=
t always authenticated, because they are not always forwarded to the regist=
rar to begin
 with. They are terminated by an edge/outbound proxy.<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>When we implemented keep-alive using REGISTER we used an in memory DB in t=
he edge proxy to store registration and authentication information. We used=
 this DB to aggregate frequent REGISTER requests on the edge proxy into a m=
uch less frequent requests to the Registration proxy. So, this is implement=
able but a lot more work.</div><div><div class=3D"gmail_signature" data-sma=
rtmail=3D"gmail_signature">_____________<br>Roman Shpount</div></div><div>=
=C2=A0</div></div></div></div>

--94eb2c1144f85f80b305378aad1b--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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From nobody Wed Jul 13 14:44:51 2016
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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On 7/13/16 4:53 PM, Christer Holmberg wrote:
> Hi,
>
> REGISTER requests used for keep-alive are not always authenticated,
> because they are not always forwarded to the registrar to begin with.
> They are terminated by an edge/outbound proxy.

I know this is done, but I have always thought it to be broken.

The response to REGISTER is supposed to include *all* contacts that have 
been registered, not just those that were offered in *this* REGISTER. 
Also, the expiration time is supposed to reflect what the registrar 
thinks it is.

If the request is short-circuited in a proxy, even a stateful proxy, the 
response likely won't have that property.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Hi,

>> REGISTER requests used for keep-alive are not always authenticated,=20
>> because they are not always forwarded to the registrar to begin with.
>> They are terminated by an edge/outbound proxy.
>
> I know this is done, but I have always thought it to be broken.
>
> The response to REGISTER is supposed to include *all* contacts that have =
been registered, not just those that were offered in *this* REGISTER.=20

The edge proxy can store that information from the previous REGISTER respon=
se it received from the registrar, and include it in the REGISTER responses=
 it triggers itself. It anyway needs to keep track of the registration, to =
make sure some REGISTER requests *are* forwarded towards the registrar (in =
order for the registration to not expire).=20

> Also, the expiration time is supposed to reflect what the registrar think=
s it is.
>
> If the request is short-circuited in a proxy, even a stateful proxy, the =
response likely won't have that property.

But it works :)

Regards,

Christer


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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On 7/13/16 6:31 PM, Christer Holmberg wrote:
> Hi,
>
>>> REGISTER requests used for keep-alive are not always authenticated,
>>> because they are not always forwarded to the registrar to begin with.
>>> They are terminated by an edge/outbound proxy.
>>
>> I know this is done, but I have always thought it to be broken.
>>
>> The response to REGISTER is supposed to include *all* contacts that have been registered, not just those that were offered in *this* REGISTER.
>
> The edge proxy can store that information from the previous REGISTER response it received from the registrar, and include it in the REGISTER responses it triggers itself. It anyway needs to keep track of the registration, to make sure some REGISTER requests *are* forwarded towards the registrar (in order for the registration to not expire).

That only works if the proxy is in the path for *all* REGISTER requests 
for that AoR. Otherwise there can be another contact, added by a UA 
through a different proxy. And then *this* UA won't get that info until 
the proxy decides to pass a request on to the registrar.

>> Also, the expiration time is supposed to reflect what the registrar thinks it is.
>>
>> If the request is short-circuited in a proxy, even a stateful proxy, the response likely won't have that property.
>
> But it works :)

For *some* definitions of "works". There are some things that won't 
work. You may judge that those don't matter, but somebody else may not.

	Thanks,
	Paul


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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "Olle E. Johansson" <oej@edvina.net>, "Gurbani, Vijay (Nokia - US)" <vijay.gurbani@nokia-bell-labs.com>
Thread-Topic: [sipcore] Declaring SIP 1.0 historic?
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> On 14 Jul 2016, at 03:00, Drage, Keith (Nokia - GB) =
<keith.drage@nokia.com> wrote:
>=20
> So can you be clear on where you are trying to get to.
>=20
> 1)	deprecate SIP 1.0 --> but it never existed.
>=20
> 2)	deprecate RFC 2543 --> but this RFC is historic and therefore is =
of that status.
Where can I see that status? I only see =E2=80=9Cstandard track=E2=80=9D =
and =E2=80=9Cproposed standard=E2=80=9D.
https://www.rfc-editor.org/search/rfc_search_detail.php

I don=E2=80=99t agree with you here. It=E2=80=99s obsoleted by never =
declared historic as far as I
understand. What would happen if we did?
>=20
> 3)	deprecate mechanisms for interoperability of RFC 3261 with RFC =
2543?
Maybe that would be the effect. Can we try to figure out what would =
happen?
One clear benefit highlighted seems to be to get rid of the restrictions =
on=20
changes of the from/to headers. Now, how would that work with STIR?

/O
>=20
> Regards
>=20
> Keith Drage
>=20
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E. =
Johansson
> Sent: 13 July 2016 21:28
> To: Gurbani, Vijay (Nokia - US)
> Cc: sipcore@ietf.org; Olle E Johansson
> Subject: Re: [sipcore] Declaring SIP 1.0 historic?
>=20
>=20
>> On 13 Jul 2016, at 22:12, Vijay K. Gurbani <vkg@bell-labs.com> wrote:
>>=20
>> On 07/13/2016 10:36 AM, Jonathan Lennox wrote:
>>>> I would have no problem hearing that story - why did we keep the
>>>> 2.0 version in 3261 - while gently SIPping a german beer in Berlin=20=

>>>> next week :-)
>>>=20
>>> We kept it because SIP routing rules (notably forking) mean that=20
>>> there's no good way to try to send a request with a newer version,=20=

>>> and then if it fails, fall back to an older version.  This is why =
all=20
>>> backward compatibility in SIP has to be single-pass.
>>=20
>> Right, that matches my recollection as well.  When we were designing=20=

>> what eventually became rfc3261, a prime directive was do no harm and=20=

>> be backwardly compatible.  Because rfc2543 was deployed and being=20
>> used, a forklift upgrade by changing the protocol version was not=20
>> looked at kindly.
>>=20
>> Almost a decade ago, a brief email exchange surfaced to entertain the=20=

>> idea of a SIP/3.0 [1].  It was ignominiously short lived :-)
>>=20
>> [1]=20
>> https://www.ietf.org/mail-archive/web/sip/current/thrd19.html#18421
>=20
> Just love this quote by Dale Worley:
>=20
> "SIP 3.0 is like fully participatory democracy, the ultimate form of =
communism, and the earthly Kingdom of God -- a thorough reform of what =
presently exists that will remove all of its flaws.
>=20
> Similar to those other three projects, there is no WG timeline yet for =
SIP 3.0 development.=E2=80=9D
>=20
> Let=E2=80=99s start SIP 4.0 :-)
>=20
> /O
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Thu Jul 14 00:27:31 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Date: Thu, 14 Jul 2016 07:26:52 +0000
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Hi,

>>>>REGISTER requests used for keep-alive are not always authenticated,
>>>> because they are not always forwarded to the registrar to begin with.
>>>> They are terminated by an edge/outbound proxy.
>>>
>>> I know this is done, but I have always thought it to be broken.
>>>
>>> The response to REGISTER is supposed to include *all* contacts that
>>>have been registered, not just those that were offered in *this*
>>>REGISTER.
>>
>> The edge proxy can store that information from the previous REGISTER
>>response it received from the registrar, and include it in the REGISTER
>>responses it triggers itself. It anyway needs to keep track of the
>>registration, to make sure some REGISTER requests *are* forwarded
>>towards the registrar (in order for the registration to not expire).
>
>That only works if the proxy is in the path for *all* REGISTER requests
>for that AoR. Otherwise there can be another contact, added by a UA
>through a different proxy. And then *this* UA won't get that info until
>the proxy decides to pass a request on to the registrar.

Sure, but that is the case even if REGISTERs aren't used for keep-alives.

>>> Also, the expiration time is supposed to reflect what the registrar
>>>thinks it is.
>>>
>>> If the request is short-circuited in a proxy, even a stateful proxy,
>>>the response likely won't have that property.
>>
>> But it works :)
>
>For *some* definitions of "works". There are some things that won't
>work. You may judge that those don't matter, but somebody else may not.


Sure.

And, I didn=B9t mean to say this is the best way of doing things - I was
only explaining how many do use REGISTER for keep-olives.

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Date: Thu, 14 Jul 2016 07:30:33 +0000
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From nobody Thu Jul 14 00:34:22 2016
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Even using OPTIONs for keepalives may generate problems considering a =
SIP implementation
that use OPTIONs for discovery.

In the case of Asterisk OPTIONs are answered blindly. But INVITEs are =
authenticated and
generated depending upon whether we have a configuration for that =
particular user.

This means that OPTIONs and INVITEs have different responses. If the UA =
use the
OPTIONs answer (including SDP) for discovery, the result is all wrong=E2=80=
=A6

For keepalives - I would use what=E2=80=99s describe in section 4.4 of =
RFC 5626, regardless of
if I follow outbound or not. CRLF or STUN.

/O=


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Date: Thu, 14 Jul 2016 07:43:04 +0000
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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>...
>
>
>For keepalives - I would use what=B9s describe in section 4.4 of RFC 5626,
>regardless of
>if I follow outbound or not. CRLF or STUN.

The negotiation and usage of those, without outbound, is described in RFC
6223.

https://tools.ietf.org/rfc/rfc6223.txt


Regards,

Christer


From nobody Thu Jul 14 00:49:58 2016
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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> On 14 Jul 2016, at 09:43, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>> ...
>>=20
>>=20
>> For keepalives - I would use what=C4=85s describe in section 4.4 of =
RFC 5626,
>> regardless of
>> if I follow outbound or not. CRLF or STUN.
>=20
> The negotiation and usage of those, without outbound, is described in =
RFC
> 6223.
>=20
> https://tools.ietf.org/rfc/rfc6223.txt
>=20
>=20
Thank you for that reference, I forgot about that.

We do need to update The Hitchhiker=E2=80=99s guide to SIP. The extended =
version :-)

/O


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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On 7/14/16 3:30 AM, Christer Holmberg wrote:
> keep-alives, that is :)

I think I prefer keep-olives :-)

> On 14/07/16 10:26, "sipcore on behalf of Christer Holmberg"
> <sipcore-bounces@ietf.org on behalf of christer.holmberg@ericsson.com>
> wrote:
>
>> Hi,
>>
>>>>>> REGISTER requests used for keep-alive are not always authenticated,
>>>>>> because they are not always forwarded to the registrar to begin with.
>>>>>> They are terminated by an edge/outbound proxy.
>>>>>
>>>>> I know this is done, but I have always thought it to be broken.
>>>>>
>>>>> The response to REGISTER is supposed to include *all* contacts that
>>>>> have been registered, not just those that were offered in *this*
>>>>> REGISTER.
>>>>
>>>> The edge proxy can store that information from the previous REGISTER
>>>> response it received from the registrar, and include it in the REGISTER
>>>> responses it triggers itself. It anyway needs to keep track of the
>>>> registration, to make sure some REGISTER requests *are* forwarded
>>>> towards the registrar (in order for the registration to not expire).
>>>
>>> That only works if the proxy is in the path for *all* REGISTER requests
>>> for that AoR. Otherwise there can be another contact, added by a UA
>>> through a different proxy. And then *this* UA won't get that info until
>>> the proxy decides to pass a request on to the registrar.
>>
>> Sure, but that is the case even if REGISTERs aren't used for keep-alives.
>>
>>>>> Also, the expiration time is supposed to reflect what the registrar
>>>>> thinks it is.
>>>>>
>>>>> If the request is short-circuited in a proxy, even a stateful proxy,
>>>>> the response likely won't have that property.
>>>>
>>>> But it works :)
>>>
>>> For *some* definitions of "works". There are some things that won't
>>> work. You may judge that those don't matter, but somebody else may not.
>>
>>
>> Sure.
>>
>> And, I didn©öt mean to say this is the best way of doing things - I was
>> only explaining how many do use REGISTER for keep-olives.
>>
>> Regards,
>>
>> Christer
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>


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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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On 7/14/16 1:44 AM, Olle E. Johansson wrote:
>
>> On 14 Jul 2016, at 03:00, Drage, Keith (Nokia - GB) <keith.drage@nokia.com> wrote:
>>
>> So can you be clear on where you are trying to get to.
>>
>> 1)	deprecate SIP 1.0 --> but it never existed.
>>
>> 2)	deprecate RFC 2543 --> but this RFC is historic and therefore is of that status.
> Where can I see that status? I only see â€œstandard trackâ€ and â€œproposed standardâ€.
> https://www.rfc-editor.org/search/rfc_search_detail.php
>
> I donâ€™t agree with you here. Itâ€™s obsoleted by never declared historic as far as I
> understand. What would happen if we did?

The IESG has a statement on moving RFCs to Historic:

https://www.ietf.org/iesg/statement/designating-rfcs-as-historic

Basically - If the technology has been "retired", then its RFC can be 
given an "Historic" designation. If the tech is still in use, but has 
been updated by a replacement version, then the RFC is "Obsolete".

RFC 2543 is correctly classified as "Obsoleted by RFC 3265, RFC 3264, 
RFC 3261, RFC 3262, RFC 3263".

Jean

>>
>> 3)	deprecate mechanisms for interoperability of RFC 3261 with RFC 2543?
> Maybe that would be the effect. Can we try to figure out what would happen?
> One clear benefit highlighted seems to be to get rid of the restrictions on
> changes of the from/to headers. Now, how would that work with STIR?
>
> /O
>>
>> Regards
>>
>> Keith Drage
>>
>> -----Original Message-----
>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E. Johansson
>> Sent: 13 July 2016 21:28
>> To: Gurbani, Vijay (Nokia - US)
>> Cc: sipcore@ietf.org; Olle E Johansson
>> Subject: Re: [sipcore] Declaring SIP 1.0 historic?
>>
>>
>>> On 13 Jul 2016, at 22:12, Vijay K. Gurbani <vkg@bell-labs.com> wrote:
>>>
>>> On 07/13/2016 10:36 AM, Jonathan Lennox wrote:
>>>>> I would have no problem hearing that story - why did we keep the
>>>>> 2.0 version in 3261 - while gently SIPping a german beer in Berlin
>>>>> next week :-)
>>>>
>>>> We kept it because SIP routing rules (notably forking) mean that
>>>> there's no good way to try to send a request with a newer version,
>>>> and then if it fails, fall back to an older version.  This is why all
>>>> backward compatibility in SIP has to be single-pass.
>>>
>>> Right, that matches my recollection as well.  When we were designing
>>> what eventually became rfc3261, a prime directive was do no harm and
>>> be backwardly compatible.  Because rfc2543 was deployed and being
>>> used, a forklift upgrade by changing the protocol version was not
>>> looked at kindly.
>>>
>>> Almost a decade ago, a brief email exchange surfaced to entertain the
>>> idea of a SIP/3.0 [1].  It was ignominiously short lived :-)
>>>
>>> [1]
>>> https://www.ietf.org/mail-archive/web/sip/current/thrd19.html#18421
>>
>> Just love this quote by Dale Worley:
>>
>> "SIP 3.0 is like fully participatory democracy, the ultimate form of communism, and the earthly Kingdom of God -- a thorough reform of what presently exists that will remove all of its flaws.
>>
>> Similar to those other three projects, there is no WG timeline yet for SIP 3.0 development.â€
>>
>> Letâ€™s start SIP 4.0 :-)
>>
>> /O
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> > For instance, RFC 3261 section 12.2.1.1 indicates
> > an expectation that the "mandatory reflection of
> > the original To and From URI in mid-dialog requests
> > will be deprecated in a subsequent revision".  I'm
> > not sure why; however some vendors and operators want
> > to allow the To/From URI to change without any
> > restrictions (e.g. they don't want to be restricted
> > by RFC 4916).
>
> There are many call transfer scenarios involving b2bua=E2=80=99s
> that need to modify/update caller IDs and there are too
> many implementations pre-4916 out there.

Yes; support for RFC 3325 and RFC 5876 appears to be more common.

Updating RFC 3261 isn't necessarily going to increase support of RFC 4916 o=
r
allow To/From URI changes in a non-controlled fashion.  As far as I know,
RFC 4916 was the solution to allow To/From URI changes; however I guess tha=
t
another one can be defined.


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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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"Vijay K. Gurbani" <vkg@bell-labs.com> writes:
> Almost a decade ago, a brief email exchange surfaced to entertain the
> idea of a SIP/3.0 [1].  It was ignominiously short lived :-)
>
> [1] https://www.ietf.org/mail-archive/web/sip/current/thrd19.html#18421

What ever happened to the proposal for "peer-to-peer-to-peer-to-peer"
SIP?  My vague memory is that it was "SIP/4.0" and was an April fool's
RFC, but its not in the Wikipedia list of same.

Dale


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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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> On 14 Jul 2016, at 19:23, A. Jean Mahoney <mahoney@nostrum.com> wrote:
>=20
>=20
>=20
> On 7/14/16 1:44 AM, Olle E. Johansson wrote:
>>=20
>>> On 14 Jul 2016, at 03:00, Drage, Keith (Nokia - GB) =
<keith.drage@nokia.com> wrote:
>>>=20
>>> So can you be clear on where you are trying to get to.
>>>=20
>>> 1)	deprecate SIP 1.0 --> but it never existed.
>>>=20
>>> 2)	deprecate RFC 2543 --> but this RFC is historic and therefore is =
of that status.
>> Where can I see that status? I only see =E2=80=9Cstandard track=E2=80=9D=
 and =E2=80=9Cproposed standard=E2=80=9D.
>> https://www.rfc-editor.org/search/rfc_search_detail.php
>>=20
>> I don=E2=80=99t agree with you here. It=E2=80=99s obsoleted by never =
declared historic as far as I
>> understand. What would happen if we did?
>=20
> The IESG has a statement on moving RFCs to Historic:
>=20
> https://www.ietf.org/iesg/statement/designating-rfcs-as-historic
>=20
> Basically - If the technology has been "retired", then its RFC can be =
given an "Historic" designation. If the tech is still in use, but has =
been updated by a replacement version, then the RFC is "Obsolete".
>=20
> RFC 2543 is correctly classified as "Obsoleted by RFC 3265, RFC 3264, =
RFC 3261, RFC 3262, RFC 3263=E2=80=9D.
THank you for the explanation Jean!



/Olle

>=20
> Jean
>=20
>>>=20
>>> 3)	deprecate mechanisms for interoperability of RFC 3261 with RFC =
2543?
>> Maybe that would be the effect. Can we try to figure out what would =
happen?
>> One clear benefit highlighted seems to be to get rid of the =
restrictions on
>> changes of the from/to headers. Now, how would that work with STIR?
>>=20
>> /O
>>>=20
>>> Regards
>>>=20
>>> Keith Drage
>>>=20
>>> -----Original Message-----
>>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E. =
Johansson
>>> Sent: 13 July 2016 21:28
>>> To: Gurbani, Vijay (Nokia - US)
>>> Cc: sipcore@ietf.org; Olle E Johansson
>>> Subject: Re: [sipcore] Declaring SIP 1.0 historic?
>>>=20
>>>=20
>>>> On 13 Jul 2016, at 22:12, Vijay K. Gurbani <vkg@bell-labs.com> =
wrote:
>>>>=20
>>>> On 07/13/2016 10:36 AM, Jonathan Lennox wrote:
>>>>>> I would have no problem hearing that story - why did we keep the
>>>>>> 2.0 version in 3261 - while gently SIPping a german beer in =
Berlin
>>>>>> next week :-)
>>>>>=20
>>>>> We kept it because SIP routing rules (notably forking) mean that
>>>>> there's no good way to try to send a request with a newer version,
>>>>> and then if it fails, fall back to an older version.  This is why =
all
>>>>> backward compatibility in SIP has to be single-pass.
>>>>=20
>>>> Right, that matches my recollection as well.  When we were =
designing
>>>> what eventually became rfc3261, a prime directive was do no harm =
and
>>>> be backwardly compatible.  Because rfc2543 was deployed and being
>>>> used, a forklift upgrade by changing the protocol version was not
>>>> looked at kindly.
>>>>=20
>>>> Almost a decade ago, a brief email exchange surfaced to entertain =
the
>>>> idea of a SIP/3.0 [1].  It was ignominiously short lived :-)
>>>>=20
>>>> [1]
>>>> https://www.ietf.org/mail-archive/web/sip/current/thrd19.html#18421
>>>=20
>>> Just love this quote by Dale Worley:
>>>=20
>>> "SIP 3.0 is like fully participatory democracy, the ultimate form of =
communism, and the earthly Kingdom of God -- a thorough reform of what =
presently exists that will remove all of its flaws.
>>>=20
>>> Similar to those other three projects, there is no WG timeline yet =
for SIP 3.0 development.=E2=80=9D
>>>=20
>>> Let=E2=80=99s start SIP 4.0 :-)
>>>=20
>>> /O
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20
> _______________________________________________
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Date: Thu, 14 Jul 2016 21:35:17 +0000
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Cc: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--_000_D3A5256C1A51FFjonpetersonneustarbiz_
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I think at this point, if you want to propose your way of setting conferenc=
e policy with SIP as a problem space, feel free - I don't happen to think t=
here's much of a gap there, given the existence of means other than RFC4579=
 ad hoc INVITEs to set up conferences when you have more complicated securi=
ty needs than an ad hoc conference ordinarily would. If the group agrees th=
at there is 1) a need to convey conference policy in SIP, and 2) a need for=
 attribute-based authorization tokens issued by IdPs to support that confer=
ence policy,

Again, I am not suggesting to use SIP to set the policy (1); SIP will only =
be used to carry the authorization (2).

At its core, this is our disagreement in a nutshell. If SIP is not the prot=
ocol that is setting the conference policy, then it makes no sense to me th=
at SIP should convey the authorization for conference policy. Setting confe=
rence policy is a task that is much more aligned with the semantics and fea=
tures available in HTTP, and that's why HTTP is what the conferencing frame=
work recommends for this. Surely the protocol that needs to carry the autho=
rization for the conference policy is the protocol that carries the confere=
nce policy. Conveniently, HTTP already supports OAuth.

Jon Peterson
Neustar, Inc.


Regards,
 Rifaat



--_000_D3A5256C1A51FFjonpetersonneustarbiz_
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>
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e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
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<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
<br>
</div>
<div>I think at this point, if you want to propose your way of setting conf=
erence policy with SIP as a problem space, feel free - I don't happen to th=
ink there's much of a gap there, given the existence of means other than RF=
C4579 ad hoc INVITEs to set up conferences
 when you have more complicated security needs than an ad hoc conference or=
dinarily would. If the group agrees that there is 1) a need to convey confe=
rence policy in SIP, and 2) a need for attribute-based authorization tokens=
 issued by IdPs to support that
 conference policy,</div>
</div>
</blockquote>
<div><br>
</div>
<div>Again, I am <b>not</b> suggesting to use SIP to set the policy (1); SI=
P will only be used to carry the authorization (2).</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>At its core, this is our disagreement in a nutshell. If SIP is not the=
 protocol that is setting the conference policy, then it makes no sense to =
me that SIP should convey the authorization for conference policy. Setting =
conference policy is a task that
 is much more aligned with the semantics and features available in HTTP, an=
d that's why HTTP is what the conferencing framework recommends for this. S=
urely the protocol that needs to carry the authorization for the conference=
 policy is the protocol that carries
 the conference policy. Conveniently, HTTP already supports OAuth.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
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<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D3A5256C1A51FFjonpetersonneustarbiz_--


From nobody Thu Jul 14 17:08:49 2016
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [sipcore] Declaring SIP 1.0 historic?
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Cc: "Gurbani, Vijay \(Nokia - US\)" <vijay.gurbani@nokia-bell-labs.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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Subject: Re: [sipcore] Declaring SIP 1.0 historic?
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Although I thought it was an excellent draft, draft-kaplan-sip-four-oh-00
never became an RFC.

https://tools.ietf.org/id/draft-kaplan-sip-four-oh-00.txt


> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Dale R.
Worley
> Sent: Thursday, July 14, 2016 2:23 PM
> To: Vijay K. Gurbani
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Declaring SIP 1.0 historic?
>
> "Vijay K. Gurbani" <vkg@bell-labs.com> writes:
> > Almost a decade ago, a brief email exchange surfaced to entertain the
> > idea of a SIP/3.0 [1].  It was ignominiously short lived :-)
> >
> > [1]
> > https://www.ietf.org/mail-archive/web/sip/current/thrd19.html#18421
>
> What ever happened to the proposal for "peer-to-peer-to-peer-to-peer"
> SIP?  My vague memory is that it was "SIP/4.0" and was an April fool's
RFC,
> but its not in the Wikipedia list of same.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Brett Tate <brett@broadsoft.com> writes:
> Although I thought it was an excellent draft, draft-kaplan-sip-four-oh-00
> never became an RFC.
>
> https://tools.ietf.org/id/draft-kaplan-sip-four-oh-00.txt

Ah, yes:  "P2P2PSIP addresses the following issues in SIPv2: ... 9. Only
Alice and Bob can make phone calls".

Dale


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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Christer Holmberg <christer.holmberg@ericsson.com> writes:
>>> REGISTER requests used for keep-alive are not always authenticated, 
>>> because they are not always forwarded to the registrar to begin with.
>>> They are terminated by an edge/outbound proxy.
>>
>> I know this is done, but I have always thought it to be broken.
>>
>> The response to REGISTER is supposed to include *all* contacts that
>> have been registered, not just those that were offered in *this*
>> REGISTER. 
>
> The edge proxy can store that information from the previous REGISTER
> response it received from the registrar, and include it in the
> REGISTER responses it triggers itself. It anyway needs to keep track
> of the registration, to make sure some REGISTER requests *are*
> forwarded towards the registrar (in order for the registration to not
> expire). 

If the edge proxy has the information to provide correct responses to
the REGISTER request, then there's no problem.  Of course, that requires
that the edge proxy arranges to be updated by the registrar with
whatever changes it needs to know.

I do wonder whether it would be possible for the UA to send a REGISTER
with Max-Forwards: 0 as a keep-alive.  The edge proxy could then reply
with a 483 without needing to know any registration information, and
would be totally correct.

Dale


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde - "optimized" REGISTER
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"Olle E. Johansson" <oej@edvina.net> writes:
> Even using OPTIONs for keepalives may generate problems considering a
> SIP implementation that use OPTIONs for discovery.
>
> In the case of Asterisk OPTIONs are answered blindly. But INVITEs are
> authenticated and generated depending upon whether we have a
> configuration for that particular user.
>
> This means that OPTIONs and INVITEs have different responses. If the
> UA use the OPTIONs answer (including SDP) for discovery, the result is
> all wrong...

If OPTIONS (don't forget the S!) has Max-Forwards: 0, then an edge proxy
will give a 483 response, which is completely correct.  I think you're
talking about the case where OPTIONS reaches a central proxy, but that
proxy would reject an INVITE from the same UA.

RFC 3261 section 11.2 seems to be aware that a proxy cannot handle
OPTIONS in exactly the same way as an INVITE:

   This use of OPTIONS has limitations due to the differences in proxy
   handling of OPTIONS and INVITE requests.  While a forked INVITE can
   result in multiple 200 (OK) responses being returned, a forked
   OPTIONS will only result in a single 200 (OK) response, since it is
   treated by proxies using the non-INVITE handling.  See Section 16.7
   for the normative details.

   If the response to an OPTIONS is generated by a proxy server, the
   proxy returns a 200 (OK), listing the capabilities of the server.
   The response does not contain a message body.

   Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
   fields SHOULD be present in a 200 (OK) response to an OPTIONS
   request.  If the response is generated by a proxy, the Allow header
   field SHOULD be omitted as it is ambiguous since a proxy is method
   agnostic.  Contact header fields MAY be present in a 200 (OK)
   response and have the same semantics as in a 3xx response.  That is,
   they may list a set of alternative names and methods of reaching the
   user.  A Warning header field MAY be present.

Other than connectivity, in what way can an OPTIONS to a proxy be used
for discovery?

Christer Holmberg <christer.holmberg@ericsson.com> wrote:
> >For keepalives - I would use what's describe in section 4.4 of RFC
> >5626, regardless of if I follow outbound or not. CRLF or STUN.
> 
> The negotiation and usage of those, without outbound, is described in RFC
> 6223.
> 
> https://tools.ietf.org/rfc/rfc6223.txt

Thanks for mentioning that!  I wasn't aware of it.

Dale


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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Roman Shpount <roman@telurix.com> writes:
>> > 5. One of the big problems with encrypted SIP traffic is that it
>> > gets modified by ALG. For a lot of hosted PBX providers avoiding
>> > the need to troubleshoot customer routers is a bigger incentive to
>> > deploy TLS [than] user privacy.
>>
>> If I understand you correctly, hosted PBX providers favor TLS to stop
>> routers from modifying SIP messages.  But ALGs terminate TLS ... and
>> then modify the SIP messages.
>>
> One of the big problems with deploying SIP on customer premises are SIP ALG
> implemented by the consumer routers. Such ALG are typically broken and
> cause SIP call setup failures or call drops. If client equipment is using
> SIP over TLS, SIP ALG in routers cannot affect the signaling and SIP
> message pass the router unmolested. Because of this, SIP calls over TLS are
> much more stable which results in big reduction of customer problems and
> associated support calls. This reduction in support burden is a much bigger
> incentive for service providers to deploy encryption for SIP then customer
> privacy (which, in practice, is almost never their concern).

We'll have to remember this as a design point.

However this does mean that the UAs need to know the edge proxy's
certificate, so the ALG can't terminate TLS.

Dale


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From: Roman Shpount <roman@telurix.com>
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--001a113ec7ec8ec0b60537dfa125
Content-Type: text/plain; charset=UTF-8

On Sun, Jul 17, 2016 at 9:37 PM, Dale R. Worley <worley@ariadne.com> wrote:

> Roman Shpount <roman@telurix.com> writes:
> >> > 5. One of the big problems with encrypted SIP traffic is that it
> >> > gets modified by ALG. For a lot of hosted PBX providers avoiding
> >> > the need to troubleshoot customer routers is a bigger incentive to
> >> > deploy TLS [than] user privacy.
> >>
> >> If I understand you correctly, hosted PBX providers favor TLS to stop
> >> routers from modifying SIP messages.  But ALGs terminate TLS ... and
> >> then modify the SIP messages.
> >>
> > One of the big problems with deploying SIP on customer premises are SIP
> ALG
> > implemented by the consumer routers. Such ALG are typically broken and
> > cause SIP call setup failures or call drops. If client equipment is using
> > SIP over TLS, SIP ALG in routers cannot affect the signaling and SIP
> > message pass the router unmolested. Because of this, SIP calls over TLS
> are
> > much more stable which results in big reduction of customer problems and
> > associated support calls. This reduction in support burden is a much
> bigger
> > incentive for service providers to deploy encryption for SIP then
> customer
> > privacy (which, in practice, is almost never their concern).
>
> We'll have to remember this as a design point.
>
> However this does mean that the UAs need to know the edge proxy's
> certificate, so the ALG can't terminate TLS.
>

UA do not need to know the edge proxy's certificate. This certificate can
be verified against the well known certificate authorities, similar to how
HTTPS web site certificates are verified. This works unless end user
specifically added ALG's certificate to the list of CAs on the UA, which is
typically not done.
_____________
Roman Shpount

--001a113ec7ec8ec0b60537dfa125
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" data-smartmail=3D"gmail_signature">On Sun, Jul 17, 2016 at 9:37 PM, Da=
le R. Worley <span dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com" ta=
rget=3D"_blank">worley@ariadne.com</a>&gt;</span> wrote:<br></div></div><di=
v class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><=
span class=3D"">Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roma=
n@telurix.com</a>&gt; writes:<br>
&gt;&gt; &gt; 5. One of the big problems with encrypted SIP traffic is that=
 it<br>
&gt;&gt; &gt; gets modified by ALG. For a lot of hosted PBX providers avoid=
ing<br>
&gt;&gt; &gt; the need to troubleshoot customer routers is a bigger incenti=
ve to<br>
</span>&gt;&gt; &gt; deploy TLS [than] user privacy.<br>
<span class=3D"">&gt;&gt;<br>
&gt;&gt; If I understand you correctly, hosted PBX providers favor TLS to s=
top<br>
&gt;&gt; routers from modifying SIP messages.=C2=A0 But ALGs terminate TLS =
... and<br>
&gt;&gt; then modify the SIP messages.<br>
&gt;&gt;<br>
&gt; One of the big problems with deploying SIP on customer premises are SI=
P ALG<br>
&gt; implemented by the consumer routers. Such ALG are typically broken and=
<br>
&gt; cause SIP call setup failures or call drops. If client equipment is us=
ing<br>
&gt; SIP over TLS, SIP ALG in routers cannot affect the signaling and SIP<b=
r>
&gt; message pass the router unmolested. Because of this, SIP calls over TL=
S are<br>
&gt; much more stable which results in big reduction of customer problems a=
nd<br>
&gt; associated support calls. This reduction in support burden is a much b=
igger<br>
&gt; incentive for service providers to deploy encryption for SIP then cust=
omer<br>
&gt; privacy (which, in practice, is almost never their concern).<br>
<br>
</span>We&#39;ll have to remember this as a design point.<br>
<br>
However this does mean that the UAs need to know the edge proxy&#39;s<br>
certificate, so the ALG can&#39;t terminate TLS.<br></blockquote><div><br><=
/div><div>UA do not need to know the edge proxy&#39;s certificate. This cer=
tificate can be verified against the well known certificate authorities, si=
milar to how HTTPS web site certificates are verified. This works unless en=
d user specifically added ALG&#39;s certificate to the list of CAs on the U=
A, which is typically not done.=C2=A0</div><div><div class=3D"gmail_signatu=
re" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</div>=
</div><div>=C2=A0</div></div></div></div>

--001a113ec7ec8ec0b60537dfa125--


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>>>>> "RS" == Roman Shpount <roman@telurix.com> writes:

RS> UA do not need to know the edge proxy's certificate. This certificate can
RS> be verified against the well known certificate authorities, similar to how
RS> HTTPS web site certificates are verified.

Most sip software I've looked at also supports not verifying the certs
at all.  If the only reason for using tls is to avoid mangling by the
edge routers, then at least in the short term (next few years, before
the edge router manufactures catch on) tls w/o auth would get the job
done.

Of course w/ auth is nicer, and letsencrypt makes that essentially free
(just the labor cost of setting it up).

-JimC
-- 
James Cloos <cloos@jhcloos.com>         OpenPGP: 0x997A9F17ED7DAEA6


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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
Date: Tue, 19 Jul 2016 06:07:15 +0200
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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Consumer routers deploy SIP ALG to mainly fix the address/port in SDP to
enable media to traverse consumer the NAT, right? Are we saying consumer
endpoints are already doing ICE, so making those ALGs non-functional as a
result of doing SIP over TLS wouldn't be an issue?

thanks,
Muthu

On Wed, Jul 13, 2016 at 10:44 PM, Roman Shpount <roman@telurix.com> wrote:

>
> On Wed, Jul 6, 2016 at 5:54 PM, Dale R. Worley <worley@ariadne.com> wrote:
>
>> Thanks for the information!  I've added your discussion to the rough
>> draft.
>>
>> Roman Shpount <roman@telurix.com> writes:
>> > Few observations:
>> > [...]
>> > 3. For server, the most efficient way to keep the NAT hole open is to
>> send
>> > OPTIONS or NOTIFY requests to the client. This is much more efficient
>> then
>> > registrations with small timeouts.
>> >
>> > 4. Registration server which has an in-memory database for current
>> > registrations can be fairly efficient in reducing number of back-end DB
>> > updates due to registrations and subscriptions with small timeouts. It
>> is
>> > not going to be stateless, but its state does not need to be replicated
>> and
>> > would automatically recover on the stand-by server on the next
>> registration
>> > or subscription request from the client.
>>
>> I think what you are saying here is that if the server sends
>> OPTIONS/NOTIFY for keepalive, rather than making the client send
>> frequent re-REGISTERS, then the server doesn't need to record the timers
>> for the keepalives in persistent storage -- if the stand-by server is
>> activated, it simply sends keepalives to all registrations that are
>> recorded as having keepalive service.  In the long run, this requires no
>> more processing by the registration server and far less demand on the
>> persistent storage.
>>
>> Is that correct?
>>
>
> This is correct. Since OPTIONS/NOTIFY do not update the registration state
> they typically require less resources then forcing the client to
> re-register at the small interval. Registrations typically require
> authentication and cause the shared storage update for the expiration time.
> OPTIONS/NOTIFY sent from server require not of this.
>
>>
>> > 5. One of the big problems with encrypted SIP traffic is that it gets
>> > modified by ALG. For a lot of hosted PBX providers avoiding the need to
>> > troubleshoot customer routers is a bigger incentive to deploy TLS then
>> user
>> > privacy.
>>
>> If I understand you correctly, hosted PBX providers favor TLS to stop
>> routers from modifying SIP messages.  But ALGs terminate TLS ... and
>> then modify the SIP messages.
>>
>>
> One of the big problems with deploying SIP on customer premises are SIP
> ALG implemented by the consumer routers. Such ALG are typically broken and
> cause SIP call setup failures or call drops. If client equipment is using
> SIP over TLS, SIP ALG in routers cannot affect the signaling and SIP
> message pass the router unmolested. Because of this, SIP calls over TLS are
> much more stable which results in big reduction of customer problems and
> associated support calls. This reduction in support burden is a much bigger
> incentive for service providers to deploy encryption for SIP then customer
> privacy (which, in practice, is almost never their concern).
> _____________
> Roman Shpount
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>

--94eb2c03ba046b93090537f538d3
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">Consumer routers deploy SIP ALG to main=
ly fix the address/port in SDP to enable media to traverse consumer the NAT=
, right? Are we saying consumer endpoints are already doing ICE, so making =
those ALGs non-functional as a result of doing SIP over TLS wouldn&#39;t be=
 an issue?</div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif;font-size:small"><br></div><div class=3D"gmail_default" s=
tyle=3D"font-family:arial,helvetica,sans-serif;font-size:small">thanks,</di=
v><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-se=
rif;font-size:small">Muthu</div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Wed, Jul 13, 2016 at 10:44 PM, Roman Shpount <span dir=
=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@t=
elurix.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"><span class=3D""><div><div data-sma=
rtmail=3D"gmail_signature">On Wed, Jul 6, 2016 at 5:54 PM, Dale R. Worley <=
span dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com" target=3D"_blank=
">worley@ariadne.com</a>&gt;</span> wrote:<br></div></div></span><div class=
=3D"gmail_quote"><span class=3D""><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex">Thanks for the information!=C2=A0 I&#39;ve added your discussion=
 to the rough<br>
draft.<br>
<br>
Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">ro=
man@telurix.com</a>&gt; writes:<br>
&gt; Few observations:<br>
&gt; [...]<br>
<span>&gt; 3. For server, the most efficient way to keep the NAT hole open =
is to send<br>
&gt; OPTIONS or NOTIFY requests to the client. This is much more efficient =
then<br>
&gt; registrations with small timeouts.<br>
&gt;<br>
&gt; 4. Registration server which has an in-memory database for current<br>
&gt; registrations can be fairly efficient in reducing number of back-end D=
B<br>
&gt; updates due to registrations and subscriptions with small timeouts. It=
 is<br>
&gt; not going to be stateless, but its state does not need to be replicate=
d and<br>
&gt; would automatically recover on the stand-by server on the next registr=
ation<br>
&gt; or subscription request from the client.<br>
<br>
</span>I think what you are saying here is that if the server sends<br>
OPTIONS/NOTIFY for keepalive, rather than making the client send<br>
frequent re-REGISTERS, then the server doesn&#39;t need to record the timer=
s<br>
for the keepalives in persistent storage -- if the stand-by server is<br>
activated, it simply sends keepalives to all registrations that are<br>
recorded as having keepalive service.=C2=A0 In the long run, this requires =
no<br>
more processing by the registration server and far less demand on the<br>
persistent storage.<br>
<br>
Is that correct?<br></blockquote><div><br></div></span><div>This is correct=
. Since OPTIONS/NOTIFY do not update the registration state they typically =
require less resources then forcing the client to re-register at the small =
interval. Registrations typically require authentication and cause the shar=
ed storage update for the expiration time. OPTIONS/NOTIFY sent from server =
require not of this.</div><span class=3D""><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);=
padding-left:1ex">
<span><br>
&gt; 5. One of the big problems with encrypted SIP traffic is that it gets<=
br>
&gt; modified by ALG. For a lot of hosted PBX providers avoiding the need t=
o<br>
&gt; troubleshoot customer routers is a bigger incentive to deploy TLS then=
 user<br>
&gt; privacy.<br>
<br>
</span>If I understand you correctly, hosted PBX providers favor TLS to sto=
p<br>
routers from modifying SIP messages.=C2=A0 But ALGs terminate TLS ... and<b=
r>
then modify the SIP messages.<br>
<span><font color=3D"#888888"><br></font></span></blockquote><div><br></div=
></span><div>One of the big problems with deploying SIP on customer premise=
s are SIP ALG implemented by the consumer routers. Such ALG are typically b=
roken and cause SIP call setup failures or call drops. If client equipment =
is using SIP over TLS, SIP ALG in routers cannot affect the signaling and S=
IP message pass the router unmolested. Because of this, SIP calls over TLS =
are much more stable which results in big reduction of customer problems an=
d associated support calls. This reduction in support burden is a much bigg=
er incentive for service providers to deploy encryption for SIP then custom=
er privacy (which, in practice, is almost never their concern).=C2=A0</div>=
<div><div data-smartmail=3D"gmail_signature">_____________<span class=3D"HO=
EnZb"><font color=3D"#888888"><br>Roman Shpount</font></span></div></div><d=
iv>=C2=A0</div></div></div></div>
<br>_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
<br></blockquote></div><br></div></div>

--94eb2c03ba046b93090537f538d3--


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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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Muthu Arul Mozhi Perumal <muthu.arul@gmail.com> writes:
> Consumer routers deploy SIP ALG to mainly fix the address/port in SDP to
> enable media to traverse consumer the NAT, right? Are we saying consumer
> endpoints are already doing ICE, so making those ALGs non-functional as a
> result of doing SIP over TLS wouldn't be an issue?

That's the idea:  The UA should use Outbound to keep a NAT binding, and
ICE to negotiate media ports.

Looking at the SIPit 31 summary (https://www.sipit.net/SIPit31_summary),
44% of endpoints support ICE, whereas only 28% support outbound.

Dale


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
Thread-Topic: [sipcore] Happy Eyeballs: Outbounde
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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Hi,

Outbound is more than a mechanism for keeping NAT bindings open.

So, before saying "let's use outbound", what exactly are the properties/fea=
tures that you need?

Regards,

Christer

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Dale R. Worley
Sent: 19 July 2016 18:47
To: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
Cc: sipcore@ietf.org; roman@telurix.com
Subject: Re: [sipcore] Happy Eyeballs: Outbounde

Muthu Arul Mozhi Perumal <muthu.arul@gmail.com> writes:
> Consumer routers deploy SIP ALG to mainly fix the address/port in SDP=20
> to enable media to traverse consumer the NAT, right? Are we saying=20
> consumer endpoints are already doing ICE, so making those ALGs=20
> non-functional as a result of doing SIP over TLS wouldn't be an issue?

That's the idea:  The UA should use Outbound to keep a NAT binding, and ICE=
 to negotiate media ports.

Looking at the SIPit 31 summary (https://www.sipit.net/SIPit31_summary),
44% of endpoints support ICE, whereas only 28% support outbound.

Dale

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


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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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--001a114370bef979ee0538015a13
Content-Type: text/plain; charset=UTF-8

On Tue, Jul 19, 2016 at 12:07 AM, Muthu Arul Mozhi Perumal <
muthu.arul@gmail.com> wrote:

> Consumer routers deploy SIP ALG to mainly fix the address/port in SDP to
> enable media to traverse consumer the NAT, right? Are we saying consumer
> endpoints are already doing ICE, so making those ALGs non-functional as a
> result of doing SIP over TLS wouldn't be an issue?
>
>
What I am saying is that most consumer routers include ALG code which is
broken. This code is supposed to update IP addresses in SIP Via and Contact
as well as in SDP. It is also supposed to setup packet forwarding between
public IP/port to the NATed IP/port. In reality, code in most of the
routers is so bad that the person who wrote it most likely needs to change
profession to avoid spreading future misery. Not doing anything to SIP on
the router in combination with ICE or server side NAT traversal works much
better then SIP ALG. There is almost never a reason to have SIP ALG enabled
unless you are using a very restrictive firewall.

For more information you can google "SIP ALG are evil" or read any of the
following:

https://support.onsip.com/hc/en-us/articles/204514014-NAT-and-Firewall-Traversal-Recommendation

http://support.phone.com/hc/en-us/articles/204162770-Do-I-Have-To-Change-My-Network-Settings-or-Hardware-

http://www.whichvoip.com/disable-sip-alg.htm

Look at the number of routers where SIP ALG needs to be disabled:
http://businesssupport.vonage.com/app/answers/detail/a_id/21546/~/network-equipment-compatibility

You can probably find a similar article for every hosted VoIP provider.

Regards,
_____________
Roman Shpount

--001a114370bef979ee0538015a13
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" data-smartmail=3D"gmail_signature">On Tue, Jul 19, 2016 at 12:07 AM, M=
uthu Arul Mozhi Perumal <span dir=3D"ltr">&lt;<a href=3D"mailto:muthu.arul@=
gmail.com" target=3D"_blank">muthu.arul@gmail.com</a>&gt;</span> wrote:<br>=
</div></div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padd=
ing-left:1ex"><div dir=3D"ltr"><div style=3D"font-family:arial,helvetica,sa=
ns-serif;font-size:small">Consumer routers deploy SIP ALG to mainly fix the=
 address/port in SDP to enable media to traverse consumer the NAT, right? A=
re we saying consumer endpoints are already doing ICE, so making those ALGs=
 non-functional as a result of doing SIP over TLS wouldn&#39;t be an issue?=
</div><div style=3D"font-family:arial,helvetica,sans-serif;font-size:small"=
><br></div></div></blockquote><div><br></div><div>What I am saying is that =
most consumer routers include ALG code which is broken. This code is suppos=
ed to update IP addresses in SIP Via and Contact as well as in SDP. It is a=
lso supposed to setup packet forwarding between public IP/port to the NATed=
 IP/port. In reality, code in most of the routers is so bad that the person=
 who wrote it most likely needs to change profession to avoid spreading fut=
ure misery. Not doing anything to SIP on the router in combination with ICE=
 or server side NAT traversal works much better then SIP ALG. There is almo=
st never a reason to have SIP ALG enabled unless you are using a very restr=
ictive firewall.=C2=A0</div><div><br></div><div>For more information you ca=
n google &quot;SIP ALG are evil&quot; or read any of the following:</div><d=
iv><br></div><div><a href=3D"https://support.onsip.com/hc/en-us/articles/20=
4514014-NAT-and-Firewall-Traversal-Recommendation">https://support.onsip.co=
m/hc/en-us/articles/204514014-NAT-and-Firewall-Traversal-Recommendation</a>=
<br></div><div><br></div><div><a href=3D"http://support.phone.com/hc/en-us/=
articles/204162770-Do-I-Have-To-Change-My-Network-Settings-or-Hardware-">ht=
tp://support.phone.com/hc/en-us/articles/204162770-Do-I-Have-To-Change-My-N=
etwork-Settings-or-Hardware-</a><br></div><div><br></div><div><a href=3D"ht=
tp://www.whichvoip.com/disable-sip-alg.htm">http://www.whichvoip.com/disabl=
e-sip-alg.htm</a><br></div><div><br></div><div>Look at the number of router=
s where SIP ALG needs to be disabled:</div><div><a href=3D"http://businesss=
upport.vonage.com/app/answers/detail/a_id/21546/~/network-equipment-compati=
bility">http://businesssupport.vonage.com/app/answers/detail/a_id/21546/~/n=
etwork-equipment-compatibility</a><br></div><div><br></div><div>You can pro=
bably find a similar article for every hosted VoIP provider.</div><div><br>=
</div><div>Regards,</div><div><div class=3D"gmail_signature" data-smartmail=
=3D"gmail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0<=
/div></div></div></div>

--001a114370bef979ee0538015a13--


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From: Roman Shpount <roman@telurix.com>
Date: Tue, 19 Jul 2016 14:45:12 -0400
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>, "Dale R. Worley" <worley@ariadne.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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--001a113f8aae3056400538017c4f
Content-Type: text/plain; charset=UTF-8

On Tue, Jul 19, 2016 at 12:02 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Outbound is more than a mechanism for keeping NAT bindings open.
>
> So, before saying "let's use outbound", what exactly are the
> properties/features that you need?
>

I have mentioned this in another thread but Happy Eyeballs do not need to
have anything to do with SIP Outbound. Main connection that I do see is
using SIP Outbound Keep-Alive messages to detect the working flow for SIP
messages when multiple local networks or multiple SIP Edge proxy addresses
are available.

I have also mentioned that this is not the only solution for Happy
Eyeballs, since figuring out rules for sending messages in the same SIP
transaction across multiple flows can serve as an alternative, and possibly
better, solution.

Obviously, if we restrict SIP to TLS only as the only acceptable protocol
for Happy Eyeballs, then SIP Outbound is the only standard solution which
attempts to describe how to make SIP work with TLS from end points behind
NAT. This, as I have mentioned before, have serious scalability and
fail-over issues for current deployments.

Regards,
_____________
Roman Shpount

--001a113f8aae3056400538017c4f
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, Jul 19, 2016 at 12:02 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:1px solid rgb(204,204,204)=
;padding-left:1ex">Outbound is more than a mechanism for keeping NAT bindin=
gs open.<br>
<br>
So, before saying &quot;let&#39;s use outbound&quot;, what exactly are the =
properties/features that you need?<br></blockquote><div><br></div><div>I ha=
ve mentioned this in another thread but Happy Eyeballs do not need to have =
anything to do with SIP Outbound. Main connection that I do see is using SI=
P Outbound Keep-Alive messages to detect the working flow for SIP messages =
when multiple local networks or multiple SIP Edge proxy addresses are avail=
able.</div><div><br></div><div>I have also mentioned that this is not the o=
nly solution for Happy Eyeballs, since figuring out rules for sending messa=
ges in the same SIP transaction across multiple flows can serve as an alter=
native, and possibly better, solution.</div><div><br></div><div>Obviously, =
if we restrict SIP to TLS only as the only acceptable protocol for Happy Ey=
eballs, then SIP Outbound is the only standard solution which attempts to d=
escribe how to make SIP work with TLS from end points behind NAT. This, as =
I have mentioned before, have serious scalability and fail-over issues for =
current deployments.</div><div><br></div><div>Regards,</div><div><div class=
=3D"gmail_signature" data-smartmail=3D"gmail_signature">_____________<br>Ro=
man Shpount</div></div><div>=C2=A0</div></div></div></div>

--001a113f8aae3056400538017c4f--


From nobody Wed Jul 20 11:40:33 2016
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From: "A. Jean Mahoney" <mahoney@nostrum.com>
To: SIPCORE <sipcore@ietf.org>, "Dale R. Worley" <worley@ariadne.com>
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Subject: [sipcore] Quick notes from the SIPCORE WG session
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Thanks to Dale for presenting at the meeting today! And although we had 
some technical glitches, I thought the discussion went pretty well.

This is what I took away from the meeting (these aren't the minutes, 
just my notes):


Action: Focus on flow setup with separate drafts for each transport.

Action: Roman Shpount to send to the list an analysis of DTLS.

Action: Chairs to discuss updating milestones with AD.

Reducing transaction timeouts is out of scope (Robert Sparks briefly had 
an action item to send to the list an analysis of reducing timers F and 
B, but this does not appear to be needed at the moment).

The following topics are interesting, and need discussion, but are 
broader than happy eyeballs:
- call/connection recovery
- signaling encryption/ALG traversal
- DTLS lite
- Outbound lite


Let me know if I didn't capture it accurately.

Thanks!

Jean


From nobody Wed Jul 20 12:18:31 2016
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Why forking calls does not solve transport problems
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The question was raised during the Sipcore session of why forking does
not solve dual-stack problems.  I attempted to outline how problems
arise with forking, but I made some mistakes in the details.  This is a
cleaned-up version.

Naively, one thinks that sending a duplicate of a request to an
alternative target should not cause a problem, because the ultimate
destination UA can arbitrate between multiple requests, rejecting all
but the first with 482.

I used to think that myself.

But I ran into a customer whose SIP system failed in ways that point out
problems.  They had two offices.  Each was served by a proxy/registrar
that handled the phones in that office.  The two proxies routed calls to
phones in the other office via the other office's proxy.  Each proxy was
paired with a voicemail server that took voicemail for its office's
phones.

Voicemail was handed in the usual way:  A call was serially forked first
to the registered contacts for the AOR, and if those timed out, the call
was forked to the voicemail server.

The ultimate cause of the problem was that each proxy forwarded to the
other using the proxy's DNS name, but without a transport specification.
In effect, forwarded calls could be serially forked, first by UDP
transmission, then by TCP transmission.  (Or perhaps it was the other
order.)  This isn't *real* forking, since the sending proxy doesn't
record alternative targets (network addresses) for one destination URI
as forks, but the destination can receive two copies of one INVITE, with
different branch values (see RFC 3263 section 4.3), so they look like
different forks to the recipient.

Occasionally, the network between the two proxies was slow, and when a
proxy sent an INVITE to the other, the other proxy didn't respond
quickly enough, and so the proxy sent another "fork" of the INVITE to
the other proxy using the other transport.

At this point, one "fork" runs from proxy 1 to proxy 2 and then to the
phone, where it rings.  The second "fork" runs from proxy 1 to proxy 2
-- which handles it as specified -- sending the second INVITE to the
phone.

The phone rejects the second INVITE with 482.  This causes the second
proxy to step the second "fork" to the second destination, viz., the
voicemail server.

The voicemail server answers immediately, sending a 200 response.  That
propagates back to proxy 1, which considers the call established.  But
since proxy 1 does not see forking, it does not cancel the first INVITE
it sent.  So the phone continues to ring until the user picks up the
call.  But the caller is talking to the voicemail system.

In the case of our customer, the solution was to change the URIs that
each used for the other proxy to specify a transport parameter and
eliminate the implicit forking.

Dale


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From: Roman Shpount <roman@telurix.com>
Date: Thu, 21 Jul 2016 15:53:44 -0400
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Subject: Re: [sipcore] Why forking calls does not solve transport problems
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This sounds correct (and underlines the existing problem with forking).

One of the solutions that we deployed for dual network scenario was to send
multiple INVITE or REGISTRATION requests over all available networks.
Identical VIA branch value was used for all of those messages to indicate
they are part of the same transaction. The way the system was designed all
of these messages would be terminated by the same B2BUA/Registrations agent
which was selected based on the VIA branch hash value. This
B2BUA/Registration agent was specifically designed to aggregate messages
received over multiple transports into a single transaction which prevented
problems like the one described above. I know this is not a generic
solution which can be applied everywhere but it worked quite well with
variety of fixed and mobile networks across multiple countries for a fairly
large user base.

Regards,
_____________
Roman Shpount

On Wed, Jul 20, 2016 at 3:18 PM, Dale R. Worley <worley@ariadne.com> wrote:

> The question was raised during the Sipcore session of why forking does
> not solve dual-stack problems.  I attempted to outline how problems
> arise with forking, but I made some mistakes in the details.  This is a
> cleaned-up version.
>
> Naively, one thinks that sending a duplicate of a request to an
> alternative target should not cause a problem, because the ultimate
> destination UA can arbitrate between multiple requests, rejecting all
> but the first with 482.
>
> I used to think that myself.
>
> But I ran into a customer whose SIP system failed in ways that point out
> problems.  They had two offices.  Each was served by a proxy/registrar
> that handled the phones in that office.  The two proxies routed calls to
> phones in the other office via the other office's proxy.  Each proxy was
> paired with a voicemail server that took voicemail for its office's
> phones.
>
> Voicemail was handed in the usual way:  A call was serially forked first
> to the registered contacts for the AOR, and if those timed out, the call
> was forked to the voicemail server.
>
> The ultimate cause of the problem was that each proxy forwarded to the
> other using the proxy's DNS name, but without a transport specification.
> In effect, forwarded calls could be serially forked, first by UDP
> transmission, then by TCP transmission.  (Or perhaps it was the other
> order.)  This isn't *real* forking, since the sending proxy doesn't
> record alternative targets (network addresses) for one destination URI
> as forks, but the destination can receive two copies of one INVITE, with
> different branch values (see RFC 3263 section 4.3), so they look like
> different forks to the recipient.
>
> Occasionally, the network between the two proxies was slow, and when a
> proxy sent an INVITE to the other, the other proxy didn't respond
> quickly enough, and so the proxy sent another "fork" of the INVITE to
> the other proxy using the other transport.
>
> At this point, one "fork" runs from proxy 1 to proxy 2 and then to the
> phone, where it rings.  The second "fork" runs from proxy 1 to proxy 2
> -- which handles it as specified -- sending the second INVITE to the
> phone.
>
> The phone rejects the second INVITE with 482.  This causes the second
> proxy to step the second "fork" to the second destination, viz., the
> voicemail server.
>
> The voicemail server answers immediately, sending a 200 response.  That
> propagates back to proxy 1, which considers the call established.  But
> since proxy 1 does not see forking, it does not cancel the first INVITE
> it sent.  So the phone continues to ring until the user picks up the
> call.  But the caller is talking to the voicemail system.
>
> In the case of our customer, the solution was to change the URIs that
> each used for the other proxy to specify a transport parameter and
> eliminate the implicit forking.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

--001a113f7e1cf57d8305382aac30
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra">This sounds correct (and un=
derlines the existing problem with forking).</div><div class=3D"gmail_extra=
"><br></div><div class=3D"gmail_extra">One of the solutions that we deploye=
d for dual network scenario was to send multiple INVITE or REGISTRATION req=
uests over all available networks. Identical VIA branch value was used for =
all of those messages to indicate they are part of the same transaction. Th=
e way the system was designed all of these messages would be terminated by =
the same B2BUA/Registrations agent which was selected based on the VIA bran=
ch hash value. This B2BUA/Registration agent was specifically designed to a=
ggregate messages received over multiple transports into a single transacti=
on which prevented problems like the one described above. I know this is no=
t a generic solution which can be applied everywhere but it worked quite we=
ll with variety of fixed and mobile networks across multiple countries for =
a fairly large user base.</div><div class=3D"gmail_extra"><br></div><div cl=
ass=3D"gmail_extra">Regards,<br clear=3D"all"><div><div class=3D"gmail_sign=
ature" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</d=
iv></div>
<br><div class=3D"gmail_quote">On Wed, Jul 20, 2016 at 3:18 PM, Dale R. Wor=
ley <span dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com" target=3D"_=
blank">worley@ariadne.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">The question was raised during the Sipcore session of why forking do=
es<br>
not solve dual-stack problems.=C2=A0 I attempted to outline how problems<br=
>
arise with forking, but I made some mistakes in the details.=C2=A0 This is =
a<br>
cleaned-up version.<br>
<br>
Naively, one thinks that sending a duplicate of a request to an<br>
alternative target should not cause a problem, because the ultimate<br>
destination UA can arbitrate between multiple requests, rejecting all<br>
but the first with 482.<br>
<br>
I used to think that myself.<br>
<br>
But I ran into a customer whose SIP system failed in ways that point out<br=
>
problems.=C2=A0 They had two offices.=C2=A0 Each was served by a proxy/regi=
strar<br>
that handled the phones in that office.=C2=A0 The two proxies routed calls =
to<br>
phones in the other office via the other office&#39;s proxy.=C2=A0 Each pro=
xy was<br>
paired with a voicemail server that took voicemail for its office&#39;s<br>
phones.<br>
<br>
Voicemail was handed in the usual way:=C2=A0 A call was serially forked fir=
st<br>
to the registered contacts for the AOR, and if those timed out, the call<br=
>
was forked to the voicemail server.<br>
<br>
The ultimate cause of the problem was that each proxy forwarded to the<br>
other using the proxy&#39;s DNS name, but without a transport specification=
.<br>
In effect, forwarded calls could be serially forked, first by UDP<br>
transmission, then by TCP transmission.=C2=A0 (Or perhaps it was the other<=
br>
order.)=C2=A0 This isn&#39;t *real* forking, since the sending proxy doesn&=
#39;t<br>
record alternative targets (network addresses) for one destination URI<br>
as forks, but the destination can receive two copies of one INVITE, with<br=
>
different branch values (see RFC 3263 section 4.3), so they look like<br>
different forks to the recipient.<br>
<br>
Occasionally, the network between the two proxies was slow, and when a<br>
proxy sent an INVITE to the other, the other proxy didn&#39;t respond<br>
quickly enough, and so the proxy sent another &quot;fork&quot; of the INVIT=
E to<br>
the other proxy using the other transport.<br>
<br>
At this point, one &quot;fork&quot; runs from proxy 1 to proxy 2 and then t=
o the<br>
phone, where it rings.=C2=A0 The second &quot;fork&quot; runs from proxy 1 =
to proxy 2<br>
-- which handles it as specified -- sending the second INVITE to the<br>
phone.<br>
<br>
The phone rejects the second INVITE with 482.=C2=A0 This causes the second<=
br>
proxy to step the second &quot;fork&quot; to the second destination, viz., =
the<br>
voicemail server.<br>
<br>
The voicemail server answers immediately, sending a 200 response.=C2=A0 Tha=
t<br>
propagates back to proxy 1, which considers the call established.=C2=A0 But=
<br>
since proxy 1 does not see forking, it does not cancel the first INVITE<br>
it sent.=C2=A0 So the phone continues to ring until the user picks up the<b=
r>
call.=C2=A0 But the caller is talking to the voicemail system.<br>
<br>
In the case of our customer, the solution was to change the URIs that<br>
each used for the other proxy to specify a transport parameter and<br>
eliminate the implicit forking.<br>
<br>
Dale<br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</blockquote></div><br></div></div>

--001a113f7e1cf57d8305382aac30--


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From: "Ben Campbell" <ben@nostrum.com>
To: "Dale R. Worley" <worley@ariadne.com>
Date: Wed, 27 Jul 2016 17:37:32 -0500
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I am not aware of any rule against a PS updating an informational RFC. A 
PS is assumed to have greater "maturity" than an informational. The 
opposite would be more of a problem.

If such text is added, it's probably worth mentioning the status, and 
the fact that the update is in a PS is not intended to change that. 
(That is, any text added by the update would still be informational.)

Thanks!

Ben.

On 16 Jun 2016, at 11:25, Dale R. Worley wrote:

> "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com> writes:
>> A reminder that RFC 3325 is an informational document, so I am not
>> sure whether a standards track document can update an informational
>> document. Anyone able to clarify?
>
> I searched, but I couldn't find much guidance on what update documents
> are or are not allowed to do in the BCP directory.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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"Ben Campbell" <ben@nostrum.com> writes:
> I am not aware of any rule against a PS updating an informational RFC. A 
> PS is assumed to have greater "maturity" than an informational. The 
> opposite would be more of a problem.
>
> If such text is added, it's probably worth mentioning the status, and 
> the fact that the update is in a PS is not intended to change that. 
> (That is, any text added by the update would still be informational.)

I suppose the thing to do is mark that part of the PS as
"non-normative".  Maybe "This section is a non-normative update to the
informational RFC nnnn."?

Dale


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On 28 Jul 2016, at 14:08, Dale R. Worley wrote:

> "Ben Campbell" <ben@nostrum.com> writes:
>> I am not aware of any rule against a PS updating an informational 
>> RFC. A
>> PS is assumed to have greater "maturity" than an informational. The
>> opposite would be more of a problem.
>>
>> If such text is added, it's probably worth mentioning the status, and
>> the fact that the update is in a PS is not intended to change that.
>> (That is, any text added by the update would still be informational.)
>
> I suppose the thing to do is mark that part of the PS as
> "non-normative".  Maybe "This section is a non-normative update to the
> informational RFC nnnn."?

I fear that a discussion about what it means to normatively or 
non-normatively update something would be harder to resolve than we 
might like. Simply noting that the RFC being updated is informational, 
and remains so as updated seems sufficient.

OTOH, it would also be perfectly reasonable to split any such update 
into a separate informational draft.

Ben.



>
> Dale
>
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