
From nobody Tue Mar  1 08:23:34 2016
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Progressing draft-ietf-sipcore-dns-dual-stack
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Since there seems to be no controversy, I'm planning on asking for a
WGLC on 8 Mar 2016, which is two weeks after
draft-ietf-sipcore-dns-dual-stack-04 was submitted.

Alternatively, we could ask for time in Buenos Aires to discuss it.

Opinions?

Dale


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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
> Why do we need to have a solution for UDP?

My experience is that UDP is universal, so if there's a straightforward
way to do it in UDP, it will make implementation much quicker.

In the SIPit 31 report, I see:

    Implementations using each transport for SIP messages:
       UDP   100% 
       TCP    86%
       TLS    81% (20% server-auth-only)
       SCTP   10%
       DTLS    5%

Dale


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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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"Olle E. Johansson" <oej@edvina.net> writes:
>>    C. Two or more addresses may be contacted simultaneously rather than
>>    sequentially if provisions are made to avoid merged requests:
>
> Again, this only applies to UDP.

I don't see this as limited to UDP.  If Happy Eyeballs processing
decides to race two TCP connections to their targets (typically one IPv4
and one IPv6), ideally we don't want to have to wait a complete RTT
before a the request can be sent.  Better to send the request along with
the SYN packet on one connection, so if we don't have to fail over to
the other connection, we don't lose an RTT.

> The migration to IPv6 needs to be easy, and asking for dual simultaneous
> connections over TCP/TLS is big enough for many implementations. Implementing
> a whole new proxy merge scheme in the same document will in my view make it as
> ignored as SIP outbound currently is

I see the merge scheme as being *much easier* to implement than
Outbound, and probably simpler than dual simultaneous connections in
TCP.

> b) recommending server side DNS SRV separation, one prio per address family,
>     no dual stack hosts for UDP

If you use SRV to prioritize address families, then don't hosts that
can't access the destination with its preferred address family pay an
RTT penalty?  Or actually, a connection-timeout penalty?

Dale


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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
> I only skimmed this, but ISTM that the use of proxy-require with a new 
> option will make this nearly undeployable.

Let me explain the proposal more clearly.  The critical point is that it
is an *upward-compatible* way of sending simultaneous requests while
eliminating merged requests.  This is important for supporting Happy
Eyeballs with minimal delays to destinations that a device does not
maintain flows with.

Assume the device is sending this request that it received:

    INVITE sip:biloxi.example.com SIP/2.0
    Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
    Max-Forwards: 20
    To: Bob <sip:biloxi.example.com>
    From: Alice <sip:atlanta.example.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 314159 INVITE
    Contact: <sip:atlanta.example.com>

To the "first" target, 10.5.0.10, it sends this request, with particular
modifications to support merged requests:

    INVITE sip:biloxi.example.com SIP/2.0
    Route: <sip:10.5.0.10>;group=2
----^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
    Via: SIP/2.0/UDP 10.0.1.10;branch=z9hG4bKnashds8.1;group
------------------------------------------------------^^^^^^
    Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
    Max-Forwards: 19
    To: Bob <sip:biloxi.example.com>
    From: Alice <sip:atlanta.example.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 314159 INVITE
    Contact: <sip:atlanta.example.com>

The "group" parameter in the Route header indicates that this request is
part of a group.  The value of the group parameter indicates two
successive Via headers.  The branch parameter of the Via header indexed
by that value (counting from the bottom, starting with 1) indicates the
identity of this request within the group.  The branch parameter of the
next Via header down indicates the identity of the group.

If the target does not understand the group extension, it simply ignores
all these parameters.

In either case, the target deletes the Route header before forwarding
the request.

Simultaneously, to the "second" target, fd5a::0101, it sends this request:

    INVITE sip:biloxi.example.com SIP/2.0
    Proxy-Require: group
----^^^^^^^^^^^^^^^^^^^^
    Route: <sip:[fd5a::0101]>;group=2
----^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
    Via: SIP/2.0/UDP [fd5a::1000];branch=z9hG4bKnashds8.2;group
---------------------------------------------------------^^^^^^
    Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
    Max-Forwards: 19
    To: Bob <sip:biloxi.example.com>
    From: Alice <sip:atlanta.example.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 314159 INVITE
    Contact: <sip:atlanta.example.com>

This request is similar to the previous one, but it had a unique branch
parameter and includes "Proxy-Require: group".

If the second target (which is presumably the same devide as the first
target) does not implement the group extension, it rejects this request
with a 420 error.  The sending device will then postpone using this
target unless and until the request to the first target fails, at which
time it will send a request like the request to the first target, or a
request which does not use the group extension.

If the second target *does* implement the group extension, it
understands that the two requests are "equivalent", in that processing
either one has "equivalent effect" (from the user's point of view), and
that it should accept and process one of them and reject the other with
an error (495 Grouping).  Detecting whether or not another request has
already been processed is straightforward because it can search for an
ongoing transaction whose *second* branch parameter is the same as the
second branch parameter of this request.

If the second target processes this request, it deletes the the Route
header and the Proxy-Require header (per the specification of the
"group" extension) before forwarding the request.

This scheme has the advantage of minimizing the amount of state that the
devices need to maintain, which makes implementation much easier than,
say, SIP Outbound.

Dale


From nobody Tue Mar  1 09:27:18 2016
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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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On 3/1/16 11:31 AM, Dale R. Worley wrote:
> Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
>> Why do we need to have a solution for UDP?
>
> My experience is that UDP is universal, so if there's a straightforward
> way to do it in UDP, it will make implementation much quicker.
>
> In the SIPit 31 report, I see:
>
>      Implementations using each transport for SIP messages:
>         UDP   100%
>         TCP    86%
>         TLS    81% (20% server-auth-only)
>         SCTP   10%
>         DTLS    5%

The TCP and TLS numbers are now getting pretty big. Of course UPD is 
100%, since almost everybody has code in for backward compatibility.

Won't the TCP numbers get more favorable if we focus on applications 
that also support IPv6 and then also require implementation of a new Via 
parameter?

Wouldn't those who are doing that much new development also just bite 
the bullet and use TCP, thus avoiding the need for the new Via parameter?

	Thanks,
	Paul


From nobody Tue Mar  1 09:35:41 2016
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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Comment at end.

On 3/1/16 11:54 AM, Dale R. Worley wrote:
> Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
>> I only skimmed this, but ISTM that the use of proxy-require with a new
>> option will make this nearly undeployable.
>
> Let me explain the proposal more clearly.  The critical point is that it
> is an *upward-compatible* way of sending simultaneous requests while
> eliminating merged requests.  This is important for supporting Happy
> Eyeballs with minimal delays to destinations that a device does not
> maintain flows with.
>
> Assume the device is sending this request that it received:
>
>      INVITE sip:biloxi.example.com SIP/2.0
>      Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
>      Max-Forwards: 20
>      To: Bob <sip:biloxi.example.com>
>      From: Alice <sip:atlanta.example.com>;tag=1928301774
>      Call-ID: a84b4c76e66710
>      CSeq: 314159 INVITE
>      Contact: <sip:atlanta.example.com>
>
> To the "first" target, 10.5.0.10, it sends this request, with particular
> modifications to support merged requests:
>
>      INVITE sip:biloxi.example.com SIP/2.0
>      Route: <sip:10.5.0.10>;group=2
> ----^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>      Via: SIP/2.0/UDP 10.0.1.10;branch=z9hG4bKnashds8.1;group
> ------------------------------------------------------^^^^^^
>      Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
>      Max-Forwards: 19
>      To: Bob <sip:biloxi.example.com>
>      From: Alice <sip:atlanta.example.com>;tag=1928301774
>      Call-ID: a84b4c76e66710
>      CSeq: 314159 INVITE
>      Contact: <sip:atlanta.example.com>
>
> The "group" parameter in the Route header indicates that this request is
> part of a group.  The value of the group parameter indicates two
> successive Via headers.  The branch parameter of the Via header indexed
> by that value (counting from the bottom, starting with 1) indicates the
> identity of this request within the group.  The branch parameter of the
> next Via header down indicates the identity of the group.
>
> If the target does not understand the group extension, it simply ignores
> all these parameters.
>
> In either case, the target deletes the Route header before forwarding
> the request.
>
> Simultaneously, to the "second" target, fd5a::0101, it sends this request:
>
>      INVITE sip:biloxi.example.com SIP/2.0
>      Proxy-Require: group
> ----^^^^^^^^^^^^^^^^^^^^
>      Route: <sip:[fd5a::0101]>;group=2
> ----^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>      Via: SIP/2.0/UDP [fd5a::1000];branch=z9hG4bKnashds8.2;group
> ---------------------------------------------------------^^^^^^
>      Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
>      Max-Forwards: 19
>      To: Bob <sip:biloxi.example.com>
>      From: Alice <sip:atlanta.example.com>;tag=1928301774
>      Call-ID: a84b4c76e66710
>      CSeq: 314159 INVITE
>      Contact: <sip:atlanta.example.com>
>
> This request is similar to the previous one, but it had a unique branch
> parameter and includes "Proxy-Require: group".
>
> If the second target (which is presumably the same devide as the first
> target) does not implement the group extension, it rejects this request
> with a 420 error.  The sending device will then postpone using this
> target unless and until the request to the first target fails, at which
> time it will send a request like the request to the first target, or a
> request which does not use the group extension.
>
> If the second target *does* implement the group extension, it
> understands that the two requests are "equivalent", in that processing
> either one has "equivalent effect" (from the user's point of view), and
> that it should accept and process one of them and reject the other with
> an error (495 Grouping).  Detecting whether or not another request has
> already been processed is straightforward because it can search for an
> ongoing transaction whose *second* branch parameter is the same as the
> second branch parameter of this request.
>
> If the second target processes this request, it deletes the the Route
> header and the Proxy-Require header (per the specification of the
> "group" extension) before forwarding the request.
>
> This scheme has the advantage of minimizing the amount of state that the
> devices need to maintain, which makes implementation much easier than,
> say, SIP Outbound.

My concern is that *every* proxy on the path needs to support the 
'group' option for the second message to be successful. Not just the the 
one nearest the target. That makes it a high bar.

Otherwise what you describe does seem quite clever.

	Thanks,
	Paul


From nobody Tue Mar  1 10:41:16 2016
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Subject: Re: [sipcore] Progressing draft-ietf-sipcore-dns-dual-stack
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On 3/1/16 10:23, Dale R. Worley wrote:
> Since there seems to be no controversy, I'm planning on asking for a
> WGLC on 8 Mar 2016, which is two weeks after
> draft-ietf-sipcore-dns-dual-stack-04 was submitted.
>
> Alternatively, we could ask for time in Buenos Aires to discuss it.
>

[as chair]

I think what I'd like to do is a little of both.

This is a relatively important issue, and we've requested time for it in 
Buenos Aires just to make sure that it gets the attention that it 
deserves. At the same time, we'd like to see it move forward with all 
due speed. So I propose that we call for a three-week WGLC, starting on 
March 21st, and running through the Friday of the IETF meeting (April 8th).

If no substantial issues are raised on-list or at the meeting, then this 
gives us a clear signal that the document is ready to request 
publication. It also ensures that we have one final face-to-face meeting 
to hash out any potential issues or concerns.

Sound good?

/a


From nobody Tue Mar  1 21:03:10 2016
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [sipcore] Progressing draft-ietf-sipcore-dns-dual-stack
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From nobody Wed Mar  2 07:41:30 2016
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Subject: Re: [sipcore] Progressing draft-ietf-sipcore-dns-dual-stack
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Adam Roach <adam@nostrum.com> writes:
> I think what I'd like to do is a little of both.
>
> This is a relatively important issue, and we've requested time for it in 
> Buenos Aires just to make sure that it gets the attention that it 
> deserves. At the same time, we'd like to see it move forward with all 
> due speed. So I propose that we call for a three-week WGLC, starting on 
> March 21st, and running through the Friday of the IETF meeting (April 8th).
>
> If no substantial issues are raised on-list or at the meeting, then this 
> gives us a clear signal that the document is ready to request 
> publication. It also ensures that we have one final face-to-face meeting 
> to hash out any potential issues or concerns.
>
> Sound good?

Make sense to me.

Dale


From nobody Wed Mar  2 11:22:20 2016
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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
> My concern is that *every* proxy on the path needs to support the 
> 'group' option for the second message to be successful. Not just the the 
> one nearest the target. That makes it a high bar.

I believe that is not the case.  The Proxy-Require is inserted by the
sending device that we are describing, but it is *removed* by the
receiving device at the end of the hop.  At least, that's what I
intended to say:

> If the second target processes this request, it deletes the the Route
> header and the Proxy-Require header (per the specification of the
> "group" extension) before forwarding the request.

Am I overlooking something?

BTW, there's an error in my message:

>      Via: SIP/2.0/UDP [fd5a::1000];branch=z9hG4bKnashds8.2;group

When IPv6 addresses appear as the transport address in a Via header,
they do *not* appear inside [...].  [...] is only added when the IPv6
address appears as part of a SIP URI.

Dale


From nobody Wed Mar  2 11:42:11 2016
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On 3/2/16 2:22 PM, Dale R. Worley wrote:
> Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
>> My concern is that *every* proxy on the path needs to support the
>> 'group' option for the second message to be successful. Not just the the
>> one nearest the target. That makes it a high bar.
>
> I believe that is not the case.  The Proxy-Require is inserted by the
> sending device that we are describing, but it is *removed* by the
> receiving device at the end of the hop.  At least, that's what I
> intended to say:
>
>> If the second target processes this request, it deletes the the Route
>> header and the Proxy-Require header (per the specification of the
>> "group" extension) before forwarding the request.
>
> Am I overlooking something?

This assumes that the proxy that needs to do the merging is the next 
hop. Why is that the case? Couldn't there be intermediate proxies?

If that isn't a concern then I am less concerned.

	Thanks,
	Paul

> BTW, there's an error in my message:
>
>>       Via: SIP/2.0/UDP [fd5a::1000];branch=z9hG4bKnashds8.2;group
>
> When IPv6 addresses appear as the transport address in a Via header,
> they do *not* appear inside [...].  [...] is only added when the IPv6
> address appears as part of a SIP URI.
>
> Dale
>


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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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Hi,

> BTW, there's an error in my message:
>
> >      Via: SIP/2.0/UDP [fd5a::1000];branch=z9hG4bKnashds8.2;group
>
> When IPv6 addresses appear as the transport address in
> a Via header, they do *not* appear inside [...].
> [...] is only added when the IPv6 address appears as
> part of a SIP URI.

Using the brackets within Via's sent-by is correct; however the brackets
would not be added as part of Via's received parameter value.

The following ABNF is from RFC 3261 (some aspects were changed by RFC
5954).

Via               =  ( "Via" / "v" ) HCOLON via-parm *(COMMA via-parm)
via-parm          =  sent-protocol LWS sent-by *( SEMI via-params )
via-params        =  via-ttl / via-maddr
                     / via-received / via-branch
                     / via-extension
via-received      =  "received" EQUAL (IPv4address / IPv6address)
sent-by           =  host [ COLON port ]
host             =  hostname / IPv4address / IPv6reference
IPv6reference  =  "[" IPv6address "]"
IPv6address    =  hexpart [ ":" IPv4address ]


From nobody Wed Mar  2 12:31:22 2016
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Happy Eyeballs -- UDP and RTT
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Happy Eyeballs has a mess of issues surrounding UDP.

Of course, the most desirable SIP transport is a TCP/TLS connection (or
a SIP Outbound UDP flow) that is already established with the
destination.  That allows the message to be transmitted in 1/2 RTT in
the common case, and the success of that particular transport is largely
assured.  But it requires that the sender has recently communicated with
the destination.

In general, UDP is undesirable.  OTOH, it is the most implemented
transport for SIP and its implementation is relatively simple.  Even in
2014, 15% of the SIP implementations at SIPit do not support TCP.  UDP
has the advantage of usually getting a message to its destination in 1/2
RTT, even if the sender has not previously communicated with the
destination.

Typically, TCP implementations have 3/2 RTT transmission time, since
data is not sent in the first packet from the initiator.  TCP Fast Open
(RFC 7413) can be used to shorten that time to 1/2 RTT, but it requires
that the sender has recently communicated with the destination.

Using TLS in a new connection adds a further RTT to TCP transmission
times.

Then again, perhaps we do not care if the message is delayed by an
additional RTT.  RFC 6555, which is really about the HTTP server case,
assumes that 3/2 RTT is acceptable.  OTOH, HTTP service already has to
tolerate 3/2 RTT.

With UDP and TCP Fast Open, if transmission time is important, it would
be useful to be able to simultaneously transmit duplicate requests --
"Simultaneously" in the sense that a second request is sent before the
failure of a first request is known, and "duplicate" in the sense that
the success of either request obtains the desired service.  In order to
support this, there needs to be a way to detect and eliminate duplicate
requests at some downstream point.


BTW, looking at RFC 6555 section 4, I see:

   4.  Algorithm Requirements

   A "Happy Eyeballs" algorithm has two primary goals:

   1.  Provides fast connection for users, by quickly attempting to
       connect using IPv6 and (if that connection attempt is not quickly
       successful) to connect using IPv4.
   [...]
   The basic idea is depicted in the following diagram:

           DNS Server                  Client                  Server
               |                          |                       |
         1.    |<--www.example.com A?-----|                       |
         2.    |<--www.example.com AAAA?--|                       |
         3.    |---192.0.2.1------------->|                       |
         4.    |---2001:db8::1----------->|                       |
         5.    |                          |                       |
         6.    |                          |==TCP SYN, IPv6===>X   |
         7.    |                          |--TCP SYN, IPv4------->|
         8.    |                          |<-TCP SYN+ACK, IPv4----|
         9.    |                          |--TCP ACK, IPv4------->|
        10.    |                          |==TCP SYN, IPv6===>X   |

               Figure 2: Happy Eyeballs Flow 1, IPv6 Broken

   In the diagram above, the client sends two TCP SYNs at the same time
   over IPv6 (6) and IPv4 (7).  In the diagram, the IPv6 path is broken
   but has little impact to the user because there is no long delay
   before using IPv4.  [...]

Am I misunderstanding this?  Item (1) says that there is a delay between
initiating an IPv6 connection and initiating an IPv4 connection, but the
text at the end says that they are initiated simultaneously.

Dale


From nobody Wed Mar  2 13:30:59 2016
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To: "Dale R. Worley" <worley@ariadne.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Happy Eyeballs -- UDP and RTT
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Hi Dale,

>Of course, the most desirable SIP transport is a TCP/TLS connection (or a =
SIP Outbound UDP flow) that is already established with the destination. =20

Not sure what you mean by "destination", but an outbound flow is only creat=
ed between a UA and its registrar.

Regards,

Christer


From nobody Thu Mar  3 07:35:28 2016
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Subject: Re: [sipcore] Happy Eyeballs -- UDP and RTT
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Christer Holmberg <christer.holmberg@ericsson.com> writes:
>>Of course, the most desirable SIP transport is a TCP/TLS connection
>>(or a SIP Outbound UDP flow) that is already established with the
>>destination.  
>
> Not sure what you mean by "destination", but an outbound flow is only
> created between a UA and its registrar.

By "destination" I mean "where the device is trying to send the
message".

Dale


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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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Brett Tate <brett@broadsoft.com> writes:
>> BTW, there's an error in my message:
>>
>> >      Via: SIP/2.0/UDP [fd5a::1000];branch=z9hG4bKnashds8.2;group
>>

> Using the brackets within Via's sent-by is correct; however the brackets
> would not be added as part of Via's received parameter value.

Argh, yes, my mistake.  (I forgot that sent-by is a *host*, not an
address.  Despite that many, many SIP devices can't handle a host name
for a sent-by.)

Dale


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Paul Kyzivat <pkyzivat@alum.mit.edu> writes:
> This assumes that the proxy that needs to do the merging is the next 
> hop. Why is that the case? Couldn't there be intermediate proxies?
>
> If that isn't a concern then I am less concerned.

I'm assuming that the Happy Eyeballs case is one where all of the target
addresses go to the same device, or to one of a set of tightly-coupled
devices.  In that sort of situation, the merging can be done at the
target(s), and the Proxy-Require device works.

Originally, I looked at the merging problem in the context of having
multiple routes to a distant destination, with intermediate proxies.
E.g., A wants to send two copies of a request to C, one copy via B1 and
one copy via B2.  The best solution I've found for that case is to send:

    INVITE sip:biloxi.example.com SIP/2.0
    Route: <sip:B1;lr>
    Route: <sip:C;lr>;group=2
    Via: SIP/2.0/UDP 10.0.1.10;branch=z9hG4bKnashds8.1;group
    Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
    Max-Forwards: 19
    To: Bob <sip:biloxi.example.com>
    From: Alice <sip:atlanta.example.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 314159 INVITE
    Contact: <sip:atlanta.example.com>

    INVITE sip:biloxi.example.com SIP/2.0
    Route: <sip:B2;lr>
    Route: <sip:C;lr>;group=2
    Route: <sip:0.0.0.0;lr>
    Via: SIP/2.0/UDP 10.0.1.10;branch=z9hG4bKnashds8.2;group
    Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
    Max-Forwards: 19
    To: Bob <sip:biloxi.example.com>
    From: Alice <sip:atlanta.example.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 314159 INVITE
    Contact: <sip:atlanta.example.com>

In this version of the mechanism, when either request arrives at C, it
sees the group parameter on its Route header, which notifies it to
examine the indicated Via headers.  If C does not support the group
mechanism, it does not know to ignore and delete the "Route:
<sip:0.0.0.0;lr>" header, so that copy of the request will error out.
If C does support the group mechanism, it deletes "Route:
<sip:0.0.0.0;lr>" and either responds 495 or routes the request to
bilozi.example.com.

In either version, the copies of the request have to be routed to a
merge point that can correlate all SIP requests that arrive and remove
duplicates.  I'm trying to figure out whether it's possible to remove
that restriction, and instead depend on/allow further downstream devices
to notice the group parameter on the Vias to detect pairs of requests
that come from different branches of a split.  If I am lucky (and I
don't think I am), this could all be put into the merged requests logic
for UASs.

Also, "lr" should be a header field parameter, not a URI parameter.  Too
late to change that, though.

Dale


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From: Roman Shpount <roman@telurix.com>
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Subject: [sipcore] Happy Eyeballs and UDP
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Since everyone is talking about how desirable TLS is and how easy it is to
deprecate UDP, can anybody point me to large scale implementations of
TLS+Outbound? I know a lot of implementations for UDP which handle 1M+ of
concurrent registrations. When someone tries to implement the same using
TLS and Outbound it much harder:

1. Maintaining large number of TLS connections is resource intensive. If
you have 1M concurrent registrations you need edge proxies which will
terminate 1M concurrent TLS connections and respond to keep alive messages.
This requires 10x number the servers then UDP registrations. This also
exposes you to more risk of DOS attacks

2. Fail-over is much harder: For TLS/Outbound you either need to put edge
proxy on VM that you can migrate to a different server (still had to deal
with software failure). To deal with software failures, you need to design
the mechanism for the client to recover the connection to a different edge
proxy. If call is in progress this involves for the client detecting the
flow failure based on keep alive failure, re-registering to get the gruu,
and re-INVITE with new gruu in the contact. If server needs to send a
message on the client call while this is happening, it needs to queue this
message or implement some sort of non-standard re-transmit timer. This
typically means call gets dropped. There is no mechanism at all to recover
the transaction which is in progress while outbound proxy fails. For UDP
you can move the IP address for the edge proxy to a different server. If
registration and flow state is in some sort of shared DB, the fail-over
occurs naturally for both the client and server.

I went through couple of large scale deployments of Outbound with SIPS for
large number of consumer devices. In all of them SIP registrations were
scrapped and replaced with something proprietary.

So, if we are talking about SIP for server to server communications, SIPS
is great but it does not need Happy Eyeballs since everything can be
provisioned in advance.

If we are dealing with deployments that deal with large end user devices
(which is where you need Happy Eyeballs) then a lot of things need to be
fixed in SIP Outbound before UDP can be discarded.

Regards,
_____________
Roman Shpount

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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Since everyone is talking about how desirable TLS is and h=
ow easy it is to deprecate UDP, can anybody point me to large scale impleme=
ntations of TLS+Outbound? I know a lot of implementations for UDP which han=
dle 1M+ of concurrent registrations. When someone tries to implement the sa=
me using TLS and Outbound it much harder:<div><br></div><div>1. Maintaining=
 large number of TLS connections is resource intensive. If you have 1M conc=
urrent registrations you need edge proxies which will terminate 1M concurre=
nt TLS connections and respond to keep alive messages. This requires 10x nu=
mber the servers then UDP registrations. This also exposes you to more risk=
 of DOS attacks</div><div><br></div><div>2. Fail-over is much harder: For T=
LS/Outbound you either need to put edge proxy on VM that you can migrate to=
 a different server (still had to deal with software failure). To deal with=
 software failures, you need to design the mechanism for the client to reco=
ver the connection to a different edge proxy. If call is in progress this i=
nvolves for the client detecting the flow failure based on keep alive failu=
re, re-registering to get the gruu, and re-INVITE with new gruu in the cont=
act. If server needs to send a message on the client call while this is hap=
pening, it needs to queue this message or implement some sort of non-standa=
rd re-transmit timer. This typically means call gets dropped. There is no m=
echanism at all to recover the transaction which is in progress while outbo=
und proxy fails. For UDP you can move the IP address for the edge proxy to =
a different server. If registration and flow state is in some sort of share=
d DB, the fail-over occurs naturally for both the client and server.</div><=
div><br></div><div>I went through couple of large scale deployments of Outb=
ound with SIPS for large number of consumer devices. In all of them SIP reg=
istrations were scrapped and replaced with something proprietary.</div><div=
><br></div><div>So, if we are talking about SIP for server to server commun=
ications, SIPS is great but it does not need Happy Eyeballs since everythin=
g can be provisioned in advance.</div><div><br></div><div>If we are dealing=
 with deployments that deal with large end user devices (which is where you=
 need Happy Eyeballs) then a lot of things need to be fixed in SIP Outbound=
 before UDP can be discarded.</div><div><br></div><div>Regards,</div><div><=
div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></di=
v>
</div></div>

--001a1140d71e3104ab052d2ad9f9--


From nobody Thu Mar  3 12:38:32 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Happy Eyeballs -- UDP and RTT
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Subject: Re: [sipcore] Happy Eyeballs -- UDP and RTT
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Hi,

>>>Of course, the most desirable SIP transport is a TCP/TLS connection=20
>>>(or a SIP Outbound UDP flow) that is already established with the=20
>>>destination.
>>
>> Not sure what you mean by "destination", but an outbound flow is only=20
>> created between a UA and its registrar.
>
> By "destination" I mean "where the device is trying to send the message".

If you refer to the remote UA, there is no outbound flow established with t=
he UAs. Again, the outbound flow is between a UA and its registrar.

Regards,

Christer


From nobody Thu Mar  3 14:11:58 2016
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Happy Eyeballs and UDP
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On 3/3/16 3:28 PM, Roman Shpount wrote:
> Since everyone is talking about how desirable TLS is and how easy it is
> to deprecate UDP, can anybody point me to large scale implementations of
> TLS+Outbound? I know a lot of implementations for UDP which handle 1M+
> of concurrent registrations. When someone tries to implement the same
> using TLS and Outbound it much harder:
>
> 1. Maintaining large number of TLS connections is resource intensive. If
> you have 1M concurrent registrations you need edge proxies which will
> terminate 1M concurrent TLS connections and respond to keep alive
> messages. This requires 10x number the servers then UDP registrations.
> This also exposes you to more risk of DOS attacks
>
> 2. Fail-over is much harder: For TLS/Outbound you either need to put
> edge proxy on VM that you can migrate to a different server (still had
> to deal with software failure). To deal with software failures, you need
> to design the mechanism for the client to recover the connection to a
> different edge proxy. If call is in progress this involves for the
> client detecting the flow failure based on keep alive failure,
> re-registering to get the gruu, and re-INVITE with new gruu in the
> contact. If server needs to send a message on the client call while this
> is happening, it needs to queue this message or implement some sort of
> non-standard re-transmit timer. This typically means call gets dropped.
> There is no mechanism at all to recover the transaction which is in
> progress while outbound proxy fails. For UDP you can move the IP address
> for the edge proxy to a different server. If registration and flow state
> is in some sort of shared DB, the fail-over occurs naturally for both
> the client and server.

Isn't that the point of multiple flows in outbound?
(Of course that multiplies the number of registrations and TLS connections.)

> I went through couple of large scale deployments of Outbound with SIPS
> for large number of consumer devices. In all of them SIP registrations
> were scrapped and replaced with something proprietary.
>
> So, if we are talking about SIP for server to server communications,
> SIPS is great but it does not need Happy Eyeballs since everything can
> be provisioned in advance.

If we assume every end device has an outbound proxy, does that mean you 
are saying there is no problem for SIP Happy Eyeballs to solve?

> If we are dealing with deployments that deal with large end user devices
> (which is where you need Happy Eyeballs) then a lot of things need to be
> fixed in SIP Outbound before UDP can be discarded.

What do you mean by "large end user devices"? Do you mean enterprise SIP 
servers, such as for IVR or call center? Or enterprise SIP PBXs?

	Thanks,
	Paul


From nobody Fri Mar  4 14:02:41 2016
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On Thu, 3 Mar 2016 17:11:53 -0500, Paul Kyzivat <pkyzivat@alum.mit.edu
> wrote:

> On 3/3/16 3:28 PM, Roman Shpount wrote:
> > 2. Fail-over is much harder: For TLS/Outbound you either need to put
> > edge proxy on VM that you can migrate to a different server (still had
> > to deal with software failure). To deal with software failures, you need
> > to design the mechanism for the client to recover the connection to a
> > different edge proxy. If call is in progress this involves for the
> > client detecting the flow failure based on keep alive failure,
> > re-registering to get the gruu, and re-INVITE with new gruu in the
> > contact. If server needs to send a message on the client call while this
> > is happening, it needs to queue this message or implement some sort of
> > non-standard re-transmit timer. This typically means call gets dropped.
> > There is no mechanism at all to recover the transaction which is in
> > progress while outbound proxy fails. For UDP you can move the IP address
> > for the edge proxy to a different server. If registration and flow state
> > is in some sort of shared DB, the fail-over occurs naturally for both
> > the client and server.
>
> Isn't that the point of multiple flows in outbound?
> (Of course that multiplies the number of registrations and TLS
> connections.)
>

Multiple flows somewhat mitigate the problem of edge proxy failure at the
cost of multiplying the cost of connection management. There is still and
issue of temporary network failure between the end point and the proxy. In
case of TLS this terminates all flows from the client to all proxies and
typically terminates the call. And then there is a use case of end point
migrating from one IP to another. Here the quicker flow recovery with UDP
increases the chances of keeping the call running vs 7 or 8 round trips
required to recover the TLS flow (TCP connection setup, TLS connection
setup, registration with response and then re-invite with new GRUU). And
then there is a use case when client used a temporary GRUU which was flow
specific. And there is still no way to recover the transaction since, I
believe, transactions are flow bound.


> > I went through couple of large scale deployments of Outbound with SIPS
> > for large number of consumer devices. In all of them SIP registrations
> > were scrapped and replaced with something proprietary.
> >
> > So, if we are talking about SIP for server to server communications,
> > SIPS is great but it does not need Happy Eyeballs since everything can
> > be provisioned in advance.
>
> If we assume every end device has an outbound proxy, does that mean you
> are saying there is no problem for SIP Happy Eyeballs to solve?
>

I meant that if SIP is used for server to server interconnects. These are
interconnects between devices with capabilities well known in advance
located on well known addresses. When you are interconnecting two service
providers you typically know in advance if IPv6 is used, so there is no
need for Happy Eyeballs.

> If we are dealing with deployments that deal with large end user devices
> > (which is where you need Happy Eyeballs) then a lot of things need to be
> > fixed in SIP Outbound before UDP can be discarded.
>
> What do you mean by "large end user devices"? Do you mean enterprise SIP
> servers, such as for IVR or call center? Or enterprise SIP PBXs?
>

It was a typing mistake. I meant "deployments that deal with large numbers
of end user devices"

Happy Eyeballs is needed to address situation where you have large number
of end user devices located on multiple customer networks with different
capabilities trying to connect to a central public service which is
accessible over both IPv4 and IPv6. The point I am trying to make that all
of such SIP services primarily use UDP so it is too early to discard it.

_____________
Roman Shpount

--089e011605a2b1d7d3052d4045a8
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On=C2=A0Thu, 3 Mar 2016 17:11:53 -0500,=C2=A0Paul Kyzivat &lt;<a href=
=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;=C2=A0wrote:=
<br></div></div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color=
:rgb(204,204,204);border-left-style:solid;padding-left:1ex">On 3/3/16 3:28 =
PM, Roman Shpount wrote:<br>&gt; 2. Fail-over is much harder: For TLS/Outbo=
und you either need to put<br>
&gt; edge proxy on VM that you can migrate to a different server (still had=
<br>
&gt; to deal with software failure). To deal with software failures, you ne=
ed<br>
&gt; to design the mechanism for the client to recover the connection to a<=
br>
&gt; different edge proxy. If call is in progress this involves for the<br>
&gt; client detecting the flow failure based on keep alive failure,<br>
&gt; re-registering to get the gruu, and re-INVITE with new gruu in the<br>
&gt; contact. If server needs to send a message on the client call while th=
is<br>
&gt; is happening, it needs to queue this message or implement some sort of=
<br>
&gt; non-standard re-transmit timer. This typically means call gets dropped=
.<br>
&gt; There is no mechanism at all to recover the transaction which is in<br=
>
&gt; progress while outbound proxy fails. For UDP you can move the IP addre=
ss<br>
&gt; for the edge proxy to a different server. If registration and flow sta=
te<br>
&gt; is in some sort of shared DB, the fail-over occurs naturally for both<=
br>
&gt; the client and server.<br>
<br>
Isn&#39;t that the point of multiple flows in outbound?<br>
(Of course that multiplies the number of registrations and TLS connections.=
)<br></blockquote><div><br></div><div>Multiple flows somewhat mitigate the =
problem of edge proxy failure at the cost of multiplying the cost of connec=
tion management. There is still and issue of temporary network failure betw=
een the end point and the proxy. In case of TLS this terminates all flows f=
rom the client to all proxies and typically terminates the call. And then t=
here is a use case of end point migrating from one IP to another. Here the =
quicker flow recovery with UDP increases the chances of keeping the call ru=
nning vs 7 or 8 round trips required to recover the TLS flow (TCP connectio=
n setup, TLS connection setup, registration with response and then re-invit=
e with new GRUU). And then there is a use case when client used a temporary=
 GRUU which was flow specific. And there is still no way to recover the tra=
nsaction since, I believe, transactions are flow bound.</div><div>=C2=A0</d=
iv><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bord=
er-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:soli=
d;padding-left:1ex">&gt; I went through couple of large scale deployments o=
f Outbound with SIPS<br>
&gt; for large number of consumer devices. In all of them SIP registrations=
<br>
&gt; were scrapped and replaced with something proprietary.<br>
&gt;<br>
&gt; So, if we are talking about SIP for server to server communications,<b=
r>
&gt; SIPS is great but it does not need Happy Eyeballs since everything can=
<br>
&gt; be provisioned in advance.<br>
<br>
If we assume every end device has an outbound proxy, does that mean you<br>
are saying there is no problem for SIP Happy Eyeballs to solve?<br></blockq=
uote><div>=C2=A0</div><div>I meant that if SIP is used for server to server=
 interconnects. These are interconnects between devices with capabilities w=
ell known in advance located on well known addresses. When you are intercon=
necting two service providers you typically know in advance if IPv6 is used=
, so there is no need for Happy Eyeballs.</div><div><br></div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1p=
x;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1=
ex">&gt; If we are dealing with deployments that deal with large end user d=
evices<br>
&gt; (which is where you need Happy Eyeballs) then a lot of things need to =
be<br>
&gt; fixed in SIP Outbound before UDP can be discarded.<br>
<br>
What do you mean by &quot;large end user devices&quot;? Do you mean enterpr=
ise SIP<br>
servers, such as for IVR or call center? Or enterprise SIP PBXs?<br></block=
quote><div><br></div><div>It was a typing mistake. I meant &quot;deployment=
s that deal with large numbers of end user devices&quot;</div><div><br></di=
v><div><div>Happy Eyeballs is needed to address situation where you have la=
rge number of end user devices located on multiple customer networks with d=
ifferent capabilities trying to connect to a central public service which i=
s accessible over both IPv4 and IPv6. The point I am trying to make that al=
l of such SIP services primarily use UDP so it is too early to discard it.<=
/div></div><div><br></div><div><div class=3D"gmail_signature">_____________=
<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--089e011605a2b1d7d3052d4045a8--


From nobody Mon Mar  7 09:32:13 2016
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To: Roman Shpount <roman@telurix.com>, sipcore@ietf.org
References: <mailman.19.1457121604.20561.sipcore@ietf.org> <CAD5OKxuej3cmuna_T9f80ZdCb5junGa+NaTNDG0epyGhFQ+FLQ@mail.gmail.com>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Mon, 7 Mar 2016 12:32:00 -0500
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Subject: [sipcore] SIP TLS issues
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Inline.

On 3/4/16 5:02 PM, Roman Shpount wrote:
> On Thu, 3 Mar 2016 17:11:53 -0500, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     On 3/3/16 3:28 PM, Roman Shpount wrote:
>      > 2. Fail-over is much harder: For TLS/Outbound you either need to put
>      > edge proxy on VM that you can migrate to a different server
>     (still had
>      > to deal with software failure). To deal with software failures,
>     you need
>      > to design the mechanism for the client to recover the connection to a
>      > different edge proxy. If call is in progress this involves for the
>      > client detecting the flow failure based on keep alive failure,
>      > re-registering to get the gruu, and re-INVITE with new gruu in the
>      > contact. If server needs to send a message on the client call
>     while this
>      > is happening, it needs to queue this message or implement some
>     sort of
>      > non-standard re-transmit timer. This typically means call gets
>     dropped.
>      > There is no mechanism at all to recover the transaction which is in
>      > progress while outbound proxy fails. For UDP you can move the IP
>     address
>      > for the edge proxy to a different server. If registration and
>     flow state
>      > is in some sort of shared DB, the fail-over occurs naturally for both
>      > the client and server.
>
>     Isn't that the point of multiple flows in outbound?
>     (Of course that multiplies the number of registrations and TLS
>     connections.)
>
>
> Multiple flows somewhat mitigate the problem of edge proxy failure at
> the cost of multiplying the cost of connection management. There is
> still and issue of temporary network failure between the end point and
> the proxy. In case of TLS this terminates all flows from the client to
> all proxies and typically terminates the call. And then there is a use
> case of end point migrating from one IP to another. Here the quicker
> flow recovery with UDP increases the chances of keeping the call running
> vs 7 or 8 round trips required to recover the TLS flow (TCP connection
> setup, TLS connection setup, registration with response and then
> re-invite with new GRUU). And then there is a use case when client used
> a temporary GRUU which was flow specific. And there is still no way to
> recover the transaction since, I believe, transactions are flow bound.
>
>      > I went through couple of large scale deployments of Outbound with
>     SIPS
>      > for large number of consumer devices. In all of them SIP
>     registrations
>      > were scrapped and replaced with something proprietary.
>      >
>      > So, if we are talking about SIP for server to server communications,
>      > SIPS is great but it does not need Happy Eyeballs since
>     everything can
>      > be provisioned in advance.
>
>     If we assume every end device has an outbound proxy, does that mean you
>     are saying there is no problem for SIP Happy Eyeballs to solve?
>
> I meant that if SIP is used for server to server interconnects. These
> are interconnects between devices with capabilities well known in
> advance located on well known addresses. When you are interconnecting
> two service providers you typically know in advance if IPv6 is used, so
> there is no need for Happy Eyeballs.

Do you mean interconnect of service providers that have individual 
peering relationships?

What about interconnections between arbitrary enterprises? (Possibly big 
ones.) It seems like there are likely cases there they won't have 
individual agreements.

>      > If we are dealing with deployments that deal with large end user
>     devices
>      > (which is where you need Happy Eyeballs) then a lot of things
>     need to be
>      > fixed in SIP Outbound before UDP can be discarded.
>
>     What do you mean by "large end user devices"? Do you mean enterprise SIP
>     servers, such as for IVR or call center? Or enterprise SIP PBXs?
>
>
> It was a typing mistake. I meant "deployments that deal with large
> numbers of end user devices"
>
> Happy Eyeballs is needed to address situation where you have large
> number of end user devices located on multiple customer networks with
> different capabilities trying to connect to a central public service
> which is accessible over both IPv4 and IPv6. The point I am trying to
> make that all of such SIP services primarily use UDP so it is too early
> to discard it.

The obvious problems here are:

1) security
2) large sip messages and UDP message fragmentation

If only (1) were a problem, then maybe the answer would be SIP over DTLS.

But that has never happened. My impression is that it was deemed to not 
be worthwhile because of (2). That it people should jump right to TLS to 
solve both problems at once.

	Thanks,
	Paul


From nobody Mon Mar  7 09:57:47 2016
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From: Roman Shpount <roman@telurix.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIP TLS issues
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--089e01494ff2004529052d793246
Content-Type: text/plain; charset=UTF-8

On Mon, Mar 7, 2016 at 12:32 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Inline.
>
> On 3/4/16 5:02 PM, Roman Shpount wrote:
>
>> On Thu, 3 Mar 2016 17:11:53 -0500, Paul Kyzivat <pkyzivat@alum.mit.edu
>> <mailto:pkyzivat@alum.mit.edu>> wrote:
>>
>>     On 3/3/16 3:28 PM, Roman Shpount wrote:
>>      > 2. Fail-over is much harder: For TLS/Outbound you either need to
>> put
>>      > edge proxy on VM that you can migrate to a different server
>>     (still had
>>      > to deal with software failure). To deal with software failures,
>>     you need
>>      > to design the mechanism for the client to recover the connection
>> to a
>>      > different edge proxy. If call is in progress this involves for the
>>      > client detecting the flow failure based on keep alive failure,
>>      > re-registering to get the gruu, and re-INVITE with new gruu in the
>>      > contact. If server needs to send a message on the client call
>>     while this
>>      > is happening, it needs to queue this message or implement some
>>     sort of
>>      > non-standard re-transmit timer. This typically means call gets
>>     dropped.
>>      > There is no mechanism at all to recover the transaction which is in
>>      > progress while outbound proxy fails. For UDP you can move the IP
>>     address
>>      > for the edge proxy to a different server. If registration and
>>     flow state
>>      > is in some sort of shared DB, the fail-over occurs naturally for
>> both
>>      > the client and server.
>>
>>     Isn't that the point of multiple flows in outbound?
>>     (Of course that multiplies the number of registrations and TLS
>>     connections.)
>>
>>
>> Multiple flows somewhat mitigate the problem of edge proxy failure at
>> the cost of multiplying the cost of connection management. There is
>> still and issue of temporary network failure between the end point and
>> the proxy. In case of TLS this terminates all flows from the client to
>> all proxies and typically terminates the call. And then there is a use
>> case of end point migrating from one IP to another. Here the quicker
>> flow recovery with UDP increases the chances of keeping the call running
>> vs 7 or 8 round trips required to recover the TLS flow (TCP connection
>> setup, TLS connection setup, registration with response and then
>> re-invite with new GRUU). And then there is a use case when client used
>> a temporary GRUU which was flow specific. And there is still no way to
>> recover the transaction since, I believe, transactions are flow bound.
>>
>>      > I went through couple of large scale deployments of Outbound with
>>     SIPS
>>      > for large number of consumer devices. In all of them SIP
>>     registrations
>>      > were scrapped and replaced with something proprietary.
>>      >
>>      > So, if we are talking about SIP for server to server
>> communications,
>>      > SIPS is great but it does not need Happy Eyeballs since
>>     everything can
>>      > be provisioned in advance.
>>
>>     If we assume every end device has an outbound proxy, does that mean
>> you
>>     are saying there is no problem for SIP Happy Eyeballs to solve?
>>
>> I meant that if SIP is used for server to server interconnects. These
>> are interconnects between devices with capabilities well known in
>> advance located on well known addresses. When you are interconnecting
>> two service providers you typically know in advance if IPv6 is used, so
>> there is no need for Happy Eyeballs.
>>
>
> Do you mean interconnect of service providers that have individual peering
> relationships?
>
> What about interconnections between arbitrary enterprises? (Possibly big
> ones.) It seems like there are likely cases there they won't have
> individual agreements.
>

I meant most of the federation type interconnects, including service
provider to service provider, SIP trunks from service provider to
enterprise, and SIP trunks between enterprises. These as well as SIP IP
phones in enterprise are the main deployment scenarios for SIP.

     > If we are dealing with deployments that deal with large end user
>>     devices
>>      > (which is where you need Happy Eyeballs) then a lot of things
>>     need to be
>>      > fixed in SIP Outbound before UDP can be discarded.
>>
>>     What do you mean by "large end user devices"? Do you mean enterprise
>> SIP
>>     servers, such as for IVR or call center? Or enterprise SIP PBXs?
>>
>>
>> It was a typing mistake. I meant "deployments that deal with large
>> numbers of end user devices"
>>
>> Happy Eyeballs is needed to address situation where you have large
>> number of end user devices located on multiple customer networks with
>> different capabilities trying to connect to a central public service
>> which is accessible over both IPv4 and IPv6. The point I am trying to
>> make that all of such SIP services primarily use UDP so it is too early
>> to discard it.
>>
>
> The obvious problems here are:
>
> 1) security
> 2) large sip messages and UDP message fragmentation
>
> If only (1) were a problem, then maybe the answer would be SIP over DTLS.
>
> But that has never happened. My impression is that it was deemed to not be
> worthwhile because of (2). That it people should jump right to TLS to solve
> both problems at once.
>

DTLS is not an answer either since encryption state will need to be failed
over with the edge proxy. What you need is a protocol where client can
recover a flow to the server, including encryption, using a single round
trip.

I think deployments of large numbers of SIP devices connected to the public
service (like SIP based Skype) never happened and will never happen because
of the complexity and design issues of SIP Outbound in combination with
TLS. I think this is why none of the large scale consumer VoIP applications
used SIP. The companies that built these products used RTP, ICE and media
components, but SIP was ignored. I do not think all these companies spent
all the design effort to build something completely new just because they
wanted to build something different. From my experience with couple of
similar projects the issues of scaling the deployment and handling
redundancy were the main driving factors.

Putting aside all the problems with SIP Outbound, there are still a lot of
companies which offer SIP over UDP to the end users. I am talking about all
the hosted PBX solutions. Most of these deployments do not use ICE and use
host based NAT traversal for signaling and media to keep the message size
small. This is not a solution you would design if you have started the
design from scratch, but these installations definitely need a migration
path from IPv4 to IPv6. Telling them to switch to SIPS with SIP Outbound is
unrealistic, so I believe Happy Eyeballs must provide the solution for UDP.

Regards,
_____________
Roman Shpount

--089e01494ff2004529052d793246
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Mar 7, 2016 at 12:32 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.=
edu</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex">Inline.<br>
<br>
On 3/4/16 5:02 PM, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
On Thu, 3 Mar 2016 17:11:53 -0500, Paul Kyzivat &lt;<a href=3D"mailto:pkyzi=
vat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a><br>
&lt;mailto:<a href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzi=
vat@alum.mit.edu</a>&gt;&gt; wrote:<br>
<br>
=C2=A0 =C2=A0 On 3/3/16 3:28 PM, Roman Shpount wrote:<br>
=C2=A0 =C2=A0 =C2=A0&gt; 2. Fail-over is much harder: For TLS/Outbound you =
either need to put<br>
=C2=A0 =C2=A0 =C2=A0&gt; edge proxy on VM that you can migrate to a differe=
nt server<br>
=C2=A0 =C2=A0 (still had<br>
=C2=A0 =C2=A0 =C2=A0&gt; to deal with software failure). To deal with softw=
are failures,<br>
=C2=A0 =C2=A0 you need<br>
=C2=A0 =C2=A0 =C2=A0&gt; to design the mechanism for the client to recover =
the connection to a<br>
=C2=A0 =C2=A0 =C2=A0&gt; different edge proxy. If call is in progress this =
involves for the<br>
=C2=A0 =C2=A0 =C2=A0&gt; client detecting the flow failure based on keep al=
ive failure,<br>
=C2=A0 =C2=A0 =C2=A0&gt; re-registering to get the gruu, and re-INVITE with=
 new gruu in the<br>
=C2=A0 =C2=A0 =C2=A0&gt; contact. If server needs to send a message on the =
client call<br>
=C2=A0 =C2=A0 while this<br>
=C2=A0 =C2=A0 =C2=A0&gt; is happening, it needs to queue this message or im=
plement some<br>
=C2=A0 =C2=A0 sort of<br>
=C2=A0 =C2=A0 =C2=A0&gt; non-standard re-transmit timer. This typically mea=
ns call gets<br>
=C2=A0 =C2=A0 dropped.<br>
=C2=A0 =C2=A0 =C2=A0&gt; There is no mechanism at all to recover the transa=
ction which is in<br>
=C2=A0 =C2=A0 =C2=A0&gt; progress while outbound proxy fails. For UDP you c=
an move the IP<br>
=C2=A0 =C2=A0 address<br>
=C2=A0 =C2=A0 =C2=A0&gt; for the edge proxy to a different server. If regis=
tration and<br>
=C2=A0 =C2=A0 flow state<br>
=C2=A0 =C2=A0 =C2=A0&gt; is in some sort of shared DB, the fail-over occurs=
 naturally for both<br>
=C2=A0 =C2=A0 =C2=A0&gt; the client and server.<br>
<br>
=C2=A0 =C2=A0 Isn&#39;t that the point of multiple flows in outbound?<br>
=C2=A0 =C2=A0 (Of course that multiplies the number of registrations and TL=
S<br>
=C2=A0 =C2=A0 connections.)<br>
<br>
<br>
Multiple flows somewhat mitigate the problem of edge proxy failure at<br>
the cost of multiplying the cost of connection management. There is<br>
still and issue of temporary network failure between the end point and<br>
the proxy. In case of TLS this terminates all flows from the client to<br>
all proxies and typically terminates the call. And then there is a use<br>
case of end point migrating from one IP to another. Here the quicker<br>
flow recovery with UDP increases the chances of keeping the call running<br=
>
vs 7 or 8 round trips required to recover the TLS flow (TCP connection<br>
setup, TLS connection setup, registration with response and then<br>
re-invite with new GRUU). And then there is a use case when client used<br>
a temporary GRUU which was flow specific. And there is still no way to<br>
recover the transaction since, I believe, transactions are flow bound.<br>
<br>
=C2=A0 =C2=A0 =C2=A0&gt; I went through couple of large scale deployments o=
f Outbound with<br>
=C2=A0 =C2=A0 SIPS<br>
=C2=A0 =C2=A0 =C2=A0&gt; for large number of consumer devices. In all of th=
em SIP<br>
=C2=A0 =C2=A0 registrations<br>
=C2=A0 =C2=A0 =C2=A0&gt; were scrapped and replaced with something propriet=
ary.<br>
=C2=A0 =C2=A0 =C2=A0&gt;<br>
=C2=A0 =C2=A0 =C2=A0&gt; So, if we are talking about SIP for server to serv=
er communications,<br>
=C2=A0 =C2=A0 =C2=A0&gt; SIPS is great but it does not need Happy Eyeballs =
since<br>
=C2=A0 =C2=A0 everything can<br>
=C2=A0 =C2=A0 =C2=A0&gt; be provisioned in advance.<br>
<br>
=C2=A0 =C2=A0 If we assume every end device has an outbound proxy, does tha=
t mean you<br>
=C2=A0 =C2=A0 are saying there is no problem for SIP Happy Eyeballs to solv=
e?<br>
<br>
I meant that if SIP is used for server to server interconnects. These<br>
are interconnects between devices with capabilities well known in<br>
advance located on well known addresses. When you are interconnecting<br>
two service providers you typically know in advance if IPv6 is used, so<br>
there is no need for Happy Eyeballs.<br>
</blockquote>
<br>
Do you mean interconnect of service providers that have individual peering =
relationships?<br>
<br>
What about interconnections between arbitrary enterprises? (Possibly big on=
es.) It seems like there are likely cases there they won&#39;t have individ=
ual agreements.<br></blockquote><div><br></div><div>I meant most of the fed=
eration type interconnects, including service provider to service provider,=
 SIP trunks from service provider to enterprise, and SIP trunks between ent=
erprises. These as well as SIP IP phones in enterprise are the main deploym=
ent scenarios for SIP.=C2=A0</div><div><br></div><blockquote class=3D"gmail=
_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left=
-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-widt=
h:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-le=
ft:1ex">=C2=A0 =C2=A0 =C2=A0&gt; If we are dealing with deployments that de=
al with large end user<br>
=C2=A0 =C2=A0 devices<br>
=C2=A0 =C2=A0 =C2=A0&gt; (which is where you need Happy Eyeballs) then a lo=
t of things<br>
=C2=A0 =C2=A0 need to be<br>
=C2=A0 =C2=A0 =C2=A0&gt; fixed in SIP Outbound before UDP can be discarded.=
<br>
<br>
=C2=A0 =C2=A0 What do you mean by &quot;large end user devices&quot;? Do yo=
u mean enterprise SIP<br>
=C2=A0 =C2=A0 servers, such as for IVR or call center? Or enterprise SIP PB=
Xs?<br>
<br>
<br>
It was a typing mistake. I meant &quot;deployments that deal with large<br>
numbers of end user devices&quot;<br>
<br>
Happy Eyeballs is needed to address situation where you have large<br>
number of end user devices located on multiple customer networks with<br>
different capabilities trying to connect to a central public service<br>
which is accessible over both IPv4 and IPv6. The point I am trying to<br>
make that all of such SIP services primarily use UDP so it is too early<br>
to discard it.<br>
</blockquote>
<br>
The obvious problems here are:<br>
<br>
1) security<br>
2) large sip messages and UDP message fragmentation<br>
<br>
If only (1) were a problem, then maybe the answer would be SIP over DTLS.<b=
r>
<br>
But that has never happened. My impression is that it was deemed to not be =
worthwhile because of (2). That it people should jump right to TLS to solve=
 both problems at once.<br></blockquote><div><br></div><div>DTLS is not an =
answer either since encryption state will need to be failed over with the e=
dge proxy. What you need is a protocol where client can recover a flow to t=
he server, including encryption, using a single round trip.=C2=A0</div><div=
><br></div><div>I think deployments of large numbers of SIP devices connect=
ed to the public service (like SIP based Skype) never happened and will nev=
er happen because of the complexity and design issues of SIP Outbound in co=
mbination with TLS. I think this is why none of the large scale consumer Vo=
IP applications used SIP. The companies that built these products used RTP,=
 ICE and media components, but SIP was ignored. I do not think all these co=
mpanies spent all the design effort to build something completely new just =
because they wanted to build something different. From my experience with c=
ouple of similar projects the issues of scaling the deployment and handling=
 redundancy were the main driving factors.</div><div><br></div><div>Putting=
 aside all the problems with SIP Outbound, there are still a lot of compani=
es which offer SIP over UDP to the end users. I am talking about all the ho=
sted PBX solutions. Most of these deployments do not use ICE and use host b=
ased NAT traversal for signaling and media to keep the message size small. =
This is not a solution you would design if you have started the design from=
 scratch, but these installations definitely need a migration path from IPv=
4 to IPv6. Telling them to switch to SIPS with SIP Outbound is unrealistic,=
 so I believe Happy Eyeballs must provide the solution for UDP.</div><div><=
br></div><div>Regards,</div><div><div class=3D"gmail_signature">___________=
__<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--089e01494ff2004529052d793246--


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Subject: Re: [sipcore] SIP TLS issues
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On 3/7/16 12:57 PM, Roman Shpount wrote:

> Putting aside all the problems with SIP Outbound, there are still a lot
> of companies which offer SIP over UDP to the end users. I am talking
> about all the hosted PBX solutions. Most of these deployments do not use
> ICE and use host based NAT traversal for signaling and media to keep the
> message size small. This is not a solution you would design if you have
> started the design from scratch, but these installations definitely need
> a migration path from IPv4 to IPv6. Telling them to switch to SIPS with
> SIP Outbound is unrealistic, so I believe Happy Eyeballs must provide
> the solution for UDP.

Then what is the solution for security. ISTM that people are finally 
realizing that it is important.

Is this for the stupid people market?

	Thanks,
	Paul


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Subject: Re: [sipcore] SIP TLS issues
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Roman Shpount <roman@telurix.com> writes:
> DTLS is not an answer either since encryption state will need to be failed
> over with the edge proxy. What you need is a protocol where client can
> recover a flow to the server, including encryption, using a single round
> trip.
>
> I think deployments of large numbers of SIP devices connected to the public
> service (like SIP based Skype) never happened and will never happen because
> of the complexity and design issues of SIP Outbound in combination with
> TLS. I think this is why none of the large scale consumer VoIP applications
> used SIP. The companies that built these products used RTP, ICE and media
> components, but SIP was ignored. I do not think all these companies spent
> all the design effort to build something completely new just because they
> wanted to build something different. From my experience with couple of
> similar projects the issues of scaling the deployment and handling
> redundancy were the main driving factors.

That sounds to me like the requirements for a useful work item.

Dale


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Subject: Re: [sipcore] Happy Eyeballs -- UDP and RTT
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worley@ariadne.com (Dale R. Worley) writes:
> Happy Eyeballs has a mess of issues surrounding UDP.
> [etc.]

What I'm looking for here is some feedback:  Is adding one or two RTT of
delay to SIP signaling a problem or not?  If two RTT of additional delay
is not a problem, we can be much more restrictive regarding what
mechanisms we define.

Dale


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Subject: Re: [sipcore] Outline of Happy Eyeballs for SIP
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worley@ariadne.com (Dale R. Worley) writes:
>     INVITE sip:biloxi.example.com SIP/2.0
>     Route: <sip:B2;lr>
>     Route: <sip:C;lr>;group=2
>     Route: <sip:0.0.0.0;lr>
>     Via: SIP/2.0/UDP 10.0.1.10;branch=z9hG4bKnashds8.2;group
>     Via: SIP/2.0/UDP 10.0.1.123;branch=z9hG4bK77ef4c2312983
>     Max-Forwards: 19
>     To: Bob <sip:biloxi.example.com>
>     From: Alice <sip:atlanta.example.com>;tag=1928301774
>     Call-ID: a84b4c76e66710
>     CSeq: 314159 INVITE
>     Contact: <sip:atlanta.example.com>
>
> In this version of the mechanism, when either request arrives at C, it
> sees the group parameter on its Route header, which notifies it to
> examine the indicated Via headers.  If C does not support the group
> mechanism, it does not know to ignore and delete the "Route:
> <sip:0.0.0.0;lr>" header, so that copy of the request will error out.
> If C does support the group mechanism, it deletes "Route:
> <sip:0.0.0.0;lr>" and either responds 495 or routes the request to
> bilozi.example.com.

It looks like the proper thing to do is use a different address than
0.0.0.0.  I think we would want to allocate an address from the
192.0.0.0/24 special-purpose block (RFC 5735,
http://www.iana.org/assignments/iana-ipv4-special-registry/iana-ipv4-special-registry.xhtml#iana-ipv4-special-registry-1).
The next available address seems to be 192.0.0.10.  That changes the
example message to:

Probably the 10.0.0.0/24 addresses should be changed to one of the
documentation blocks, e.g., 192.0.2.0/24.

     INVITE sip:biloxi.example.com SIP/2.0
     Route: <sip:B2;lr>
     Route: <sip:C;lr>;group=2
     Route: <sip:192.0.0.10;lr>
     Via: SIP/2.0/UDP 192.0.2.10;branch=z9hG4bKnashds8.2;group
     Via: SIP/2.0/UDP 192.0.2.123;branch=z9hG4bK77ef4c2312983
     Max-Forwards: 19
     To: Bob <sip:biloxi.example.com>
     From: Alice <sip:atlanta.example.com>;tag=1928301774
     Call-ID: a84b4c76e66710
     CSeq: 314159 INVITE
     Contact: <sip:atlanta.example.com>

Dale


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From: Roman Shpount <roman@telurix.com>
To: "Dale R. Worley" <worley@ariadne.com>
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Subject: Re: [sipcore] Happy Eyeballs -- UDP and RTT
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On Tue, Mar 8, 2016 at 1:57 PM, Dale R. Worley <worley@ariadne.com> wrote:

> worley@ariadne.com (Dale R. Worley) writes:
> > Happy Eyeballs has a mess of issues surrounding UDP.
> > [etc.]
>
> What I'm looking for here is some feedback:  Is adding one or two RTT of
> delay to SIP signaling a problem or not?  If two RTT of additional delay
> is not a problem, we can be much more restrictive regarding what
> mechanisms we define.
>

I have mentioned in my earlier email if UDP is replaced by TLS or TCP, then
it needs to be used in conjunction with SIP Outbound. For SIP Outbound the
initial flow setup or flow recovery adds more then two RTT.

One of the common application flows would be to wait for a new call
notification which is sent using a push notification such as Apple Push or
Android Push to a mobile device. Once push notification is received, mobile
device needs to place a call and pick up the inbound call from the proxy.
For this, mobile device needs to establish a TLS connection to the edge
proxy (3.5 RTT), send a registration message to get GRUU (2RTT, first to
get nonce, second to register and get GRUU), and then place the call over
the flow (0.5 RTT). Same delay applies to call recovery if flow gets
disconnected for any reason such as edge proxy failure or local IP address
loss.

Without SIP Outbound with UDP both scenarios take 0.5 or 1.5 RTT depending
on device having a recent nonce for authentication.

Regards,
_____________
Roman Shpount

--089e01494ff22e4eab052d8e9521
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Mar 8, 2016 at 1:57 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"margin:0px 0px 0px 0.8ex;border-left-width:1=
px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:=
1ex"><a href=3D"mailto:worley@ariadne.com">worley@ariadne.com</a> (Dale R. =
Worley) writes:<br>
&gt; Happy Eyeballs has a mess of issues surrounding UDP.<br>
&gt; [etc.]<br>
<br>
What I&#39;m looking for here is some feedback:=C2=A0 Is adding one or two =
RTT of<br>
delay to SIP signaling a problem or not?=C2=A0 If two RTT of additional del=
ay<br>
is not a problem, we can be much more restrictive regarding what<br>
mechanisms we define.<br></blockquote><div><br></div><div>I have mentioned =
in my earlier email if UDP is replaced by TLS or TCP, then it needs to be u=
sed in conjunction with SIP Outbound. For SIP Outbound the initial flow set=
up or flow recovery adds more then two RTT.</div><div><br></div><div>One of=
 the common application flows would be to wait for a new call notification =
which is sent using a push notification such as Apple Push or Android Push =
to a mobile device. Once push notification is received, mobile device needs=
 to place a call and pick up the inbound call from the proxy. For this, mob=
ile device needs to establish a TLS connection to the edge proxy (3.5 RTT),=
 send a registration message to get GRUU (2RTT, first to get nonce, second =
to register and get GRUU), and then place the call over the flow (0.5 RTT).=
 Same delay applies to call recovery if flow gets disconnected for any reas=
on such as edge proxy failure or local IP address loss.</div><div><br></div=
><div>Without SIP Outbound with UDP both scenarios take 0.5 or 1.5 RTT depe=
nding on device having a recent nonce for authentication.</div><div><br></d=
iv><div>Regards,</div><div><div class=3D"gmail_signature">_____________<br>=
Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--089e01494ff22e4eab052d8e9521--


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From: Roman Shpount <roman@telurix.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIP TLS issues
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--047d7bfea0b65914da052d8f1f5e
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On Tue, Mar 8, 2016 at 11:08 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Then what is the solution for security. ISTM that people are finally
> realizing that it is important.
>

You would be surprised -- most of the hosted PBX users do not care or
understand the security related issues. Most of them run their IP phones
with default passwords.


> Is this for the stupid people market?
>

You might think that but I couldn't possibly comment. I would suggest you
lookup SIP configuration settings for any large Hosted PBX provider and
draw your own conclusion. I cannot think of one of them using SIPS.

The question we need to ask is should we help there providers to transition
to IPv6 or should we try to force them to solve their security issues
first? Keep in mind that current solutions for security related issues will
cost those providers substantial amounts of money and their customer do not
care enough about security to pay extra for it.
_____________
Roman Shpount

--047d7bfea0b65914da052d8f1f5e
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Mar 8, 2016 at 11:08 AM, Paul Kyzivat <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.=
edu</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex">Then what is the solution for security. ISTM that people are fina=
lly realizing that it is important.<br></blockquote><div><br></div><div>You=
 would be surprised -- most of the hosted PBX users do not care or understa=
nd the security related issues. Most of them run their IP phones with defau=
lt passwords.</div><div>=C2=A0=C2=A0</div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:=
rgb(204,204,204);border-left-style:solid;padding-left:1ex">Is this for the =
stupid people market?<br></blockquote><div><br></div><div>You might think t=
hat but I couldn&#39;t possibly comment. I would suggest you lookup SIP con=
figuration settings for any large Hosted PBX provider and draw your own con=
clusion. I cannot think of one of them using SIPS.</div><div><br></div><div=
>The question we need to ask is should we help there providers to transitio=
n to IPv6 or should we try to force them to solve their security issues fir=
st? Keep in mind that current solutions for security related issues will co=
st those providers substantial amounts of money and their customer do not c=
are enough about security to pay extra for it.</div><div><div class=3D"gmai=
l_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></d=
iv></div></div>

--047d7bfea0b65914da052d8f1f5e--


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To: Roman Shpount <roman@telurix.com>
References: <mailman.19.1457121604.20561.sipcore@ietf.org> <CAD5OKxuej3cmuna_T9f80ZdCb5junGa+NaTNDG0epyGhFQ+FLQ@mail.gmail.com> <56DDBB10.6050909@alum.mit.edu> <CAD5OKxuYj+LHRBU761ted8fee3gjfwEu44qbZOvR_NZAwss-+w@mail.gmail.com> <56DEF90C.80105@alum.mit.edu> <CAD5OKxtzq8DrnvSX4yhcO0UixmV4+5qbKPR1c7R0ZQJE_jZtLQ@mail.gmail.com>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIP TLS issues
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On 3/8/16 3:06 PM, Roman Shpount wrote:
> On Tue, Mar 8, 2016 at 11:08 AM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     Then what is the solution for security. ISTM that people are finally
>     realizing that it is important.
>
>
> You would be surprised -- most of the hosted PBX users do not care or
> understand the security related issues. Most of them run their IP phones
> with default passwords.

I can "sort of" understand doing this behind a corporate firewall.

I guess that would extend to hosted, if that was over a VPN.

Otherwise this is dubious at best.

>     Is this for the stupid people market?
>
>
> You might think that but I couldn't possibly comment. I would suggest
> you lookup SIP configuration settings for any large Hosted PBX provider
> and draw your own conclusion. I cannot think of one of them using SIPS.

Sorry for sounding disrespectful. (I didn't mean you. I was intending to 
pass comment on all who don't seem to care.)

SIPS is another story. It is kind of a mess. But TLS without SIPS is 
possible.

> The question we need to ask is should we help there providers to
> transition to IPv6 or should we try to force them to solve their
> security issues first? Keep in mind that current solutions for security
> related issues will cost those providers substantial amounts of money
> and their customer do not care enough about security to pay extra for it.

I suppose that is fair.

OTOH, we are losing an opportunity to bundle stuff that people should do 
if they weren't so cheap with other stuff they are more motivated to do.

A lot of work went into outbound. It is complicated, but was the best 
anybody could find that could provide security for the last hop for 
those who don't have a certificate. If we knew an easier way we would 
have specified it.

	Thanks,
	Paul


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Subject: [sipcore]  SIP OAuth - Use Cases
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Hi,

We have submitted a new version of the SIP OAuth document, and added few
more use cases as described in section 1.3.
http://www.ietf.org/id/draft-yusef-sipcore-sip-oauth-03.txt


We would appreciate any feedback on these use cases.

Regards,
 Rifaat

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<div dir=3D"ltr">Hi,<div><br></div><div>We have submitted a new version of =
the SIP OAuth document, and added few more use cases as described in sectio=
n 1.3.</div><div><a href=3D"http://www.ietf.org/id/draft-yusef-sipcore-sip-=
oauth-03.txt" target=3D"_blank">http://www.ietf.org/id/draft-yusef-sipcore-=
sip-oauth-03.txt</a><br></div><div><br></div><div><br></div><div>We would a=
ppreciate any feedback on these use cases.</div><div><br></div><div>Regards=
,</div><div>=C2=A0Rifaat</div><div>=C2=A0</div></div>

--001a113d391c4bec05052d930f61--


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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Happy Eyeballs -- UDP and RTT
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On 3/8/16 2:28 PM, Roman Shpount wrote:
> On Tue, Mar 8, 2016 at 1:57 PM, Dale R. Worley <worley@ariadne.com
> <mailto:worley@ariadne.com>> wrote:
>
>     worley@ariadne.com <mailto:worley@ariadne.com> (Dale R. Worley) writes:
>      > Happy Eyeballs has a mess of issues surrounding UDP.
>      > [etc.]
>
>     What I'm looking for here is some feedback:  Is adding one or two RTT of
>     delay to SIP signaling a problem or not?  If two RTT of additional delay
>     is not a problem, we can be much more restrictive regarding what
>     mechanisms we define.
>
>
> I have mentioned in my earlier email if UDP is replaced by TLS or TCP,
> then it needs to be used in conjunction with SIP Outbound. For SIP
> Outbound the initial flow setup or flow recovery adds more then two RTT.
>
> One of the common application flows would be to wait for a new call
> notification which is sent using a push notification such as Apple Push
> or Android Push to a mobile device. Once push notification is received,
> mobile device needs to place a call and pick up the inbound call from
> the proxy.

This is a common application flow?

I'm not aware of this being a standard usage of sip.

> For this, mobile device needs to establish a TLS connection
> to the edge proxy (3.5 RTT), send a registration message to get GRUU
> (2RTT, first to get nonce, second to register and get GRUU), and then
> place the call over the flow (0.5 RTT). Same delay applies to call
> recovery if flow gets disconnected for any reason such as edge proxy
> failure or local IP address loss.
>
> Without SIP Outbound with UDP both scenarios take 0.5 or 1.5 RTT
> depending on device having a recent nonce for authentication.

I understand that this is very "traditional".

It would also never get past a security review these days.

If we have no secure solution that is deployable in practice, then I 
guess we ought to be setting up an effort to come up with one.

Maybe the answer is "use webrtc".

	Thanks,
	Paul


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From: Roman Shpount <roman@telurix.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Happy Eyeballs -- UDP and RTT
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On Wed, Mar 9, 2016 at 12:19 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 3/8/16 2:28 PM, Roman Shpount wrote:
>
>> One of the common application flows would be to wait for a new call
>> notification which is sent using a push notification such as Apple Push
>> or Android Push to a mobile device. Once push notification is received,
>> mobile device needs to place a call and pick up the inbound call from
>> the proxy.
>>
>
> This is a common application flow?
>
> I'm not aware of this being a standard usage of sip.


This is not completely non-standard either. SIP here is only used to
actually place the call. SIP registration is replaced by the platform
native push functionality. This flow has an upside of reducing the server
load on the provider side (no idle registration connection waiting for
inbound calls) and reducing power usage on the client side (no  idle
registration connection waiting for inbound calls or and CPU does not need
to periodically wake up to send a keep alive).

For this, mobile device needs to establish a TLS connection
>> to the edge proxy (3.5 RTT), send a registration message to get GRUU
>> (2RTT, first to get nonce, second to register and get GRUU), and then
>> place the call over the flow (0.5 RTT). Same delay applies to call
>> recovery if flow gets disconnected for any reason such as edge proxy
>> failure or local IP address loss.
>>
>> Without SIP Outbound with UDP both scenarios take 0.5 or 1.5 RTT
>> depending on device having a recent nonce for authentication.
>>
>
> I understand that this is very "traditional".
>
> It would also never get past a security review these days.
>
> If we have no secure solution that is deployable in practice, then I guess
> we ought to be setting up an effort to come up with one.
>
> Maybe the answer is "use webrtc".
>

WebRTC might be the answer. Since there is a lot of effort is already being
spent to optimize HTTP request processing and to reduce the number of round
trips required to get HTTP response, there is no reason not to reuse this
work.

Unfortunately this still leaves the question of what to do with all the
existing IP phones deployed with hosted PBXs. They would need to be
migrated to IPv6 sooner rather then later.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Wed, Mar 9, 2016 at 12:19 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.=
edu</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex"><span class=3D"">On 3/8/16 2:28 PM, Roman Shpount wrote:<br>
</span><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:=
solid;padding-left:1ex"><span class=3D"">One of the common application flow=
s would be to wait for a new call<br>
notification which is sent using a push notification such as Apple Push<br>
or Android Push to a mobile device. Once push notification is received,<br>
mobile device needs to place a call and pick up the inbound call from<br>
the proxy.<br>
</span></blockquote>
<br>
This is a common application flow?<br>
<br>
I&#39;m not aware of this being a standard usage of sip.</blockquote><div><=
br></div><div>This is not completely non-standard either. SIP here is only =
used to actually place the call. SIP registration is replaced by the platfo=
rm native push functionality. This flow has an upside of reducing the serve=
r load on the provider side (no idle registration connection waiting for in=
bound calls) and reducing power usage on the client side (no =C2=A0idle reg=
istration connection waiting for inbound calls or and CPU does not need to =
periodically wake up to send a keep alive).</div><div><br></div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:=
1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left=
:1ex"><span class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0p=
x 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bo=
rder-left-style:solid;padding-left:1ex">For this, mobile device needs to es=
tablish a TLS connection<br>
to the edge proxy (3.5 RTT), send a registration message to get GRUU<br>
(2RTT, first to get nonce, second to register and get GRUU), and then<br>
place the call over the flow (0.5 RTT). Same delay applies to call<br>
recovery if flow gets disconnected for any reason such as edge proxy<br>
failure or local IP address loss.<br>
<br>
Without SIP Outbound with UDP both scenarios take 0.5 or 1.5 RTT<br>
depending on device having a recent nonce for authentication.<br>
</blockquote>
<br></span>
I understand that this is very &quot;traditional&quot;.<br>
<br>
It would also never get past a security review these days.<br>
<br>
If we have no secure solution that is deployable in practice, then I guess =
we ought to be setting up an effort to come up with one.<br>
<br>
Maybe the answer is &quot;use webrtc&quot;.<br></blockquote><div><br></div>=
<div>WebRTC might be the answer. Since there is a lot of effort is already =
being spent to optimize HTTP request processing and to reduce the number of=
 round trips required to get HTTP response, there is no reason not to reuse=
 this work.</div><div><br></div><div>Unfortunately this still leaves the qu=
estion of what to do with all the existing IP phones deployed with hosted P=
BXs. They would need to be migrated to IPv6 sooner rather then later.</div>=
<div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></d=
iv><div>=C2=A0</div></div></div></div>

--001a1140b472de1ed7052da1da40--


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> BTW, looking at RFC 6555 section 4, I see:

<snip>

> Am I misunderstanding this?  Item (1) says that there is
> a delay between initiating an IPv6 connection and
> initiating an IPv4 connection, but the text at the end
> says that they are initiated simultaneously.

I agree that the text is contradictory.

Section 5.5 indicates that stateful algorithms can more aggressive than
the recommended 150-250 ms delay; however it still recommends "connection
attempts be paced to give connections a chance to complete".


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> > Happy Eyeballs has a mess of issues surrounding UDP.
> > [etc.]

I agree.  In addition to the issues that have already been raised, the
solution potentially would need to update RFC 4320 concerning sending 100
over UDP for non-INVITE.


> What I'm looking for here is some feedback:  Is adding
> one or two RTT of delay to SIP signaling a problem or not?
> If two RTT of additional delay is not a problem, we can be
> much more restrictive regarding what mechanisms we define.

It likely depends upon the reason and frequency.  The
draft-ietf-sipcore-dns-dual-stack recommendation already has the potential
to double the amount of time it takes to advance from primary to secondary
when primary is down.  For example, trying primary AAAA and A before
advancing to SRV record for secondary which also has AAAA and A.
Hopefully, the Happy Eyeballs resolution allows hitting the forth location
(secondary A record) within an acceptable amount of time.  And hopefully
the Happy Eyeballs resolution doesn't introduce too much extra traffic.


From nobody Fri Mar 11 15:06:15 2016
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Subject: [sipcore] sipcore - Requested session has been scheduled for IETF 95
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Dear Adam Roach,

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

sipcore Session 1 (1:00:00)
    Friday, Afternoon Session I 1220-1320
    Room Name: Buen Ayre C size: 250
    ---------------------------------------------
    


Request Information:


---------------------------------------------------------
Working Group Name: Session Initiation Protocol Core
Area Name: Applications and Real-Time Area
Session Requester: Adam Roach

Number of Sessions: 1
Length of Session(s):  1 Hour
Number of Attendees: 100
Conflicts to Avoid: 
 First Priority: netvc xrblock webpush stir mmusic rtcweb avtext perc avtcore dispatch




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


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Subject: [sipcore] "Roman's problem"
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Going over Roman's messages regarding hosted PBX systems, I want to
isolate that usage situation from the rest of Happy Eyeballs.  It seems
like the problems there are seperable from what Happy Eyeballs needs to
solve.  No doubt SIP endpoints in a hosted PBX situation need Happy
Eyeballs also, but rapid failover does not involve the endpoint trying
alternative transport destinations.  (Happy Eyeballs is involved in the
endpoint selecting a transport destination to contact initially, but
presumably H.E. will allow the endpoint to continue to use that one
destination as long as it is responding.)

I've extracted parts of a number of messages that characterize the
situation and put them at the end of this message.  The summary of the
situation seems to be:

----------
Failover is done by moving the edge-proxy IP address to the backup endge
proxy and using a protocol without connection state, i.e., UDP
without DTLS.

The registration state is stored in a database shared by all edge
proxy instances.  There need be no dialog state in the edge proxies,
and transaction failure during failover is acceptable.

Digest authentication nonces can be constructed to be verifiable by
any edge proxy.

NAT traversal is handled by NAT compensation done by the edge proxy
rather than ICE (thus keeping messages small enough for UDP).

The unsolved problem is security:
- message integrity
- message privacy
without requiring connection state.
----------

Nonces can be constructed by the edge proxies in a way that doesn't
require connection state to check their validity.  E.g.,

    nonce = T || R || H(T || R || PK)

where

    T = timestamp
    R = random value
    PK = private key known by all the edge proxies

Security seems to be more of a problem in that all of the security
protocols I know of require connection state.  However, it shouldn't be
difficult to construct an adequate security protocol.  E.g., assuming
the phone and the proxies have a shared key SK for a user identity U,
one could encrypt a message via:

    encrypted message = "@" || U || MD || E(MK, message)

where

    message digest (MD) = H(message)
    message key (MK) = H(U || MD || SK)
    U = fixed-length user identity
    "@" is chosen as a flag because no SIP message can begin with "@"

That should provide sufficient message integrity and privacy.  Or more
likely, a considerably better scheme could be devised by someone who
knows encryption algorithms.  But we only need a stateless security
mechanism that's much better than Digest authentication.

Dale
----------------------------------------------------------------------
I know a lot of implementations for UDP which handle 1M+ of concurrent
registrations.

For UDP you can move the IP address for the edge proxy to a different
server. If registration and flow state is in some sort of shared DB,
the fail-over occurs naturally for both the client and server.

Happy Eyeballs is needed to address situation where you have large
number of end user devices located on multiple customer networks with
different capabilities trying to connect to a central public service
which is accessible over both IPv4 and IPv6. The point I am trying to
make that all of such SIP services primarily use UDP so it is too
early to discard it.

The obvious problems here are:

1) security
2) large sip messages and UDP message fragmentation

DTLS is not an answer either since encryption state will need to be
failed over with the edge proxy. What you need is a protocol where
client can recover a flow to the server, including encryption, using a
single round trip.

I think deployments of large numbers of SIP devices connected to the
public service (like SIP based Skype) never happened and will never
happen because of the complexity and design issues of SIP Outbound in
combination with TLS. I think this is why none of the large scale
consumer VoIP applications used SIP. The companies that built these
products used RTP, ICE and media components, but SIP was ignored. I do
not think all these companies spent all the design effort to build
something completely new just because they wanted to build something
different. From my experience with couple of similar projects the
issues of scaling the deployment and handling redundancy were the main
driving factors.

Putting aside all the problems with SIP Outbound, there are still a
lot of companies which offer SIP over UDP to the end users. I am
talking about all the hosted PBX solutions. Most of these deployments
do not use ICE and use host based NAT traversal for signaling and
media to keep the message size small. This is not a solution you would
design if you have started the design from scratch, but these
installations definitely need a migration path from IPv4 to
IPv6. Telling them to switch to SIPS with SIP Outbound is unrealistic,
so I believe Happy Eyeballs must provide the solution for UDP.


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Date: Wed, 16 Mar 2016 14:08:47 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [sipcore] "Roman's problem"
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--001a11c3165ec5886e052e2e6757
Content-Type: text/plain; charset=UTF-8

This looks correct. The reason I brought this scenario up in relationship
with Happy Eyeballs is to make sure that Happy Eyeballs works with UDP for
initial call setup. There was a discussion of dropping UDP support in Happy
Eyeballs compliant end points. You are correct, that once the initial
registration is established, Happy Eyeballs does not need to be involved,
but this entire scenario relies on using UDP based SIP.

One small correction: In such a setup scenario, if edge proxies are
stateless and proxy fail-over takes less then 32 sec (Timer B), event SIP
transactions would normally survive an edge-proxy fail-over.

Regards,
_____________
Roman Shpount

On Wed, Mar 16, 2016 at 1:40 PM, Dale R. Worley <worley@ariadne.com> wrote:

> Going over Roman's messages regarding hosted PBX systems, I want to
> isolate that usage situation from the rest of Happy Eyeballs.  It seems
> like the problems there are seperable from what Happy Eyeballs needs to
> solve.  No doubt SIP endpoints in a hosted PBX situation need Happy
> Eyeballs also, but rapid failover does not involve the endpoint trying
> alternative transport destinations.  (Happy Eyeballs is involved in the
> endpoint selecting a transport destination to contact initially, but
> presumably H.E. will allow the endpoint to continue to use that one
> destination as long as it is responding.)
>
> I've extracted parts of a number of messages that characterize the
> situation and put them at the end of this message.  The summary of the
> situation seems to be:
>
> ----------
> Failover is done by moving the edge-proxy IP address to the backup endge
> proxy and using a protocol without connection state, i.e., UDP
> without DTLS.
>
> The registration state is stored in a database shared by all edge
> proxy instances.  There need be no dialog state in the edge proxies,
> and transaction failure during failover is acceptable.
>
> Digest authentication nonces can be constructed to be verifiable by
> any edge proxy.
>
> NAT traversal is handled by NAT compensation done by the edge proxy
> rather than ICE (thus keeping messages small enough for UDP).
>
> The unsolved problem is security:
> - message integrity
> - message privacy
> without requiring connection state.
> ----------
>
> Nonces can be constructed by the edge proxies in a way that doesn't
> require connection state to check their validity.  E.g.,
>
>     nonce = T || R || H(T || R || PK)
>
> where
>
>     T = timestamp
>     R = random value
>     PK = private key known by all the edge proxies
>
> Security seems to be more of a problem in that all of the security
> protocols I know of require connection state.  However, it shouldn't be
> difficult to construct an adequate security protocol.  E.g., assuming
> the phone and the proxies have a shared key SK for a user identity U,
> one could encrypt a message via:
>
>     encrypted message = "@" || U || MD || E(MK, message)
>
> where
>
>     message digest (MD) = H(message)
>     message key (MK) = H(U || MD || SK)
>     U = fixed-length user identity
>     "@" is chosen as a flag because no SIP message can begin with "@"
>
> That should provide sufficient message integrity and privacy.  Or more
> likely, a considerably better scheme could be devised by someone who
> knows encryption algorithms.  But we only need a stateless security
> mechanism that's much better than Digest authentication.
>
> Dale
> ----------------------------------------------------------------------
> I know a lot of implementations for UDP which handle 1M+ of concurrent
> registrations.
>
> For UDP you can move the IP address for the edge proxy to a different
> server. If registration and flow state is in some sort of shared DB,
> the fail-over occurs naturally for both the client and server.
>
> Happy Eyeballs is needed to address situation where you have large
> number of end user devices located on multiple customer networks with
> different capabilities trying to connect to a central public service
> which is accessible over both IPv4 and IPv6. The point I am trying to
> make that all of such SIP services primarily use UDP so it is too
> early to discard it.
>
> The obvious problems here are:
>
> 1) security
> 2) large sip messages and UDP message fragmentation
>
> DTLS is not an answer either since encryption state will need to be
> failed over with the edge proxy. What you need is a protocol where
> client can recover a flow to the server, including encryption, using a
> single round trip.
>
> I think deployments of large numbers of SIP devices connected to the
> public service (like SIP based Skype) never happened and will never
> happen because of the complexity and design issues of SIP Outbound in
> combination with TLS. I think this is why none of the large scale
> consumer VoIP applications used SIP. The companies that built these
> products used RTP, ICE and media components, but SIP was ignored. I do
> not think all these companies spent all the design effort to build
> something completely new just because they wanted to build something
> different. From my experience with couple of similar projects the
> issues of scaling the deployment and handling redundancy were the main
> driving factors.
>
> Putting aside all the problems with SIP Outbound, there are still a
> lot of companies which offer SIP over UDP to the end users. I am
> talking about all the hosted PBX solutions. Most of these deployments
> do not use ICE and use host based NAT traversal for signaling and
> media to keep the message size small. This is not a solution you would
> design if you have started the design from scratch, but these
> installations definitely need a migration path from IPv4 to
> IPv6. Telling them to switch to SIPS with SIP Outbound is unrealistic,
> so I believe Happy Eyeballs must provide the solution for UDP.
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

--001a11c3165ec5886e052e2e6757
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">This looks correct. The reason I brought this scenario up =
in relationship with Happy=C2=A0Eyeballs=C2=A0is to make sure that=C2=A0Hap=
py=C2=A0Eyeballs works with UDP for initial call setup. There was a discuss=
ion of dropping UDP support in Happy Eyeballs compliant end points. You are=
 correct, that once the initial registration is established, Happy Eyeballs=
 does not need to be involved, but this entire scenario relies on using UDP=
 based SIP.<div><br></div><div>One small correction: In such a setup scenar=
io, if edge proxies are stateless and proxy fail-over takes less then 32 se=
c (Timer B), event SIP transactions would normally survive an edge-proxy fa=
il-over.</div><div><div><br></div><div>Regards,<div class=3D"gmail_extra"><=
div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></di=
v>
<br><div class=3D"gmail_quote">On Wed, Mar 16, 2016 at 1:40 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:0px 0px 0px 0.8ex;border-left-width:1px;border-le=
ft-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">Going o=
ver Roman&#39;s messages regarding hosted PBX systems, I want to<br>
isolate that usage situation from the rest of Happy Eyeballs.=C2=A0 It seem=
s<br>
like the problems there are seperable from what Happy Eyeballs needs to<br>
solve.=C2=A0 No doubt SIP endpoints in a hosted PBX situation need Happy<br=
>
Eyeballs also, but rapid failover does not involve the endpoint trying<br>
alternative transport destinations.=C2=A0 (Happy Eyeballs is involved in th=
e<br>
endpoint selecting a transport destination to contact initially, but<br>
presumably H.E. will allow the endpoint to continue to use that one<br>
destination as long as it is responding.)<br>
<br>
I&#39;ve extracted parts of a number of messages that characterize the<br>
situation and put them at the end of this message.=C2=A0 The summary of the=
<br>
situation seems to be:<br>
<br>
----------<br>
Failover is done by moving the edge-proxy IP address to the backup endge<br=
>
proxy and using a protocol without connection state, i.e., UDP<br>
without DTLS.<br>
<br>
The registration state is stored in a database shared by all edge<br>
proxy instances.=C2=A0 There need be no dialog state in the edge proxies,<b=
r>
and transaction failure during failover is acceptable.<br>
<br>
Digest authentication nonces can be constructed to be verifiable by<br>
any edge proxy.<br>
<br>
NAT traversal is handled by NAT compensation done by the edge proxy<br>
rather than ICE (thus keeping messages small enough for UDP).<br>
<br>
The unsolved problem is security:<br>
- message integrity<br>
- message privacy<br>
without requiring connection state.<br>
----------<br>
<br>
Nonces can be constructed by the edge proxies in a way that doesn&#39;t<br>
require connection state to check their validity.=C2=A0 E.g.,<br>
<br>
=C2=A0 =C2=A0 nonce =3D T || R || H(T || R || PK)<br>
<br>
where<br>
<br>
=C2=A0 =C2=A0 T =3D timestamp<br>
=C2=A0 =C2=A0 R =3D random value<br>
=C2=A0 =C2=A0 PK =3D private key known by all the edge proxies<br>
<br>
Security seems to be more of a problem in that all of the security<br>
protocols I know of require connection state.=C2=A0 However, it shouldn&#39=
;t be<br>
difficult to construct an adequate security protocol.=C2=A0 E.g., assuming<=
br>
the phone and the proxies have a shared key SK for a user identity U,<br>
one could encrypt a message via:<br>
<br>
=C2=A0 =C2=A0 encrypted message =3D &quot;@&quot; || U || MD || E(MK, messa=
ge)<br>
<br>
where<br>
<br>
=C2=A0 =C2=A0 message digest (MD) =3D H(message)<br>
=C2=A0 =C2=A0 message key (MK) =3D H(U || MD || SK)<br>
=C2=A0 =C2=A0 U =3D fixed-length user identity<br>
=C2=A0 =C2=A0 &quot;@&quot; is chosen as a flag because no SIP message can =
begin with &quot;@&quot;<br>
<br>
That should provide sufficient message integrity and privacy.=C2=A0 Or more=
<br>
likely, a considerably better scheme could be devised by someone who<br>
knows encryption algorithms.=C2=A0 But we only need a stateless security<br=
>
mechanism that&#39;s much better than Digest authentication.<br>
<br>
Dale<br>
----------------------------------------------------------------------<br>
I know a lot of implementations for UDP which handle 1M+ of concurrent<br>
registrations.<br>
<br>
For UDP you can move the IP address for the edge proxy to a different<br>
server. If registration and flow state is in some sort of shared DB,<br>
the fail-over occurs naturally for both the client and server.<br>
<br>
Happy Eyeballs is needed to address situation where you have large<br>
number of end user devices located on multiple customer networks with<br>
different capabilities trying to connect to a central public service<br>
which is accessible over both IPv4 and IPv6. The point I am trying to<br>
make that all of such SIP services primarily use UDP so it is too<br>
early to discard it.<br>
<br>
The obvious problems here are:<br>
<br>
1) security<br>
2) large sip messages and UDP message fragmentation<br>
<br>
DTLS is not an answer either since encryption state will need to be<br>
failed over with the edge proxy. What you need is a protocol where<br>
client can recover a flow to the server, including encryption, using a<br>
single round trip.<br>
<br>
I think deployments of large numbers of SIP devices connected to the<br>
public service (like SIP based Skype) never happened and will never<br>
happen because of the complexity and design issues of SIP Outbound in<br>
combination with TLS. I think this is why none of the large scale<br>
consumer VoIP applications used SIP. The companies that built these<br>
products used RTP, ICE and media components, but SIP was ignored. I do<br>
not think all these companies spent all the design effort to build<br>
something completely new just because they wanted to build something<br>
different. From my experience with couple of similar projects the<br>
issues of scaling the deployment and handling redundancy were the main<br>
driving factors.<br>
<br>
Putting aside all the problems with SIP Outbound, there are still a<br>
lot of companies which offer SIP over UDP to the end users. I am<br>
talking about all the hosted PBX solutions. Most of these deployments<br>
do not use ICE and use host based NAT traversal for signaling and<br>
media to keep the message size small. This is not a solution you would<br>
design if you have started the design from scratch, but these<br>
installations definitely need a migration path from IPv4 to<br>
IPv6. Telling them to switch to SIPS with SIP Outbound is unrealistic,<br>
so I believe Happy Eyeballs must provide the solution for UDP.<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></div></div>

--001a11c3165ec5886e052e2e6757--


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From: Roman Shpount <roman@telurix.com>
To: "Dale R. Worley" <worley@ariadne.com>
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Subject: Re: [sipcore] "Roman's problem"
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--001a1140b47281ebaa052e2ef4e4
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On Wed, Mar 16, 2016 at 2:08 PM, Roman Shpount <roman@telurix.com> wrote:

> On Wed, Mar 16, 2016 at 1:40 PM, Dale R. Worley <worley@ariadne.com>
> wrote:
>
>> Going over Roman's messages regarding hosted PBX systems, I want to
>> isolate that usage situation from the rest of Happy Eyeballs.  It seems
>> like the problems there are seperable from what Happy Eyeballs needs to
>> solve.  No doubt SIP endpoints in a hosted PBX situation need Happy
>> Eyeballs also, but rapid failover does not involve the endpoint trying
>> alternative transport destinations.  (Happy Eyeballs is involved in the
>> endpoint selecting a transport destination to contact initially, but
>> presumably H.E. will allow the endpoint to continue to use that one
>> destination as long as it is responding.)
>>
>> I've extracted parts of a number of messages that characterize the
>> situation and put them at the end of this message.  The summary of the
>> situation seems to be:
>>
>> ----------
>> Failover is done by moving the edge-proxy IP address to the backup endge
>> proxy and using a protocol without connection state, i.e., UDP
>> without DTLS.
>>
>> The registration state is stored in a database shared by all edge
>> proxy instances.  There need be no dialog state in the edge proxies,
>> and transaction failure during failover is acceptable.
>>
>> Digest authentication nonces can be constructed to be verifiable by
>> any edge proxy.
>>
>> NAT traversal is handled by NAT compensation done by the edge proxy
>> rather than ICE (thus keeping messages small enough for UDP).
>>
>> The unsolved problem is security:
>> - message integrity
>> - message privacy
>> without requiring connection state.
>> ----------
>>
>> Nonces can be constructed by the edge proxies in a way that doesn't
>> require connection state to check their validity.  E.g.,
>>
>>     nonce = T || R || H(T || R || PK)
>>
>> where
>>
>>     T = timestamp
>>     R = random value
>>     PK = private key known by all the edge proxies
>>
>> Security seems to be more of a problem in that all of the security
>> protocols I know of require connection state.  However, it shouldn't be
>> difficult to construct an adequate security protocol.  E.g., assuming
>> the phone and the proxies have a shared key SK for a user identity U,
>> one could encrypt a message via:
>>
>>     encrypted message = "@" || U || MD || E(MK, message)
>>
>> where
>>
>>     message digest (MD) = H(message)
>>     message key (MK) = H(U || MD || SK)
>>     U = fixed-length user identity
>>     "@" is chosen as a flag because no SIP message can begin with "@"
>>
>> That should provide sufficient message integrity and privacy.  Or more
>> likely, a considerably better scheme could be devised by someone who
>> knows encryption algorithms.  But we only need a stateless security
>> mechanism that's much better than Digest authentication.
>>
>>
As far as encryption is concerned, do you expect it to replace Digest or to
be used concurrently with it? i would assume there is no reason to run
digest in parallel with encryption.

If you want to use it to replace Digest (there is no reason why it should
not), encrypted message should include nonce. Presence of nonce will allow
to prevent message replay.

How would responses or messages from the server to the client going to be
encrypted?

My suggestion would be a bit more complex:

1. Client starts by sending a setup message with crypto capabilities

2. Server sends a response with server crypto capabilities, nonce and
connection specific server public key (nonce can contain information
regarding which specific server key is used for this connection). This
message should be signed using server public certificate

3. Send the registration message from the client to the server encrypted
with server public key. Include user identity, nonce, client public key and
message digest signed with user shared key.

4. On the proxy side store connection specific server private key, client
public key and crypto capabilities in the same DB as registration
information.

This protocol should provide sufficient security including server
certificate validation, replay protection, forward secrecy one registration
is torn down. On the other hand it will have the same connection recovery
properties.

Regards,
_____________
Roman Shpount

--001a1140b47281ebaa052e2ef4e4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Wed, Mar 16, 2016 at 2:08 PM, Roman Shpount <span dir=3D"ltr">&lt;<=
a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>=
&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockquote cl=
ass=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:1e=
x"><div dir=3D"ltr"><div><div><div class=3D"gmail_extra"><div><div class=3D=
"h5"><div class=3D"gmail_quote">On Wed, Mar 16, 2016 at 1:40 PM, Dale R. Wo=
rley <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"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-l=
eft-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">Going =
over Roman&#39;s messages regarding hosted PBX systems, I want to<br>
isolate that usage situation from the rest of Happy Eyeballs.=C2=A0 It seem=
s<br>
like the problems there are seperable from what Happy Eyeballs needs to<br>
solve.=C2=A0 No doubt SIP endpoints in a hosted PBX situation need Happy<br=
>
Eyeballs also, but rapid failover does not involve the endpoint trying<br>
alternative transport destinations.=C2=A0 (Happy Eyeballs is involved in th=
e<br>
endpoint selecting a transport destination to contact initially, but<br>
presumably H.E. will allow the endpoint to continue to use that one<br>
destination as long as it is responding.)<br>
<br>
I&#39;ve extracted parts of a number of messages that characterize the<br>
situation and put them at the end of this message.=C2=A0 The summary of the=
<br>
situation seems to be:<br>
<br>
----------<br>
Failover is done by moving the edge-proxy IP address to the backup endge<br=
>
proxy and using a protocol without connection state, i.e., UDP<br>
without DTLS.<br>
<br>
The registration state is stored in a database shared by all edge<br>
proxy instances.=C2=A0 There need be no dialog state in the edge proxies,<b=
r>
and transaction failure during failover is acceptable.<br>
<br>
Digest authentication nonces can be constructed to be verifiable by<br>
any edge proxy.<br>
<br>
NAT traversal is handled by NAT compensation done by the edge proxy<br>
rather than ICE (thus keeping messages small enough for UDP).<br>
<br>
The unsolved problem is security:<br>
- message integrity<br>
- message privacy<br>
without requiring connection state.<br>
----------<br>
<br>
Nonces can be constructed by the edge proxies in a way that doesn&#39;t<br>
require connection state to check their validity.=C2=A0 E.g.,<br>
<br>
=C2=A0 =C2=A0 nonce =3D T || R || H(T || R || PK)<br>
<br>
where<br>
<br>
=C2=A0 =C2=A0 T =3D timestamp<br>
=C2=A0 =C2=A0 R =3D random value<br>
=C2=A0 =C2=A0 PK =3D private key known by all the edge proxies<br>
<br>
Security seems to be more of a problem in that all of the security<br>
protocols I know of require connection state.=C2=A0 However, it shouldn&#39=
;t be<br>
difficult to construct an adequate security protocol.=C2=A0 E.g., assuming<=
br>
the phone and the proxies have a shared key SK for a user identity U,<br>
one could encrypt a message via:<br>
<br>
=C2=A0 =C2=A0 encrypted message =3D &quot;@&quot; || U || MD || E(MK, messa=
ge)<br>
<br>
where<br>
<br>
=C2=A0 =C2=A0 message digest (MD) =3D H(message)<br>
=C2=A0 =C2=A0 message key (MK) =3D H(U || MD || SK)<br>
=C2=A0 =C2=A0 U =3D fixed-length user identity<br>
=C2=A0 =C2=A0 &quot;@&quot; is chosen as a flag because no SIP message can =
begin with &quot;@&quot;<br>
<br>
That should provide sufficient message integrity and privacy.=C2=A0 Or more=
<br>
likely, a considerably better scheme could be devised by someone who<br>
knows encryption algorithms.=C2=A0 But we only need a stateless security<br=
>
mechanism that&#39;s much better than Digest authentication.<br>
<br></blockquote></div></div></div></div></div></div></div></blockquote><di=
v><br></div><div>As far as encryption is concerned, do you expect it to rep=
lace Digest or to be used concurrently with it? i would assume there is no =
reason to run digest in parallel with encryption.</div><div><br></div><div>=
If you want to use it to replace Digest (there is no reason why it should n=
ot), encrypted message should include nonce. Presence of nonce will allow t=
o prevent message replay.</div><div><br></div><div>How would responses or m=
essages from the server to the client going to be encrypted?</div><div><br>=
</div><div>My suggestion would be a bit more complex:</div><div><br></div><=
div>1. Client starts by sending a setup message with crypto capabilities</d=
iv><div><br></div><div>2. Server sends a response with server crypto capabi=
lities, nonce and connection specific server public key (nonce can contain =
information regarding which specific server key is used for this connection=
). This message should be signed using server public certificate</div><div>=
<br></div><div>3. Send the registration message from the client to the serv=
er encrypted with server public key. Include user identity, nonce, client p=
ublic key and message digest signed with user shared key.</div><div><br></d=
iv><div>4. On the proxy side store connection specific server private key, =
client public key and crypto capabilities in the same DB as registration in=
formation.</div><div><br></div><div>This protocol should provide sufficient=
 security including server certificate validation, replay protection, forwa=
rd secrecy one registration is torn down. On the other hand it will have th=
e same connection recovery properties.</div><div><br></div><div>Regards,=C2=
=A0</div><div><div class=3D"gmail_signature">_____________<br>Roman Shpount=
</div></div><div>=C2=A0</div></div></div></div>

--001a1140b47281ebaa052e2ef4e4--


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Subject: Re: [sipcore] Happy Eyeballs -- UDP and RTT
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Brett Tate <brett@broadsoft.com> writes:
> Section 5.5 indicates that stateful algorithms can more aggressive than
> the recommended 150-250 ms delay; however it still recommends "connection
> attempts be paced to give connections a chance to complete".

Thanks!  Yet another point regarding revising timeouts.

Dale


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Subject: Re: [sipcore] "Roman's problem"
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Roman Shpount <roman@telurix.com> writes:
> 4. On the proxy side store connection specific server private key, client
> public key and crypto capabilities in the same DB as registration
> information.

It struck me that the central problem was encryption without maintaining
connection-specific encryption information.  If we're willing to
maintain that, presumably DTLS would work.  I don't know much about
DTLS, though -- do you think that UDP/DTLS would be a sufficient
solution to the use cases we are discussing?

Dale


From nobody Thu Mar 17 09:18:35 2016
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From: worley@ariadne.com (Dale R. Worley)
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [sipcore] "Roman's problem"
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Roman Shpount <roman@telurix.com> writes:
> One small correction: In such a setup scenario, if edge proxies are
> stateless and proxy fail-over takes less then 32 sec (Timer B), event SIP
> transactions would normally survive an edge-proxy fail-over.

My concern about transaction failure is this:

UAC sends request
edge-proxy 1 forwards request to UAS, retaining transaction state
edge-proxy 1 fails
edge-proxy 2 replaces edge-proxy 1
UAS sends response to edge-proxy 2
edge-proxy 2 has no transaction state to match the response and discards it
UAS resends response to edge-proxy 2
...

I'm sure that the real scenario is more complicated than that, but
there's a general problem if the edge-proxy, or any proxy in the chain,
is transaction-stateful, and you generally want proxies to be
transaction-stateful.

But I think we're OK in practice, in that real UAs are resistant to
transaction failure, transaction failures degrade the user experience
but the user can easily recover from it.  And transaction failure will
be quite uncommon.

Dale


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From: Roman Shpount <roman@telurix.com>
To: "Dale R. Worley" <worley@ariadne.com>
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Subject: Re: [sipcore] "Roman's problem"
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--001a113eca0681a1b4052e410a72
Content-Type: text/plain; charset=UTF-8

On Thu, Mar 17, 2016 at 12:18 PM, Dale R. Worley <worley@ariadne.com> wrote:

> Roman Shpount <roman@telurix.com> writes:
> > One small correction: In such a setup scenario, if edge proxies are
> > stateless and proxy fail-over takes less then 32 sec (Timer B), event SIP
> > transactions would normally survive an edge-proxy fail-over.
>
> My concern about transaction failure is this:
>
> UAC sends request
> edge-proxy 1 forwards request to UAS, retaining transaction state
> edge-proxy 1 fails
> edge-proxy 2 replaces edge-proxy 1
> UAS sends response to edge-proxy 2
> edge-proxy 2 has no transaction state to match the response and discards it
> UAS resends response to edge-proxy 2
> ...
>
> I'm sure that the real scenario is more complicated than that, but
> there's a general problem if the edge-proxy, or any proxy in the chain,
> is transaction-stateful, and you generally want proxies to be
> transaction-stateful.
>

It is quite common (especially when solutions are designed for scale) that
entire proxy stack is stateless or carries relevant state in via or route
headers.

But I think we're OK in practice, in that real UAs are resistant to
> transaction failure, transaction failures degrade the user experience
> but the user can easily recover from it.  And transaction failure will
> be quite uncommon.
>

The main transaction failure recovery issue is recovery of initial INVITE
transaction. This transaction is often long (waiting for remote party to
pick up) and user experience can be inconvenient (call was picked up and
dropped), especially if early media is used (started to talk to the person
and got dropped).
_____________
Roman Shpount

--001a113eca0681a1b4052e410a72
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Thu, Mar 17, 2016 at 12:18 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"><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"><span class=3D"">Roman Shpount &lt;<a href=3D"mailto:roman@telurix.c=
om">roman@telurix.com</a>&gt; writes:<br>
&gt; One small correction: In such a setup scenario, if edge proxies are<br=
>
&gt; stateless and proxy fail-over takes less then 32 sec (Timer B), event =
SIP<br>
&gt; transactions would normally survive an edge-proxy fail-over.<br>
<br>
</span>My concern about transaction failure is this:<br>
<br>
UAC sends request<br>
edge-proxy 1 forwards request to UAS, retaining transaction state<br>
edge-proxy 1 fails<br>
edge-proxy 2 replaces edge-proxy 1<br>
UAS sends response to edge-proxy 2<br>
edge-proxy 2 has no transaction state to match the response and discards it=
<br>
UAS resends response to edge-proxy 2<br>
...<br>
<br>
I&#39;m sure that the real scenario is more complicated than that, but<br>
there&#39;s a general problem if the edge-proxy, or any proxy in the chain,=
<br>
is transaction-stateful, and you generally want proxies to be<br>
transaction-stateful.<br></blockquote><div><br></div><div>It is quite commo=
n (especially when solutions are designed for scale) that entire proxy stac=
k is stateless or carries relevant state in via or route headers.</div><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);padding-left:1ex">But I think we&#39;re OK in practice, in that r=
eal UAs are resistant to<br>
transaction failure, transaction failures degrade the user experience<br>
but the user can easily recover from it.=C2=A0 And transaction failure will=
<br>
be quite uncommon.<br></blockquote><div><br></div><div>The main transaction=
 failure recovery issue is recovery of initial INVITE transaction. This tra=
nsaction is often long (waiting for remote party to pick up) and user exper=
ience can be inconvenient (call was picked up and dropped), especially if e=
arly media is used (started to talk to the person and got dropped).</div><d=
iv><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></div=
><div>=C2=A0</div></div></div></div>

--001a113eca0681a1b4052e410a72--


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--001a113eca064e13b6052e411f27
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On Thu, Mar 17, 2016 at 12:11 PM, Dale R. Worley <worley@ariadne.com> wrote:

> Roman Shpount <roman@telurix.com> writes:
> > 4. On the proxy side store connection specific server private key, client
> > public key and crypto capabilities in the same DB as registration
> > information.
>
> It struck me that the central problem was encryption without maintaining
> connection-specific encryption information.  If we're willing to
> maintain that, presumably DTLS would work.  I don't know much about
> DTLS, though -- do you think that UDP/DTLS would be a sufficient
> solution to the use cases we are discussing?
>

There is some shared state which would be inherently present and shared
between the edge-proxies. Typically it is client user ID and specific
IP:port on which they are registered. As long as client specific state is
clearly identified and can be easily shared without too many shared DB
updates (you do want to avoid updating shared DB on each UDP message sent
or received), UDP based solution can be implemented. I think DTLS
definitely needs to be investigated regarding state sharing. Part of the
problem is that DTLS is fairly complex all inclusive protocol with multiple
key negotiation, encryption and handshake schemes. It might need to be
restricted to allow state sharing. Also, none of the common DTLS
implementations, such as OpenSSL, provide support for such state sharing.

Also, the amount of stored state can be reduced if signed nonce is used
with each transaction. For instance server can provide client with
encryption key ID in the nonce and sign it, instead of keeping this ID in
the per connection storage. This is something not supported with DTLS.

This is definitely something that requires further investigation, but I
think the right way to do this is to start with DTLS and see if it can be
adapted or improved for connection fail-over scenario.
_____________
Roman Shpount

--001a113eca064e13b6052e411f27
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div>On Thu, Mar 17, 2016 =
at 12:11 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" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:sol=
id;border-left-color:rgb(204,204,204);padding-left:1ex"><span>Roman Shpount=
 &lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.c=
om</a>&gt; writes:<br>
&gt; 4. On the proxy side store connection specific server private key, cli=
ent<br>
&gt; public key and crypto capabilities in the same DB as registration<br>
&gt; information.<br>
<br>
</span>It struck me that the central problem was encryption without maintai=
ning<br>
connection-specific encryption information.=C2=A0 If we&#39;re willing to<b=
r>
maintain that, presumably DTLS would work.=C2=A0 I don&#39;t know much abou=
t<br>
DTLS, though -- do you think that UDP/DTLS would be a sufficient<br>
solution to the use cases we are discussing?<br></blockquote><div><br></div=
><div>There is some shared state which would be inherently present and shar=
ed between the edge-proxies. Typically it is client user ID and specific IP=
:port on which they are registered. As long as client specific state is cle=
arly identified and can be easily shared without too many shared DB updates=
 (you do want to avoid updating shared DB on each UDP message sent or recei=
ved), UDP based solution can be implemented. I think DTLS definitely needs =
to be investigated regarding state sharing. Part of the problem is that DTL=
S is fairly complex all inclusive protocol with multiple key negotiation, e=
ncryption and handshake schemes. It might need to be restricted to allow st=
ate sharing. Also, none of the common DTLS implementations, such as OpenSSL=
, provide support for such state sharing.</div><div><br></div><div>Also, th=
e amount of stored state can be reduced if signed nonce is used with each t=
ransaction. For instance server can provide client with encryption key ID i=
n the nonce and sign it, instead of keeping this ID in the per connection s=
torage. This is something not supported with DTLS.</div><div><br></div><div=
>This is definitely something that requires further investigation, but I th=
ink the right way to do this is to start with DTLS and see if it can be ada=
pted or improved for connection fail-over scenario.</div><div><div>________=
_____<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--001a113eca064e13b6052e411f27--


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From: worley@ariadne.com (Dale R. Worley)
To: Roman Shpount <roman@telurix.com>
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Roman Shpount <roman@telurix.com> writes:
> This is definitely something that requires further investigation, but I
> think the right way to do this is to start with DTLS and see if it can be
> adapted or improved for connection fail-over scenario.

I agree that the things you've suggested for SIP/DTLS could be quite
useful.  And in my brief study of RFC 5246 I've spotted some items that
need to be addressed.

I looked at draft-jennings-sip-dtls-05, but it is very brief and does
not address any of the concerns.  I know that it did not progress to an
RFC, but I don't know why.  I wonder if we should organize an effort to
construct SIP/DTLS to be used in these situations (UA to edge proxy),
with identification of the requirements, usage scenarios, etc.

Dale


From nobody Mon Mar 21 12:55:58 2016
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Subject: [sipcore] WGLC: draft-ietf-sip-dual-stack
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[as chair]

Today marks the start of a three-week Working Group Last Call for the 
following document:

https://datatracker.ietf.org/doc/draft-ietf-sipcore-dns-dual-stack/

The end of this last call is specifically intended to end on the Friday 
of the upcoming face-to-face meeting in Buenos Aires. Our plan is to ask 
the authors to incorporate any feedback received during these final 
weeks as well as comments from the face-to-face meeting into a version 
of the document that will be ready to send to the IESG.

All Working Group participants are requested to review the document and 
send comments to the SIPCORE mailing list.

/a


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Subject: [sipcore] Proposed SIPCORE Agenda
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[as chair]

We have uploaded a preliminary agenda for the upcoming SIPCORE meeting 
in Buenos Aires:

https://www.ietf.org/proceedings/95/agenda/agenda-95-sipcore

Any comments on the proposed agenda should be made to the chairs before 
the end of this week.

Thanks!

/a


From nobody Tue Mar 22 06:39:29 2016
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Hi all,

I think this is quite relevant to folks on this mailing list, in particular
given that a number of recent WG documents have been a result of SIP Forum
task groups.  While it's a bit past the CFP deadline, if you are interested
in submitting something, please contact Marc Robins:
http://www.sipforum.org/content/view/374/275/

Regards,
Mary.


---------- Forwarded message ----------
From: Marc Robins <marc.robins@sipforum.org>
Date: Thu, Mar 17, 2016 at 12:23 PM
Subject: [SIPForum-fullmembers] SIPNOC 2016 Early-Bird Extended till March
31
To: fullmembers@sipforum.org


SIP Forum Full Members,



Stop dragging your feet and register for SIPNOC 2016 today!



To help grease your wheels, we=E2=80=99ve extended the early bird rate till=
 March
31=E2=80=A6take advantage of the savings and click your way to a conference=
 pass
today.
------------------------------

*SIPNOC 2016 General Info*

The SIPNOC 2016 event website is located at www.sipnoc.org.

Special pricing for SIPNOC 2016 is now in effect! Take advantage of special
early-bird pricing now through March 31, 2016. The regular SIPNOC 2016
conference registration fee is $995, but for a limited time *individuals
from SIP Forum Full Member companies save $220 off the regular price with
special early-bird pricing. *This "all-inclusive" rate includes one full
day of SIP Training and two days of conference proceedings, including all
meals (breakfasts, lunches and dinners) and break refreshments.

*Please contact Marc Robins, SIP Forum President and Managing Director, at
marc.robins@sipforum.org <marc.robins@sipforum.org> to obtain the exclusive
Full Member discount code.*

Register today at http://www.regonline.com/sipnoc2016.
------------------------------

*SIPNOC 2016 Sponsorship Information*

For information about corporate sponsorship opportunities at SIPNOC 2016,
please contact Marc Robins at 203-829-6307 or marc.robins@sipforum.org.
------------------------------



_______________________________________________
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<div dir=3D"ltr"><div class=3D"gmail_quote">Hi all,</div><div class=3D"gmai=
l_quote"><br></div><div class=3D"gmail_quote">I think this is quite relevan=
t to folks on this mailing list, in particular given that a number of recen=
t WG documents have been a result of SIP Forum task groups.=C2=A0 While it&=
#39;s a bit past the CFP deadline, if you are interested in submitting some=
thing, please contact Marc Robins: =C2=A0<a href=3D"http://www.sipforum.org=
/content/view/374/275/">http://www.sipforum.org/content/view/374/275/</a></=
div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">Regards=
,</div><div class=3D"gmail_quote">Mary.<br><br><div dir=3D"ltr"><br><div cl=
ass=3D"gmail_quote">---------- Forwarded message ----------<br>From: <b cla=
ss=3D"gmail_sendername">Marc Robins</b> <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:marc.robins@sipforum.org" target=3D"_blank">marc.robins@sipforum.org</=
a>&gt;</span><br>Date: Thu, Mar 17, 2016 at 12:23 PM<br>Subject: [SIPForum-=
fullmembers] SIPNOC 2016 Early-Bird Extended till March 31<br>To: <a href=
=3D"mailto:fullmembers@sipforum.org" target=3D"_blank">fullmembers@sipforum=
.org</a><br><br><br><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div=
><p class=3D"MsoNormal"><span style=3D"font-size:12pt">SIP Forum<span style=
=3D"color:rgb(31,73,125)"> </span>Full<span style=3D"color:rgb(31,73,125)">=
 </span>Members,<u></u><u></u></span></p><p class=3D"MsoNormal"><span style=
=3D"font-size:12pt"><u></u>=C2=A0<u></u></span></p><p class=3D"MsoNormal"><=
span style=3D"font-size:12pt">Stop dragging your feet and register for SIPN=
OC 2016 today!<u></u><u></u></span></p><p class=3D"MsoNormal"><span style=
=3D"font-size:12pt"><u></u>=C2=A0<u></u></span></p><p class=3D"MsoNormal"><=
span style=3D"font-size:12pt">To help grease your wheels, we=E2=80=99ve ext=
ended the early bird rate till March 31=E2=80=A6take advantage of the savin=
gs and click your way to a conference pass today.<u></u><u></u></span></p><=
div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span =
style=3D"font-size:12pt;font-family:&#39;MS PGothic&#39;,sans-serif"><hr si=
ze=3D"2" width=3D"100%" align=3D"center"></span></div><p><b><span style=3D"=
font-family:Calibri,sans-serif">SIPNOC 2016 General Info<u></u><u></u></spa=
n></b></p><p><span style=3D"font-family:Calibri,sans-serif">The SIPNOC 2016=
 event website is located at <a href=3D"http://www.sipnoc.org" target=3D"_b=
lank">www.sipnoc.org</a>.<u></u><u></u></span></p><p><span style=3D"font-fa=
mily:Calibri,sans-serif">Special pricing for SIPNOC 2016 is now in effect! =
Take advantage of special early-bird pricing now through March 31, 2016. Th=
e regular SIPNOC 2016 conference registration fee is $995, but for a limite=
d time <b><span style=3D"color:rgb(112,48,160)">i</span><span style=3D"colo=
r:purple">ndividuals from SIP Forum Full Member companies save $220 off the=
 regular price with special early-bird pricing. </span></b>This &quot;all-i=
nclusive&quot; rate includes one full day of SIP Training and two days of c=
onference proceedings, including all meals (breakfasts, lunches and dinners=
) and break refreshments.<u></u><u></u></span></p><p><b><span style=3D"font=
-family:Calibri,sans-serif;color:purple">Please contact Marc Robins, SIP Fo=
rum President and Managing Director, at <a href=3D"mailto:marc.robins@sipfo=
rum.org" target=3D"_blank">marc.robins@sipforum.org</a> to obtain the exclu=
sive Full Member discount code.</span></b><u></u><u></u></p><p><span style=
=3D"font-family:Calibri,sans-serif">Register today at <a href=3D"http://www=
.regonline.com/sipnoc2016" target=3D"_blank">http://www.regonline.com/sipno=
c2016</a>.=C2=A0<u></u><u></u></span></p><div class=3D"MsoNormal" align=3D"=
center" style=3D"text-align:center"><span style=3D"font-size:12pt"><hr size=
=3D"2" width=3D"100%" align=3D"center"></span></div><p><b><span style=3D"fo=
nt-family:Calibri,sans-serif">SIPNOC 2016 Sponsorship Information<u></u><u>=
</u></span></b></p><p><span style=3D"font-family:Calibri,sans-serif">For in=
formation about corporate sponsorship opportunities at SIPNOC 2016, please =
contact Marc Robins<span style=3D"color:rgb(31,73,125)"> </span>at <a href=
=3D"tel:203-829-6307" value=3D"+12038296307" target=3D"_blank">203-829-6307=
</a> or <a href=3D"mailto:marc.robins@sipforum.org" target=3D"_blank">marc.=
robins@sipforum.org</a>.<u></u><u></u></span></p><div class=3D"MsoNormal" a=
lign=3D"center" style=3D"text-align:center"><span style=3D"font-size:12pt">=
<hr size=3D"2" width=3D"100%" align=3D"center"></span></div><p class=3D"Mso=
Normal"><u></u>=C2=A0<u></u></p></div></div><br>___________________________=
____________________<br>
fullmembers mailing list<br>
<a href=3D"mailto:fullmembers@sipforum.org" target=3D"_blank">fullmembers@s=
ipforum.org</a><br>
<a href=3D"http://sipforum.org/mailman/listinfo/fullmembers" rel=3D"norefer=
rer" target=3D"_blank">http://sipforum.org/mailman/listinfo/fullmembers</a>=
<br>
<br></div><br></div>
</div><br></div>

--94eb2c04fa58a6d0b7052ea35332--


From nobody Sun Mar 27 14:27:53 2016
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Outline of a SIP-DTLS document
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I've written a draft that would be the start of the process of producing
a version of DTLS for edge access in SIP, assembling a set of
requirements.

Do people think this is worth pursuing?  If so, please speak up.  We can
do a preliminary job of assembling requirements, and then send this to
Dispatch for consideration there.

Dale
----------------------------------------------------------------------
[version of 27 March 2016]

A lightweight, secure, scalable edge access protocol for SIP

Introduction

This document describes the requirements for a lightweight, secure,
scalable edge access protocol for SIP.  It is intended for large-scale
SIP networks, where a constellation of redundant edge proxies
maintains flows with hundreds of thousands or millions of user agents.

"Edge access" means that the protocol is a transport protocol intended
primarily for use between user agents and edge proxies.  "Lightweight"
means that the protocol maintains minimal state for each flow between
a user agent and a proxy, thus easing the load on the edge proxies and
minimizing the amount of state that needs to transferred upon proxy
failover.  "Secure" means a level of security similar to that provided
by TLS.

In addition, we require logical structure of the protocol to be fairly
simple, so that it will be broadly implemented.

It is expected that these requirements will be used to design an
adaptation of DTLS for use as a transport for SIP.

Terminology:

Client:  the role of one endpoint of the transport connection,
typically a SIP user agent

Server:  the role of the other endpoint of the transport connection,
typically a SIP edge proxy for a SIP domain

Flow:  an logically-associated series of SIP messages (requests and
responses) between a client and a server

Requirements and non-requirements

REQ:  The protocol is a transport protocol for use in the hop between
a client and server.

Non-REQ:  The protocol need not be symmetric between the client and
server roles.

REQ:  The protocol must have a fairly simple logical structure.

REQ:  The protocol must allow over 1 million flows to be supported by
a single server with current commercial technology.

REQ:  The amount of state the server must maintain for a single flow
must be small, of the size of the state that must be maintained for a
SIP registration.

This allows the flow state for a server to be held in a failover
database of reasonable size.

REQ:  Once a flow is established, passing messages in either direction
must typically require no update to the flow state.

This keeps the update rate of the database proportionate to the rate
of creating and destroying flows, rather than proportional to the
message volume.

Non-REQ:  If a client loses its flow state, reestablishing the flow
may require multiple round trip times.

We assume that if a client loses its flow state, it will also lose any
current dialogs and sessions.  Thus, it does not need to recover flow
state quickly in order to keep current calls alive.

REQ:  The protocol must provide message privacy and integrity.

Non-REQ:  The protocol does not need to provide replay protection.

The use of SIP digest authentication with time-stamped nonces can
provide a substantial degree of replay protection.

REQ:  The protocol must allow authentication of the client to the
server and/or authentication of the server to the client.

REQ:  Authentication credentials (in a single direction) can be either
a certificate or a realm, username, and shared secret.

REQ:  The protocol must allow the client to present a SIP domain to
the server, so that the server can select an appropriate realm to
present to the client.

REQ:  The protocol must allow the server to present a realm to the
client, so that the client can select appropriate credentials to
present to the server, and which the client can use to validate the
credentials presented by the server.

These requirements mirror the dataflow of digest authentication.

REQ:  The protocol's security mechanism must be independent of any SIP
authentication (other than that SIP credentials can be reused for
realm/username/secret authentication, if the devices are configured to
do so).

Non-REQ:  A flow's protocol state may be bound to the client's
transport address as seen by the server (inter alia).

REQ:  If a client's flow becomes invalidated at the server (e.g., due
to a change in the client's public address), the client must be able
to detect this fact quickly (within a few RTTs), so that it can
reestablish a flow quickly to prevent outstanding transactions from
failing.

*** Is this needed?  Is this possible?

REQ:  If a client's flow becomes invalidated at the server due to a
change in the client's public address, the client must be able to
recover in a way that does not invalidate the route set of existing
dialogs.

*** Presumably this requires using GRUUs, which would be specified in
the document.  Is it possible to make this work?

Motivations

In the most recent SIPit interoperability event, SIPit 31 of October
2014, 86% of user agents supported TCP, and 81% supported TLS.  This
is 12 years after RFC 3261 envisioned universal support for both TCP
and TLS in SIP devices.

>From draft-jennings-sip-dtls-01

   There has been considerable discussion of why SIP needs DTLS when we
   have TLS.  This is the wrong question.  The right question is why SIP
   has UDP and TCP (not to mention SCTP).  There are two reasons for
   believing that UDP is likely to be an important protocol in SIP for
   the foreseeable future.

   o  In theory, there is no problem building systems that terminate a
      million TCP connections on a single host.  In practice, the common
      operating systems used for building SIP aggregation devices make
      this impossible.  To date, no one has demonstrated terminating
      over 100k SIP TCP connections to a single host.  Doing that many
      connections with UDP has not been difficult.
   o  If we want to talk about "running code" for SIP, it's UDP.  Unless
      UDP is deprecated for SIP, it is important to provide a reasonable
      level of security for it.

>From Roman Shpount <roman@telurix.com>
http://mailarchive.ietf.org/arch/msg/sipcore/jjALFjeefA1WjwmVSkv_jAsqsNQ

    Since everyone is talking about how desirable TLS is and how easy it is to
    deprecate UDP, can anybody point me to large scale implementations of
    TLS+Outbound? I know a lot of implementations for UDP which handle 1M+ of
    concurrent registrations. When someone tries to implement the same using
    TLS and Outbound it much harder:

    1. Maintaining large number of TLS connections is resource intensive. If
    you have 1M concurrent registrations you need edge proxies which will
    terminate 1M concurrent TLS connections and respond to keep alive messages.
    This requires 10x number the servers then UDP registrations. This also
    exposes you to more risk of DOS attacks

    2. Fail-over is much harder: For TLS/Outbound you either need to put edge
    proxy on VM that you can migrate to a different server (still had to deal
    with software failure). To deal with software failures, you need to design
    the mechanism for the client to recover the connection to a different edge
    proxy. If call is in progress this involves for the client detecting the
    flow failure based on keep alive failure, re-registering to get the gruu,
    and re-INVITE with new gruu in the contact. If server needs to send a
    message on the client call while this is happening, it needs to queue this
    message or implement some sort of non-standard re-transmit timer. This
    typically means call gets dropped. There is no mechanism at all to recover
    the transaction which is in progress while outbound proxy fails. For UDP
    you can move the IP address for the edge proxy to a different server. If
    registration and flow state is in some sort of shared DB, the fail-over
    occurs naturally for both the client and server.

    I went through couple of large scale deployments of Outbound with SIPS for
    large number of consumer devices. In all of them SIP registrations were
    scrapped and replaced with something proprietary.

>From Roman Shpount:
http://mailarchive.ietf.org/arch/msg/sipcore/y4E8MTU4tUVHjw7MlZnaMAzuRK8

    Multiple flows somewhat mitigate the problem of edge proxy failure at the
    cost of multiplying the cost of connection management. There is still and
    issue of temporary network failure between the end point and the proxy. In
    case of TLS this terminates all flows from the client to all proxies and
    typically terminates the call. And then there is a use case of end point
    migrating from one IP to another. Here the quicker flow recovery with UDP
    increases the chances of keeping the call running vs 7 or 8 round trips
    required to recover the TLS flow (TCP connection setup, TLS connection
    setup, registration with response and then re-invite with new GRUU). And
    then there is a use case when client used a temporary GRUU which was flow
    specific. And there is still no way to recover the transaction since, I
    believe, transactions are flow bound.

>From Roman Shpount:
http://mailarchive.ietf.org/arch/msg/sipcore/Oq5xEUlOJ5eM3Mn9HeHxNUUJX5Y

    DTLS is not an answer either since encryption state will need to be failed
    over with the edge proxy. What you need is a protocol where client can
    recover a flow to the server, including encryption, using a single round
    trip.

    I think deployments of large numbers of SIP devices connected to the public
    service (like SIP based Skype) never happened and will never happen because
    of the complexity and design issues of SIP Outbound in combination with
    TLS. I think this is why none of the large scale consumer VoIP applications
    used SIP. The companies that built these products used RTP, ICE and media
    components, but SIP was ignored. I do not think all these companies spent
    all the design effort to build something completely new just because they
    wanted to build something different. From my experience with couple of
    similar projects the issues of scaling the deployment and handling
    redundancy were the main driving factors.

    Putting aside all the problems with SIP Outbound, there are still a lot of
    companies which offer SIP over UDP to the end users. I am talking about all
    the hosted PBX solutions. Most of these deployments do not use ICE and use
    host based NAT traversal for signaling and media to keep the message size
    small. This is not a solution you would design if you have started the
    design from scratch, but these installations definitely need a migration
    path from IPv4 to IPv6. Telling them to switch to SIPS with SIP Outbound is
    unrealistic, so I believe Happy Eyeballs must provide the solution for UDP.

References

draft-jennings-sip-dtls-01, "Using DTLS as a Transport for SIP",
C. Jennings, N. Modadugu

Messages from Roman Shpount to the Sipcore mailing list.

RFC 3261

[END]


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Date: Sun, 27 Mar 2016 17:51:38 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Dale R. Worley" <worley@ariadne.com>
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Subject: Re: [sipcore] Outline of a SIP-DTLS document
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I obviously do this think this is worth pursuing.

Few more comments inline.

On Sun, Mar 27, 2016 at 5:27 PM, Dale R. Worley <worley@ariadne.com> wrote:

> REQ:  If a client's flow becomes invalidated at the server (e.g., due
> to a change in the client's public address), the client must be able
> to detect this fact quickly (within a few RTTs), so that it can
> reestablish a flow quickly to prevent outstanding transactions from
> failing.
>
> *** Is this needed?  Is this possible?
>

Typically using STUN keep alive messages (
https://tools.ietf.org/html/rfc5626#section-3.5.2) allows UDP client detect
that it's public address changed and re-register. To prevent call dialogs
from being torn down by SIP session timer this requires flow failure
detection that is faster then typical transaction timeout (32 sec) which is
possible with keep alive messages sent every 10-15 sec vs recommended 24-29
sec.

REQ:  If a client's flow becomes invalidated at the server due to a
> change in the client's public address, the client must be able to
> recover in a way that does not invalidate the route set of existing
> dialogs.
>
> *** Presumably this requires using GRUUs, which would be specified in
> the document.  Is it possible to make this work?
>

It should be possible to indicate that this is the same device being
registered and allow to preserve the same GRUU.

Once more benefit to consider -- DTLS provides message fragmentation
support which will allow using such DTLS based protocol with larger sized
SIP messages that do not fit into a single UDP packet.

Regards,
_____________
Roman Shpount

--047d7bd75708ff093f052f0ecc48
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I obviously do this think this is worth=C2=A0<span style=
=3D"color:rgb(0,0,0);font-size:12.8px">pursuing</span>.<div class=3D"gmail_=
extra"><br></div><div class=3D"gmail_extra">Few more comments inline.<br cl=
ear=3D"all"><div><div class=3D"gmail_signature"><br></div></div><div class=
=3D"gmail_quote">On Sun, Mar 27, 2016 at 5:27 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><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:soli=
d;border-left-color:rgb(204,204,204);padding-left:1ex">REQ:=C2=A0 If a clie=
nt&#39;s flow becomes invalidated at the server (e.g., due<br>
to a change in the client&#39;s public address), the client must be able<br=
>
to detect this fact quickly (within a few RTTs), so that it can<br>
reestablish a flow quickly to prevent outstanding transactions from<br>
failing.<br>
<br>
*** Is this needed?=C2=A0 Is this possible?<br></blockquote><div><br></div>=
<div>Typically using STUN keep alive messages (<a href=3D"https://tools.iet=
f.org/html/rfc5626#section-3.5.2">https://tools.ietf.org/html/rfc5626#secti=
on-3.5.2</a>) allows UDP client detect that it&#39;s public address changed=
 and re-register. To prevent call dialogs from being torn down by SIP sessi=
on timer this requires flow failure detection that is faster then typical t=
ransaction timeout (32 sec) which is possible with keep alive messages sent=
 every 10-15 sec vs recommended 24-29 sec.</div><div><br></div><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1=
px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:=
1ex">REQ:=C2=A0 If a client&#39;s flow becomes invalidated at the server du=
e to a<br>
change in the client&#39;s public address, the client must be able to<br>
recover in a way that does not invalidate the route set of existing<br>
dialogs.<br>
<br>
*** Presumably this requires using GRUUs, which would be specified in<br>
the document.=C2=A0 Is it possible to make this work?<br></blockquote><div>=
=C2=A0</div><div>It should be possible to indicate that this is the same de=
vice being registered and allow to preserve the same GRUU.</div><div><br></=
div><div>Once more benefit to consider -- DTLS provides message fragmentati=
on support which will allow using such DTLS based protocol with larger size=
d SIP messages that do not fit into a single UDP packet.</div><div><br></di=
v><div>Regards,</div><div><div><div class=3D"gmail_signature">_____________=
<br>Roman Shpount</div></div></div><div><br></div></div></div></div>

--047d7bd75708ff093f052f0ecc48--


From nobody Sun Mar 27 18:01:07 2016
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [sipcore] Outline of a SIP-DTLS document
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Subject: Re: [sipcore] Outline of a SIP-DTLS document
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+1

As I've told Dale privately, I'm interested in pursuing this and willing to=
 contribute to it.

Regards,

Gonzalo



On Mar 27, 2016, at 6:51 PM, Roman Shpount <roman@telurix.com<mailto:roman@=
telurix.com>> wrote:

I obviously do this think this is worth pursuing.

Few more comments inline.

On Sun, Mar 27, 2016 at 5:27 PM, Dale R. Worley <worley@ariadne.com<mailto:=
worley@ariadne.com>> wrote:
REQ:  If a client's flow becomes invalidated at the server (e.g., due
to a change in the client's public address), the client must be able
to detect this fact quickly (within a few RTTs), so that it can
reestablish a flow quickly to prevent outstanding transactions from
failing.

*** Is this needed?  Is this possible?

Typically using STUN keep alive messages (https://tools.ietf.org/html/rfc56=
26#section-3.5.2) allows UDP client detect that it's public address changed=
 and re-register. To prevent call dialogs from being torn down by SIP sessi=
on timer this requires flow failure detection that is faster then typical t=
ransaction timeout (32 sec) which is possible with keep alive messages sent=
 every 10-15 sec vs recommended 24-29 sec.

REQ:  If a client's flow becomes invalidated at the server due to a
change in the client's public address, the client must be able to
recover in a way that does not invalidate the route set of existing
dialogs.

*** Presumably this requires using GRUUs, which would be specified in
the document.  Is it possible to make this work?

It should be possible to indicate that this is the same device being regist=
ered and allow to preserve the same GRUU.

Once more benefit to consider -- DTLS provides message fragmentation suppor=
t which will allow using such DTLS based protocol with larger sized SIP mes=
sages that do not fit into a single UDP packet.

Regards,
_____________
Roman Shpount

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

--_000_61244F12BD4F4A5D9DFA7FD875AFE777ciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>&#43;1</div>
<div id=3D"AppleMailSignature"><br>
</div>
<div id=3D"AppleMailSignature">As I've told Dale privately, I'm interested =
in pursuing this and willing to contribute to it.<br>
<br>
<div>Regards,</div>
<div><br>
</div>
<div>Gonzalo</div>
<div><br>
</div>
<br>
</div>
<div><br>
On Mar 27, 2016, at 6:51 PM, Roman Shpount &lt;<a href=3D"mailto:roman@telu=
rix.com">roman@telurix.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">I obviously do this think this is worth&nbsp;<span style=
=3D"color:rgb(0,0,0);font-size:12.8px">pursuing</span>.
<div class=3D"gmail_extra"><br>
</div>
<div class=3D"gmail_extra">Few more comments inline.<br clear=3D"all">
<div>
<div class=3D"gmail_signature"><br>
</div>
</div>
<div class=3D"gmail_quote">On Sun, Mar 27, 2016 at 5:27 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>
<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">
REQ:&nbsp; If a client's flow becomes invalidated at the server (e.g., due<=
br>
to a change in the client's public address), the client must be able<br>
to detect this fact quickly (within a few RTTs), so that it can<br>
reestablish a flow quickly to prevent outstanding transactions from<br>
failing.<br>
<br>
*** Is this needed?&nbsp; Is this possible?<br>
</blockquote>
<div><br>
</div>
<div>Typically using STUN keep alive messages (<a href=3D"https://tools.iet=
f.org/html/rfc5626#section-3.5.2">https://tools.ietf.org/html/rfc5626#secti=
on-3.5.2</a>) allows UDP client detect that it's public address changed and=
 re-register. To prevent call dialogs
 from being torn down by SIP session timer this requires flow failure detec=
tion that is faster then typical transaction timeout (32 sec) which is poss=
ible with keep alive messages sent every 10-15 sec vs recommended 24-29 sec=
.</div>
<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">
REQ:&nbsp; If a client's flow becomes invalidated at the server due to a<br=
>
change in the client's public address, the client must be able to<br>
recover in a way that does not invalidate the route set of existing<br>
dialogs.<br>
<br>
*** Presumably this requires using GRUUs, which would be specified in<br>
the document.&nbsp; Is it possible to make this work?<br>
</blockquote>
<div>&nbsp;</div>
<div>It should be possible to indicate that this is the same device being r=
egistered and allow to preserve the same GRUU.</div>
<div><br>
</div>
<div>Once more benefit to consider -- DTLS provides message fragmentation s=
upport which will allow using such DTLS based protocol with larger sized SI=
P messages that do not fit into a single UDP packet.</div>
<div><br>
</div>
<div>Regards,</div>
<div>
<div>
<div class=3D"gmail_signature">_____________<br>
Roman Shpount</div>
</div>
</div>
<div><br>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>sipcore mailing list</span><br>
<span><a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www=
.ietf.org/mailman/listinfo/sipcore</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_61244F12BD4F4A5D9DFA7FD875AFE777ciscocom_--


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [sipcore] Outline of a SIP-DTLS document
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Cc: "Dale R. Worley" <worley@ariadne.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Outline of a SIP-DTLS document
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Yes, certainly interesting/valuable work.

And the same here as far as contributing is concerned.

Thanks,
Tolga

From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Gonzalo Salgue=
iro (gsalguei)
Sent: Sunday, March 27, 2016 8:52 PM
To: Roman Shpount <roman@telurix.com>
Cc: Dale R. Worley <worley@ariadne.com>; sipcore@ietf.org
Subject: Re: [sipcore] Outline of a SIP-DTLS document

+1

As I've told Dale privately, I'm interested in pursuing this and willing to=
 contribute to it.
Regards,

Gonzalo



On Mar 27, 2016, at 6:51 PM, Roman Shpount <roman@telurix.com<mailto:roman@=
telurix.com>> wrote:
I obviously do this think this is worth pursuing.

Few more comments inline.

On Sun, Mar 27, 2016 at 5:27 PM, Dale R. Worley <worley@ariadne.com<mailto:=
worley@ariadne.com>> wrote:
REQ:  If a client's flow becomes invalidated at the server (e.g., due
to a change in the client's public address), the client must be able
to detect this fact quickly (within a few RTTs), so that it can
reestablish a flow quickly to prevent outstanding transactions from
failing.

*** Is this needed?  Is this possible?

Typically using STUN keep alive messages (https://tools.ietf.org/html/rfc56=
26#section-3.5.2) allows UDP client detect that it's public address changed=
 and re-register. To prevent call dialogs from being torn down by SIP sessi=
on timer this requires flow failure detection that is faster then typical t=
ransaction timeout (32 sec) which is possible with keep alive messages sent=
 every 10-15 sec vs recommended 24-29 sec.

REQ:  If a client's flow becomes invalidated at the server due to a
change in the client's public address, the client must be able to
recover in a way that does not invalidate the route set of existing
dialogs.

*** Presumably this requires using GRUUs, which would be specified in
the document.  Is it possible to make this work?

It should be possible to indicate that this is the same device being regist=
ered and allow to preserve the same GRUU.

Once more benefit to consider -- DTLS provides message fragmentation suppor=
t which will allow using such DTLS based protocol with larger sized SIP mes=
sages that do not fit into a single UDP packet.

Regards,
_____________
Roman Shpount

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

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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Yes, certainly interesting/valuable w=
ork.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">And the same here as far as contribut=
ing is concerned.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Thanks,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Tolga<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,sans-serif">From:</span></b><span style=3D"font-size:11.0pt;=
font-family:&quot;Calibri&quot;,sans-serif"> sipcore [mailto:sipcore-bounce=
s@ietf.org]
<b>On Behalf Of </b>Gonzalo Salgueiro (gsalguei)<br>
<b>Sent:</b> Sunday, March 27, 2016 8:52 PM<br>
<b>To:</b> Roman Shpount &lt;roman@telurix.com&gt;<br>
<b>Cc:</b> Dale R. Worley &lt;worley@ariadne.com&gt;; sipcore@ietf.org<br>
<b>Subject:</b> Re: [sipcore] Outline of a SIP-DTLS document<o:p></o:p></sp=
an></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">&#43;1<o:p></o:p></p>
</div>
<div id=3D"AppleMailSignature">
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div id=3D"AppleMailSignature">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">As I've told Dale pri=
vately, I'm interested in pursuing this and willing to contribute to it.<o:=
p></o:p></p>
<div>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Gonzalo<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
On Mar 27, 2016, at 6:51 PM, Roman Shpount &lt;<a href=3D"mailto:roman@telu=
rix.com">roman@telurix.com</a>&gt; wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal">I obviously do this think this is worth&nbsp;<span s=
tyle=3D"font-size:9.5pt;color:black">pursuing</span>.
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Few more comments inline.<br clear=3D"all">
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">On Sun, Mar 27, 2016 at 5:27 PM, Dale R. Worley &lt;=
<a href=3D"mailto:worley@ariadne.com" target=3D"_blank">worley@ariadne.com<=
/a>&gt; wrote:<o:p></o:p></p>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal">REQ:&nbsp; If a client's flow becomes invalidated at=
 the server (e.g., due<br>
to a change in the client's public address), the client must be able<br>
to detect this fact quickly (within a few RTTs), so that it can<br>
reestablish a flow quickly to prevent outstanding transactions from<br>
failing.<br>
<br>
*** Is this needed?&nbsp; Is this possible?<o:p></o:p></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Typically using STUN keep alive messages (<a href=3D=
"https://tools.ietf.org/html/rfc5626#section-3.5.2">https://tools.ietf.org/=
html/rfc5626#section-3.5.2</a>) allows UDP client detect that it's public a=
ddress changed and re-register. To prevent
 call dialogs from being torn down by SIP session timer this requires flow =
failure detection that is faster then typical transaction timeout (32 sec) =
which is possible with keep alive messages sent every 10-15 sec vs recommen=
ded 24-29 sec.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal">REQ:&nbsp; If a client's flow becomes invalidated at=
 the server due to a<br>
change in the client's public address, the client must be able to<br>
recover in a way that does not invalidate the route set of existing<br>
dialogs.<br>
<br>
*** Presumably this requires using GRUUs, which would be specified in<br>
the document.&nbsp; Is it possible to make this work?<o:p></o:p></p>
</blockquote>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">It should be possible to indicate that this is the s=
ame device being registered and allow to preserve the same GRUU.<o:p></o:p>=
</p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Once more benefit to consider -- DTLS provides messa=
ge fragmentation support which will allow using such DTLS based protocol wi=
th larger sized SIP messages that do not fit into a single UDP packet.<o:p>=
</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">_____________<br>
Roman Shpount<o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">_______________________________________________<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">https://www.ietf.=
org/mailman/listinfo/sipcore</a><o:p></o:p></p>
</div>
</blockquote>
</div>
</div>
</body>
</html>

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I have one comment that should be addressed before requesting 
publication of this document.

I find the formulation of section 5 confusing, and possibly misleading. 
I think it is too terse, and it unnecessarily trips up on some 
provenance pointers.

6157 doesn't defer to 6724 - it points out implementers should rely on a 
new system function call and that the function call is expected to 
implement an algorithm specified in 3484. The description of that 
algorithm is updated in 6724. But why are you pointing here anyhow? How 
is it going to help the implementer?

If you're saying the implementer needs to make sure their _own code_ 
implements what 6724 describes (as opposed to relying on a system call), 
the words in this document need to be very different.

If you're saying, instead, that they should just rely on getaddrinfo() 
after having done the SRV work and ordering to resolve A or AAAA records 
(and trust the order it returns the in), just say that.

It's really not clear _what_ in 6157 this is clarifying.

RjS


On 3/21/16 11:08 AM, Adam Roach wrote:
> [as chair]
>
> Today marks the start of a three-week Working Group Last Call for the 
> following document:
>
> https://datatracker.ietf.org/doc/draft-ietf-sipcore-dns-dual-stack/
>
> The end of this last call is specifically intended to end on the 
> Friday of the upcoming face-to-face meeting in Buenos Aires. Our plan 
> is to ask the authors to incorporate any feedback received during 
> these final weeks as well as comments from the face-to-face meeting 
> into a version of the document that will be ready to send to the IESG.
>
> All Working Group participants are requested to review the document 
> and send comments to the SIPCORE mailing list.
>
> /a
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Robert Sparks <rjsparks@nostrum.com> writes:
> It's really not clear _what_ in 6157 [section 5 of this draft] is clarifying.

You're right.

The current section 5 is my revision of the section 5 in the -02 draft,
which is

   5.  Update to RFC 6157

   [RFC6157] defers to the Source and Destination Address Selection
   algorithms defined in [RFC6724] (the successor of [RFC3484]) when
   allowing a client to choose a specific server (c.f.  Section 5 in
   [RFC6157]).

   This document modifies the behavior of Section 5 in [RFC6157] to
   allow for an additional (and preferred) way to contact servers, as
   outlined in Section Section 4.2.  Implementations MUST use the DNS
   SRV records as described in Section Section 4.2 of this document
   first before resorting to the Source and Destination Address
   Selection algorithms defined in [RFC6724].

As far as I can tell, the goal of this section is to make it clear that
if SRV records translate a SIP domain into (e.g.) two SRV records with
different priorities, and one SRV record's target name only has IPv4
addresses and the other SRV record's target name only has IPv6
addresses, then which family of addresses is attempted first is to be
determined by priorities of the SRV records.  This being the process
described in section 4.2 of this draft.

The alternative that the implementor seems to be being warned to avoid
seems to be to put both the IPv4 and IPv6 addresses together into the
algorithm of 3484/6724 and having it order the destination addresses.

I don't think this is an update of 6157 (as the section title suggests),
as following 3263 as written gives the behavior described in section 4.2
of this draft.  And that seems to be the behavior that 6157 will result
in as well, as 6157 seems to describe putting each SRV's target into
getaddrinfo() separately to obtain the target's destination addresses.

So I would say that eliminating this section entirely would not change
the meaning of the draft.  But that leaves open what this section was
intended to accomplish.  I'll ask the other authors.

Dale


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[as individual]

On 3/29/16 13:19, Dale R. Worley wrote:
> As far as I can tell, the goal of this section is to make it clear that
> if SRV records translate a SIP domain into (e.g.) two SRV records with
> different priorities, and one SRV record's target name only has IPv4
> addresses and the other SRV record's target name only has IPv6
> addresses, then which family of addresses is attempted first is to be
> determined by priorities of the SRV records.  This being the process
> described in section 4.2 of this draft.

I think it's somewhat more general than that, though. if you look up an 
SRV record and get multiple A and multiple AAAA records, and those 
records in turn contain multiple literal IP addresses, then you apply 
the SRV priorities first and the RFC 6724 processes second.

I think an example might help.

Let's say I look up an SRV record (_sip._tcp.example.com) and get back 
two entries:

_sip._tcp.example.com.  300 IN  SRV  10 1 5060 sip-1.example.com.
_sip._tcp.example.com.  300 IN  SRV  20 1 5060 sip-2.example.com.


So then I grab the A and AAAA records for sip-1.example.com and 
sip-2.example.com:

sip-1.example.com.    300  IN  AAAA  2001:0db8:58:c02::face
sip-1.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:cafe
sip-1.example.com.    300  IN  AAAA  2001:0db8:44:204::d1ce

sip-1.example.com.    1800  IN  A  192.0.2.45
sip-1.example.com.    1800  IN  A  203.0.113.109
sip-1.example.com.    1800  IN  A  198.51.100.24

sip-2.example.com.    300  IN  AAAA  2001:0db8:58:c02::dead
sip-2.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:beef
sip-2.example.com.    300  IN  AAAA  2001:0db8:44:204::c0de

sip-2.example.com.    1800  IN  A  192.0.2.75
sip-2.example.com.    1800  IN  A  203.0.113.38
sip-2.example.com.    1800  IN  A  198.51.100.140


What I believe we're saying is that I will process these addresses in a 
batch, altogether, using RFC6724 section 6 rules:

  * 2001:0db8:58:c02::face
  * 2001:0db8:c:a06::2:cafe
  * 2001:0db8:44:204::d1ce
  * 192.0.2.45
  * 203.0.113.109
  * 198.51.100.24


And then (but only then, after I've exhausted that entire batch of 
addresses), I'll process these addresses in a batch, altogether, using 
RFC6724 section 6 rules:

  * 2001:0db8:58:c02::dead
  * 2001:0db8:c:a06::2:beef
  * 2001:0db8:44:204::c0de
  * 192.0.2.75
  * 203.0.113.38
  * 198.51.100.140


... and, most importantly, I won't mix the batches together.

/a

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    <div class="moz-cite-prefix">[as individual]<br>
      <br>
      On 3/29/16 13:19, Dale R. Worley wrote:<br>
    </div>
    <blockquote cite="mid:87poudw3iv.fsf@hobgoblin.ariadne.com"
      type="cite">
      <pre wrap="">As far as I can tell, the goal of this section is to make it clear that
if SRV records translate a SIP domain into (e.g.) two SRV records with
different priorities, and one SRV record's target name only has IPv4
addresses and the other SRV record's target name only has IPv6
addresses, then which family of addresses is attempted first is to be
determined by priorities of the SRV records.  This being the process
described in section 4.2 of this draft.</pre>
    </blockquote>
    <br>
    I think it's somewhat more general than that, though. if you look up
    an SRV record and get multiple A and multiple AAAA records, and
    those records in turn contain multiple literal IP addresses, then
    you apply the SRV priorities first and the RFC 6724 processes
    second.<br>
    <br>
    I think an example might help.<br>
    <br>
    Let's say I look up an SRV record (_sip._tcp.example.com) and get
    back two entries:<br>
    <br>
    _sip._tcp.example.com.  300 IN  SRV  10 1 5060 sip-1.example.com.<br>
    _sip._tcp.example.com.  300 IN  SRV  20 1 5060 sip-2.example.com.<br>
    <br>
    <br>
    So then I grab the A and AAAA records for sip-1.example.com and
    sip-2.example.com:<br>
    <br>
    sip-1.example.com.    300  IN  AAAA  2001:0db8:58:c02::face<br>
    sip-1.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:cafe<br>
    sip-1.example.com.    300  IN  AAAA  2001:0db8:44:204::d1ce<br>
    <br>
    sip-1.example.com.    1800  IN  A  192.0.2.45<br>
    sip-1.example.com.    1800  IN  A  203.0.113.109<br>
    sip-1.example.com.    1800  IN  A  198.51.100.24<br>
    <br>
    sip-2.example.com.    300  IN  AAAA  2001:0db8:58:c02::dead<br>
    sip-2.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:beef<br>
    sip-2.example.com.    300  IN  AAAA  2001:0db8:44:204::c0de<br>
    <br>
    sip-2.example.com.    1800  IN  A  192.0.2.75<br>
    sip-2.example.com.    1800  IN  A  203.0.113.38<br>
    sip-2.example.com.    1800  IN  A  198.51.100.140<br>
    <br>
    <br>
    What I believe we're saying is that I will process these addresses
    in a batch, altogether, using RFC6724 section 6 rules:<br>
    <ul>
      <li>2001:0db8:58:c02::face</li>
      <li>2001:0db8:c:a06::2:cafe</li>
      <li>2001:0db8:44:204::d1ce</li>
      <li>192.0.2.45</li>
      <li>203.0.113.109</li>
      <li>198.51.100.24</li>
    </ul>
    <br>
    And then (but only then, after I've exhausted that entire batch of
    addresses), I'll process these addresses in a batch, altogether,
    using RFC6724 section 6 rules:<br>
    <ul>
      <li>2001:0db8:58:c02::dead</li>
      <li>2001:0db8:c:a06::2:beef</li>
      <li>2001:0db8:44:204::c0de</li>
      <li>192.0.2.75</li>
      <li>203.0.113.38</li>
      <li>198.51.100.140</li>
    </ul>
    <br>
    ... and, most importantly, I won't mix the batches together.<br>
    <br>
    /a<br>
  </body>
</html>

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On 3/30/16 9:53 AM, Adam Roach wrote:
> [as individual]
>
> On 3/29/16 13:19, Dale R. Worley wrote:
>> As far as I can tell, the goal of this section is to make it clear that
>> if SRV records translate a SIP domain into (e.g.) two SRV records with
>> different priorities, and one SRV record's target name only has IPv4
>> addresses and the other SRV record's target name only has IPv6
>> addresses, then which family of addresses is attempted first is to be
>> determined by priorities of the SRV records.  This being the process
>> described in section 4.2 of this draft.
>
> I think it's somewhat more general than that, though. if you look up 
> an SRV record and get multiple A and multiple AAAA records, and those 
> records in turn contain multiple literal IP addresses, then you apply 
> the SRV priorities first and the RFC 6724 processes second.
Nit pick - Your sentence above starts "if you look up an SRV record"  
(that is _one_ SRV record).
There are no SRV priorities to apply (beyond the trivial "take the one I 
have").

Your example below of showing prioritizing multiple SRV records first 
using the SRV prioritization rules,
then applying the 6724 rules only when looking at a set of A/AAAA for a 
single SRV record makes sense.
>
> I think an example might help.
>
> Let's say I look up an SRV record (_sip._tcp.example.com) and get back 
> two entries:
>
> _sip._tcp.example.com.  300 IN  SRV  10 1 5060 sip-1.example.com.
> _sip._tcp.example.com.  300 IN  SRV  20 1 5060 sip-2.example.com.
>
>
> So then I grab the A and AAAA records for sip-1.example.com and 
> sip-2.example.com:
>
> sip-1.example.com.    300  IN  AAAA  2001:0db8:58:c02::face
> sip-1.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:cafe
> sip-1.example.com.    300  IN  AAAA  2001:0db8:44:204::d1ce
>
> sip-1.example.com.    1800  IN  A  192.0.2.45
> sip-1.example.com.    1800  IN  A  203.0.113.109
> sip-1.example.com.    1800  IN  A  198.51.100.24
>
> sip-2.example.com.    300  IN  AAAA  2001:0db8:58:c02::dead
> sip-2.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:beef
> sip-2.example.com.    300  IN  AAAA  2001:0db8:44:204::c0de
>
> sip-2.example.com.    1800  IN  A  192.0.2.75
> sip-2.example.com.    1800  IN  A  203.0.113.38
> sip-2.example.com.    1800  IN  A  198.51.100.140
>
>
> What I believe we're saying is that I will process these addresses in 
> a batch, altogether, using RFC6724 section 6 rules:
>
>   * 2001:0db8:58:c02::face
>   * 2001:0db8:c:a06::2:cafe
>   * 2001:0db8:44:204::d1ce
>   * 192.0.2.45
>   * 203.0.113.109
>   * 198.51.100.24
>
>
> And then (but only then, after I've exhausted that entire batch of 
> addresses), I'll process these addresses in a batch, altogether, using 
> RFC6724 section 6 rules:
>
>   * 2001:0db8:58:c02::dead
>   * 2001:0db8:c:a06::2:beef
>   * 2001:0db8:44:204::c0de
>   * 192.0.2.75
>   * 203.0.113.38
>   * 198.51.100.140
>
>
> ... and, most importantly, I won't mix the batches together.
>
> /a


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    <br>
    <br>
    <div class="moz-cite-prefix">On 3/30/16 9:53 AM, Adam Roach wrote:<br>
    </div>
    <blockquote cite="mid:56FBE85F.6080707@nostrum.com" type="cite">
      <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
      <div class="moz-cite-prefix">[as individual]<br>
        <br>
        On 3/29/16 13:19, Dale R. Worley wrote:<br>
      </div>
      <blockquote cite="mid:87poudw3iv.fsf@hobgoblin.ariadne.com"
        type="cite">
        <pre wrap="">As far as I can tell, the goal of this section is to make it clear that
if SRV records translate a SIP domain into (e.g.) two SRV records with
different priorities, and one SRV record's target name only has IPv4
addresses and the other SRV record's target name only has IPv6
addresses, then which family of addresses is attempted first is to be
determined by priorities of the SRV records.  This being the process
described in section 4.2 of this draft.</pre>
      </blockquote>
      <br>
      I think it's somewhat more general than that, though. if you look
      up an SRV record and get multiple A and multiple AAAA records, and
      those records in turn contain multiple literal IP addresses, then
      you apply the SRV priorities first and the RFC 6724 processes
      second.<br>
    </blockquote>
    Nit pick - Your sentence above starts "if you look up an SRV
    record"  (that is _one_ SRV record).<br>
    There are no SRV priorities to apply (beyond the trivial "take the
    one I have").<br>
    <br>
    Your example below of showing prioritizing multiple SRV records
    first using the SRV prioritization rules,<br>
    then applying the 6724 rules only when looking at a set of A/AAAA
    for a single SRV record makes sense.<br>
    <blockquote cite="mid:56FBE85F.6080707@nostrum.com" type="cite"> <br>
      I think an example might help.<br>
      <br>
      Let's say I look up an SRV record (_sip._tcp.example.com) and get
      back two entries:<br>
      <br>
      _sip._tcp.example.com.  300 IN  SRV  10 1 5060 sip-1.example.com.<br>
      _sip._tcp.example.com.  300 IN  SRV  20 1 5060 sip-2.example.com.<br>
      <br>
      <br>
      So then I grab the A and AAAA records for sip-1.example.com and
      sip-2.example.com:<br>
      <br>
      sip-1.example.com.    300  IN  AAAA  2001:0db8:58:c02::face<br>
      sip-1.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:cafe<br>
      sip-1.example.com.    300  IN  AAAA  2001:0db8:44:204::d1ce<br>
      <br>
      sip-1.example.com.    1800  IN  A  192.0.2.45<br>
      sip-1.example.com.    1800  IN  A  203.0.113.109<br>
      sip-1.example.com.    1800  IN  A  198.51.100.24<br>
      <br>
      sip-2.example.com.    300  IN  AAAA  2001:0db8:58:c02::dead<br>
      sip-2.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:beef<br>
      sip-2.example.com.    300  IN  AAAA  2001:0db8:44:204::c0de<br>
      <br>
      sip-2.example.com.    1800  IN  A  192.0.2.75<br>
      sip-2.example.com.    1800  IN  A  203.0.113.38<br>
      sip-2.example.com.    1800  IN  A  198.51.100.140<br>
      <br>
      <br>
      What I believe we're saying is that I will process these addresses
      in a batch, altogether, using RFC6724 section 6 rules:<br>
      <ul>
        <li>2001:0db8:58:c02::face</li>
        <li>2001:0db8:c:a06::2:cafe</li>
        <li>2001:0db8:44:204::d1ce</li>
        <li>192.0.2.45</li>
        <li>203.0.113.109</li>
        <li>198.51.100.24</li>
      </ul>
      <br>
      And then (but only then, after I've exhausted that entire batch of
      addresses), I'll process these addresses in a batch, altogether,
      using RFC6724 section 6 rules:<br>
      <ul>
        <li>2001:0db8:58:c02::dead</li>
        <li>2001:0db8:c:a06::2:beef</li>
        <li>2001:0db8:44:204::c0de</li>
        <li>192.0.2.75</li>
        <li>203.0.113.38</li>
        <li>198.51.100.140</li>
      </ul>
      <br>
      ... and, most importantly, I won't mix the batches together.<br>
      <br>
      /a<br>
    </blockquote>
    <br>
  </body>
</html>

--------------050601020001040302070007--


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> On 30 Mar 2016, at 16:53, Adam Roach <adam@nostrum.com> wrote:
>=20
> [as individual]
>=20
> On 3/29/16 13:19, Dale R. Worley wrote:
>> As far as I can tell, the goal of this section is to make it clear =
that
>> if SRV records translate a SIP domain into (e.g.) two SRV records =
with
>> different priorities, and one SRV record's target name only has IPv4
>> addresses and the other SRV record's target name only has IPv6
>> addresses, then which family of addresses is attempted first is to be
>> determined by priorities of the SRV records.  This being the process
>> described in section 4.2 of this draft.
>=20
> I think it's somewhat more general than that, though. if you look up =
an SRV record and get multiple A and multiple AAAA records, and those =
records in turn contain multiple literal IP addresses, then you apply =
the SRV priorities first and the RFC 6724 processes second.
>=20
> I think an example might help.
>=20
> Let's say I look up an SRV record (_sip._tcp.example.com) and get back =
two entries:
>=20
> _sip._tcp.example.com.  300 IN  SRV  10 1 5060 sip-1.example.com.
> _sip._tcp.example.com.  300 IN  SRV  20 1 5060 sip-2.example.com.
>=20
>=20
> So then I grab the A and AAAA records for sip-1.example.com and =
sip-2.example.com:
>=20
> sip-1.example.com.    300  IN  AAAA  2001:0db8:58:c02::face
> sip-1.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:cafe
> sip-1.example.com.    300  IN  AAAA  2001:0db8:44:204::d1ce
>=20
> sip-1.example.com.    1800  IN  A  192.0.2.45
> sip-1.example.com.    1800  IN  A  203.0.113.109
> sip-1.example.com.    1800  IN  A  198.51.100.24
>=20
> sip-2.example.com.    300  IN  AAAA  2001:0db8:58:c02::dead
> sip-2.example.com.    300  IN  AAAA  2001:0db8:c:a06::2:beef
> sip-2.example.com.    300  IN  AAAA  2001:0db8:44:204::c0de
>=20
> sip-2.example.com.    1800  IN  A  192.0.2.75
> sip-2.example.com.    1800  IN  A  203.0.113.38
> sip-2.example.com.    1800  IN  A  198.51.100.140
>=20
>=20
> What I believe we're saying is that I will process these addresses in =
a batch, altogether, using RFC6724 section 6 rules:
> 2001:0db8:58:c02::face
> 2001:0db8:c:a06::2:cafe
> 2001:0db8:44:204::d1ce
> 192.0.2.45
> 203.0.113.109
> 198.51.100.24
>=20
> And then (but only then, after I've exhausted that entire batch of =
addresses), I'll process these addresses in a batch, altogether, using =
RFC6724 section 6 rules:
> 2001:0db8:58:c02::dead
> 2001:0db8:c:a06::2:beef
> 2001:0db8:44:204::c0de
> 192.0.2.75
> 203.0.113.38
> 198.51.100.140
>=20
> ... and, most importantly, I won't mix the batches together.
I agree.

This example is something we need to carry with us to the actual Happy =
Eyeballs draft. How do we handle=20
connection setup with dual stacks with all six candidates - i.e. more =
than two. RFC 2782 is quite clear that=20
we need to test *ALL* SRV candidates for a selected name (after load =
balancing decision) before proceeding to another priority.
I need to check the Happy Eyeballs RFC unless Dan Wing reads this and =
can comment :-)

Cheers
/O


--Apple-Mail=_90EA9370-99FD-4F83-8B7B-9EB7DB8C059C
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<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><br class=""><div><blockquote type="cite" class=""><div class="">On 30 Mar 2016, at 16:53, Adam Roach &lt;<a href="mailto:adam@nostrum.com" class="">adam@nostrum.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><div class="">
  
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type" class="">
  
  <div bgcolor="#FFFFFF" text="#000000" class="">
    <div class="moz-cite-prefix">[as individual]<br class="">
      <br class="">
      On 3/29/16 13:19, Dale R. Worley wrote:<br class="">
    </div>
    <blockquote cite="mid:87poudw3iv.fsf@hobgoblin.ariadne.com" type="cite" class="">
      <pre wrap="" class="">As far as I can tell, the goal of this section is to make it clear that
if SRV records translate a SIP domain into (e.g.) two SRV records with
different priorities, and one SRV record's target name only has IPv4
addresses and the other SRV record's target name only has IPv6
addresses, then which family of addresses is attempted first is to be
determined by priorities of the SRV records.  This being the process
described in section 4.2 of this draft.</pre>
    </blockquote>
    <br class="">
    I think it's somewhat more general than that, though. if you look up
    an SRV record and get multiple A and multiple AAAA records, and
    those records in turn contain multiple literal IP addresses, then
    you apply the SRV priorities first and the RFC 6724 processes
    second.<br class="">
    <br class="">
    I think an example might help.<br class="">
    <br class="">
    Let's say I look up an SRV record (_sip._<a href="http://tcp.example.com" class="">tcp.example.com</a>) and get
    back two entries:<br class="">
    <br class="">
    _sip._<a href="http://tcp.example.com" class="">tcp.example.com</a>.&nbsp; 300 IN&nbsp; SRV&nbsp; 10 1 5060 <a href="http://sip-1.example.com" class="">sip-1.example.com</a>.<br class="">
    _sip._<a href="http://tcp.example.com" class="">tcp.example.com</a>.&nbsp; 300 IN&nbsp; SRV&nbsp; 20 1 5060 <a href="http://sip-2.example.com" class="">sip-2.example.com</a>.<br class="">
    <br class="">
    <br class="">
    So then I grab the A and AAAA records for <a href="http://sip-1.example.com" class="">sip-1.example.com</a> and
    <a href="http://sip-2.example.com" class="">sip-2.example.com</a>:<br class="">
    <br class="">
    <a href="http://sip-1.example.com" class="">sip-1.example.com</a>.&nbsp;&nbsp;&nbsp; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:58:c02::face<br class="">
    <a href="http://sip-1.example.com" class="">sip-1.example.com</a>.&nbsp;&nbsp;&nbsp; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:c:a06::2:cafe<br class="">
    <a href="http://sip-1.example.com" class="">sip-1.example.com</a>.&nbsp;&nbsp;&nbsp; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:44:204::d1ce<br class="">
    <br class="">
    <a href="http://sip-1.example.com" class="">sip-1.example.com</a>.&nbsp;&nbsp;&nbsp; 1800&nbsp; IN&nbsp; A&nbsp; 192.0.2.45<br class="">
    <a href="http://sip-1.example.com" class="">sip-1.example.com</a>.&nbsp;&nbsp;&nbsp; 1800&nbsp; IN&nbsp; A&nbsp; 203.0.113.109<br class="">
    <a href="http://sip-1.example.com" class="">sip-1.example.com</a>.&nbsp;&nbsp;&nbsp; 1800&nbsp; IN&nbsp; A&nbsp; 198.51.100.24<br class="">
    <br class="">
    <a href="http://sip-2.example.com" class="">sip-2.example.com</a>.&nbsp;&nbsp;&nbsp; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:58:c02::dead<br class="">
    <a href="http://sip-2.example.com" class="">sip-2.example.com</a>.&nbsp;&nbsp;&nbsp; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:c:a06::2:beef<br class="">
    <a href="http://sip-2.example.com" class="">sip-2.example.com</a>.&nbsp;&nbsp;&nbsp; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:44:204::c0de<br class="">
    <br class="">
    <a href="http://sip-2.example.com" class="">sip-2.example.com</a>.&nbsp;&nbsp;&nbsp; 1800&nbsp; IN&nbsp; A&nbsp; 192.0.2.75<br class="">
    <a href="http://sip-2.example.com" class="">sip-2.example.com</a>.&nbsp;&nbsp;&nbsp; 1800&nbsp; IN&nbsp; A&nbsp; 203.0.113.38<br class="">
    <a href="http://sip-2.example.com" class="">sip-2.example.com</a>.&nbsp;&nbsp;&nbsp; 1800&nbsp; IN&nbsp; A&nbsp; 198.51.100.140<br class="">
    <br class="">
    <br class="">
    What I believe we're saying is that I will process these addresses
    in a batch, altogether, using RFC6724 section 6 rules:<br class="">
    <ul class="">
      <li class="">2001:0db8:58:c02::face</li>
      <li class="">2001:0db8:c:a06::2:cafe</li>
      <li class="">2001:0db8:44:204::d1ce</li>
      <li class="">192.0.2.45</li>
      <li class="">203.0.113.109</li>
      <li class="">198.51.100.24</li>
    </ul>
    <br class="">
    And then (but only then, after I've exhausted that entire batch of
    addresses), I'll process these addresses in a batch, altogether,
    using RFC6724 section 6 rules:<br class="">
    <ul class="">
      <li class="">2001:0db8:58:c02::dead</li>
      <li class="">2001:0db8:c:a06::2:beef</li>
      <li class="">2001:0db8:44:204::c0de</li>
      <li class="">192.0.2.75</li>
      <li class="">203.0.113.38</li>
      <li class="">198.51.100.140</li>
    </ul>
    <br class="">
    ... and, most importantly, I won't mix the batches together.<br class=""></div></div></blockquote>I agree.</div><div><br class=""></div><div>This example is something we need to carry with us to the actual Happy Eyeballs draft. How do we handle&nbsp;</div><div>connection setup with dual stacks with all six candidates - i.e. more than two. RFC 2782 is quite clear that&nbsp;</div><div>we need to test *ALL* SRV candidates for a selected name (after load balancing decision) before proceeding to another priority.</div><div>I need to check the Happy Eyeballs RFC unless Dan Wing reads this and can comment :-)</div><div><br class=""></div><div>Cheers</div><div>/O</div><br class=""></body></html>
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: Robert Sparks <rjsparks@nostrum.com>
Thread-Topic: [sipcore] WGLC: draft-ietf-sip-dual-stack
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Subject: Re: [sipcore] WGLC: draft-ietf-sip-dual-stack
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On Mar 30, 2016, at 12:00 PM, Robert Sparks <rjsparks@nostrum.com<mailto:rj=
sparks@nostrum.com>> wrote:



On 3/30/16 9:53 AM, Adam Roach wrote:
[as individual]

On 3/29/16 13:19, Dale R. Worley wrote:

As far as I can tell, the goal of this section is to make it clear that
if SRV records translate a SIP domain into (e.g.) two SRV records with
different priorities, and one SRV record's target name only has IPv4
addresses and the other SRV record's target name only has IPv6
addresses, then which family of addresses is attempted first is to be
determined by priorities of the SRV records.  This being the process
described in section 4.2 of this draft.

I think it's somewhat more general than that, though. if you look up an SRV=
 record and get multiple A and multiple AAAA records, and those records in =
turn contain multiple literal IP addresses, then you apply the SRV prioriti=
es first and the RFC 6724 processes second.
Nit pick - Your sentence above starts "if you look up an SRV record"  (that=
 is _one_ SRV record).
There are no SRV priorities to apply (beyond the trivial "take the one I ha=
ve").

Your example below of showing prioritizing multiple SRV records first using=
 the SRV prioritization rules,
then applying the 6724 rules only when looking at a set of A/AAAA for a sin=
gle SRV record makes sense.

I agree and urging it might be tremendously useful to use such an example i=
n the draft to unambiguously drive the point home.

-G


I think an example might help.

Let's say I look up an SRV record (_sip._tcp.example.com<http://tcp.example=
.com>) and get back two entries:

_sip._tcp.example.com<http://tcp.example.com>.  300 IN  SRV  10 1 5060 sip-=
1.example.com<http://sip-1.example.com>.
_sip._tcp.example.com<http://tcp.example.com>.  300 IN  SRV  20 1 5060 sip-=
2.example.com<http://sip-2.example.com>.


So then I grab the A and AAAA records for sip-1.example.com<http://sip-1.ex=
ample.com> and sip-2.example.com<http://sip-2.example.com>:

sip-1.example.com<http://sip-1.example.com>.    300  IN  AAAA  2001:0db8:58=
:c02::face
sip-1.example.com<http://sip-1.example.com>.    300  IN  AAAA  2001:0db8:c:=
a06::2:cafe
sip-1.example.com<http://sip-1.example.com>.    300  IN  AAAA  2001:0db8:44=
:204::d1ce

sip-1.example.com<http://sip-1.example.com>.    1800  IN  A  192.0.2.45
sip-1.example.com<http://sip-1.example.com>.    1800  IN  A  203.0.113.109
sip-1.example.com<http://sip-1.example.com>.    1800  IN  A  198.51.100.24

sip-2.example.com<http://sip-2.example.com>.    300  IN  AAAA  2001:0db8:58=
:c02::dead
sip-2.example.com<http://sip-2.example.com>.    300  IN  AAAA  2001:0db8:c:=
a06::2:beef
sip-2.example.com<http://sip-2.example.com>.    300  IN  AAAA  2001:0db8:44=
:204::c0de

sip-2.example.com<http://sip-2.example.com>.    1800  IN  A  192.0.2.75
sip-2.example.com<http://sip-2.example.com>.    1800  IN  A  203.0.113.38
sip-2.example.com<http://sip-2.example.com>.    1800  IN  A  198.51.100.140


What I believe we're saying is that I will process these addresses in a bat=
ch, altogether, using RFC6724 section 6 rules:

  *   2001:0db8:58:c02::face
  *   2001:0db8:c:a06::2:cafe
  *   2001:0db8:44:204::d1ce
  *   192.0.2.45
  *   203.0.113.109
  *   198.51.100.24

And then (but only then, after I've exhausted that entire batch of addresse=
s), I'll process these addresses in a batch, altogether, using RFC6724 sect=
ion 6 rules:

  *   2001:0db8:58:c02::dead
  *   2001:0db8:c:a06::2:beef
  *   2001:0db8:44:204::c0de
  *   192.0.2.75
  *   203.0.113.38
  *   198.51.100.140

... and, most importantly, I won't mix the batches together.

/a

_______________________________________________
sipcore mailing list
sipcore@ietf.org<mailto:sipcore@ietf.org>
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div><br>
</div>
<div><br>
On Mar 30, 2016, at 12:00 PM, Robert Sparks &lt;<a href=3D"mailto:rjsparks@=
nostrum.com">rjsparks@nostrum.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div><br>
<br>
<div class=3D"moz-cite-prefix">On 3/30/16 9:53 AM, Adam Roach wrote:<br>
</div>
<blockquote cite=3D"mid:56FBE85F.6080707@nostrum.com" type=3D"cite">
<div class=3D"moz-cite-prefix">[as individual]<br>
<br>
On 3/29/16 13:19, Dale R. Worley wrote:<br>
</div>
<blockquote cite=3D"mid:87poudw3iv.fsf@hobgoblin.ariadne.com" type=3D"cite"=
>
<pre wrap=3D"">As far as I can tell, the goal of this section is to make it=
 clear that
if SRV records translate a SIP domain into (e.g.) two SRV records with
different priorities, and one SRV record's target name only has IPv4
addresses and the other SRV record's target name only has IPv6
addresses, then which family of addresses is attempted first is to be
determined by priorities of the SRV records.  This being the process
described in section 4.2 of this draft.</pre>
</blockquote>
<br>
I think it's somewhat more general than that, though. if you look up an SRV=
 record and get multiple A and multiple AAAA records, and those records in =
turn contain multiple literal IP addresses, then you apply the SRV prioriti=
es first and the RFC 6724 processes
 second.<br>
</blockquote>
Nit pick - Your sentence above starts &quot;if you look up an SRV record&qu=
ot;&nbsp; (that is _one_ SRV record).<br>
There are no SRV priorities to apply (beyond the trivial &quot;take the one=
 I have&quot;).<br>
<br>
Your example below of showing prioritizing multiple SRV records first using=
 the SRV prioritization rules,<br>
then applying the 6724 rules only when looking at a set of A/AAAA for a sin=
gle SRV record makes sense.<br>
</div>
</blockquote>
<div><br>
</div>
I agree and urging it might be tremendously useful to use such an example i=
n the draft to unambiguously drive the point home.
<div><br>
</div>
<div>-G</div>
<div><br>
<blockquote type=3D"cite">
<div>
<blockquote cite=3D"mid:56FBE85F.6080707@nostrum.com" type=3D"cite"><br>
I think an example might help.<br>
<br>
Let's say I look up an SRV record (_sip._<a href=3D"http://tcp.example.com"=
>tcp.example.com</a>) and get back two entries:<br>
<br>
_sip._<a href=3D"http://tcp.example.com">tcp.example.com</a>.&nbsp; 300 IN&=
nbsp; SRV&nbsp; 10 1 5060
<a href=3D"http://sip-1.example.com">sip-1.example.com</a>.<br>
_sip._<a href=3D"http://tcp.example.com">tcp.example.com</a>.&nbsp; 300 IN&=
nbsp; SRV&nbsp; 20 1 5060
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>.<br>
<br>
<br>
So then I grab the A and AAAA records for <a href=3D"http://sip-1.example.c=
om">sip-1.example.com</a> and
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>:<br>
<br>
<a href=3D"http://sip-1.example.com">sip-1.example.com</a>.&nbsp;&nbsp;&nbs=
p; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:58:c02::face<br>
<a href=3D"http://sip-1.example.com">sip-1.example.com</a>.&nbsp;&nbsp;&nbs=
p; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:c:a06::2:cafe<br>
<a href=3D"http://sip-1.example.com">sip-1.example.com</a>.&nbsp;&nbsp;&nbs=
p; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:44:204::d1ce<br>
<br>
<a href=3D"http://sip-1.example.com">sip-1.example.com</a>.&nbsp;&nbsp;&nbs=
p; 1800&nbsp; IN&nbsp; A&nbsp; 192.0.2.45<br>
<a href=3D"http://sip-1.example.com">sip-1.example.com</a>.&nbsp;&nbsp;&nbs=
p; 1800&nbsp; IN&nbsp; A&nbsp; 203.0.113.109<br>
<a href=3D"http://sip-1.example.com">sip-1.example.com</a>.&nbsp;&nbsp;&nbs=
p; 1800&nbsp; IN&nbsp; A&nbsp; 198.51.100.24<br>
<br>
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>.&nbsp;&nbsp;&nbs=
p; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:58:c02::dead<br>
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>.&nbsp;&nbsp;&nbs=
p; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:c:a06::2:beef<br>
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>.&nbsp;&nbsp;&nbs=
p; 300&nbsp; IN&nbsp; AAAA&nbsp; 2001:0db8:44:204::c0de<br>
<br>
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>.&nbsp;&nbsp;&nbs=
p; 1800&nbsp; IN&nbsp; A&nbsp; 192.0.2.75<br>
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>.&nbsp;&nbsp;&nbs=
p; 1800&nbsp; IN&nbsp; A&nbsp; 203.0.113.38<br>
<a href=3D"http://sip-2.example.com">sip-2.example.com</a>.&nbsp;&nbsp;&nbs=
p; 1800&nbsp; IN&nbsp; A&nbsp; 198.51.100.140<br>
<br>
<br>
What I believe we're saying is that I will process these addresses in a bat=
ch, altogether, using RFC6724 section 6 rules:<br>
<ul>
<li>2001:0db8:58:c02::face </li><li>2001:0db8:c:a06::2:cafe </li><li>2001:0=
db8:44:204::d1ce </li><li>192.0.2.45 </li><li>203.0.113.109 </li><li>198.51=
.100.24 </li></ul>
<br>
And then (but only then, after I've exhausted that entire batch of addresse=
s), I'll process these addresses in a batch, altogether, using RFC6724 sect=
ion 6 rules:<br>
<ul>
<li>2001:0db8:58:c02::dead </li><li>2001:0db8:c:a06::2:beef </li><li>2001:0=
db8:44:204::c0de </li><li>192.0.2.75 </li><li>203.0.113.38 </li><li>198.51.=
100.140 </li></ul>
<br>
... and, most importantly, I won't mix the batches together.<br>
<br>
/a<br>
</blockquote>
<br>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>sipcore mailing list</span><br>
<span><a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www=
.ietf.org/mailman/listinfo/sipcore</a></span><br>
</div>
</blockquote>
</div>
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Subject: Re: [sipcore] Outline of a SIP-DTLS document
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It looks like we have enough interest for it to be worthwhile.  I can't
work on this for a couple of weeks because the dual-stack work needs to
get done but after that we can get a draft started.

Thanks,

Dale


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Subject: [sipcore] Fwd: [Technical Errata Reported] RFC5360 (4650)
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What do people think of this errata report?

Thanks!

Ben.

Forwarded message:

> From: RFC Errata System <rfc-editor@rfc-editor.org>
> To: jdrosen@cisco.com, Gonzalo.Camarillo@ericsson.com, 
> dean.willis@softarmor.com, ben@nostrum.com, alissa@cooperw.in, 
> dean.willis@softarmor.com, drage@alcatel-lucent.com
> Cc: brett@broadsoft.com, rfc-editor@rfc-editor.org
> Subject: [Technical Errata Reported] RFC5360 (4650)
> Date: Tue, 29 Mar 2016 08:50:09 -0700 (PDT)
>
> The following errata report has been submitted for RFC5360,
> "A Framework for Consent-Based Communications in the Session 
> Initiation Protocol (SIP)".
>
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=5360&eid=4650
>
> --------------------------------------
> Type: Technical
> Reported by: Brett Tate <brett@broadsoft.com>
>
> Section: 5.9.3
>
> Original Text
> -------------
> The following is an example of a Permission-Missing header field:
>
> Corrected Text
> --------------
> If the URI contains a comma, question mark or semicolon, the
> URI MUST be enclosed in angle brackets (< and >).
>
> The following is an example of a Permission-Missing header field:
>
> Notes
> -----
> Comma and semicolon can cause decode ambiguities when the header 
> contains addr-spec values instead of name-addr values.  For 
> consistency with RFC 3261 section 20, the same bracket rule is 
> indicated to resolve the ambiguity.
>
> Instructions:
> -------------
> This erratum is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
>
> --------------------------------------
> RFC5360 (draft-ietf-sip-consent-framework-04)
> --------------------------------------
> Title               : A Framework for Consent-Based Communications in 
> the Session Initiation Protocol (SIP)
> Publication Date    : October 2008
> Author(s)           : J. Rosenberg, G. Camarillo, Ed., D. Willis
> Category            : PROPOSED STANDARD
> Source              : Session Initiation Protocol
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
>


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To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 30 Mar 2016 15:04:17 -0500
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Subject: Re: [sipcore] [Technical Errata Reported] RFC5360 (4650)
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Please include the sipcore list (copied) for this discussion.

Thanks!

Ben.

On 30 Mar 2016, at 12:19, Gonzalo Camarillo wrote:

> Hi,
>
> I like more the name-addr text but can live with the other alternative
> as well. I will let it to the ADs to decide how to resolve all these 
> errata.
>
> Cheers,
>
> Gonzalo
>
> On 30/03/2016 8:17 PM, Brett Tate wrote:
>> Hi,
>>
>> The errata currently reflects a sentence from RFC 3261 section 20 
>> which
>> means the same thing as the other RFC 3261 snippet.
>>
>> Since the mandates are equivalent, duplicating either sentence is 
>> okay with
>> me.
>>
>> If it matters, the related RFC 3325 errata from 2013 is based upon 
>> the RFC
>> 3261 section 20 sentence.  I've recently opened similar errata for 
>> RFC 3515,
>> RFC 5002, RFC 5318, RFC 5360, and RFC 5502.
>>
>> Thanks,
>> Brett
>>
>>> -----Original Message-----
>>> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>>> Sent: Wednesday, March 30, 2016 12:10 PM
>>> To: Dean Willis; RFC Errata System
>>> Cc: Keith Drage; brett@broadsoft.com; alissa@cooperw.in; 
>>> ben@nostrum.com;
>>> jdrosen@cisco.com
>>> Subject: Re: [Technical Errata Reported] RFC5360 (4650)
>>>
>>> Hi,
>>>
>>> yes, in RFC 3261 the following text is used instead:
>>>
>>>>    Even if the "display-
>>>>    name" is empty, the "name-addr" form MUST be used if the 
>>>> "addr-spec"
>>>>    contains a comma, question mark, or semicolon.
>>>
>>> So, instead of talking about angle brackets, we probably want to 
>>> talk
>>> about
>>> the "name-addr" form being used in those cases.
>>>
>>> Cheers,
>>>
>>> Gonzalo
>>>
>>> On 29/03/2016 8:51 PM, Dean Willis wrote:
>>>> Seems reasonable to me.
>>>>
>>>> On Mar 29, 2016 10:50 AM, "RFC Errata System"
>>>> <rfc-editor@rfc-editor.org <mailto:rfc-editor@rfc-editor.org>> 
>>>> wrote:
>>>>
>>>>     The following errata report has been submitted for RFC5360,
>>>>     "A Framework for Consent-Based Communications in the Session
>>>>     Initiation Protocol (SIP)".
>>>>
>>>>     --------------------------------------
>>>>     You may review the report below and at:
>>>>     http://www.rfc-editor.org/errata_search.php?rfc=5360&eid=4650
>>>>
>>>>     --------------------------------------
>>>>     Type: Technical
>>>>     Reported by: Brett Tate <brett@broadsoft.com
>>>>     <mailto:brett@broadsoft.com>>
>>>>
>>>>     Section: 5.9.3
>>>>
>>>>     Original Text
>>>>     -------------
>>>>     The following is an example of a Permission-Missing header 
>>>> field:
>>>>
>>>>     Corrected Text
>>>>     --------------
>>>>     If the URI contains a comma, question mark or semicolon, the
>>>>     URI MUST be enclosed in angle brackets (< and >).
>>>>
>>>>     The following is an example of a Permission-Missing header 
>>>> field:
>>>>
>>>>     Notes
>>>>     -----
>>>>     Comma and semicolon can cause decode ambiguities when the 
>>>> header
>>>>     contains addr-spec values instead of name-addr values.  For
>>>>     consistency with RFC 3261 section 20, the same bracket rule is
>>>>     indicated to resolve the ambiguity.
>>>>
>>>>     Instructions:
>>>>     -------------
>>>>     This erratum is currently posted as "Reported". If necessary, 
>>>> please
>>>>     use "Reply All" to discuss whether it should be verified or
>>>>     rejected. When a decision is reached, the verifying party 
>>>> (IESG)
>>>>     can log in to change the status and edit the report, if 
>>>> necessary.
>>>>
>>>>     --------------------------------------
>>>>     RFC5360 (draft-ietf-sip-consent-framework-04)
>>>>     --------------------------------------
>>>>     Title               : A Framework for Consent-Based 
>>>> Communications
>>>>     in the Session Initiation Protocol (SIP)
>>>>     Publication Date    : October 2008
>>>>     Author(s)           : J. Rosenberg, G. Camarillo, Ed., D. 
>>>> Willis
>>>>     Category            : PROPOSED STANDARD
>>>>     Source              : Session Initiation Protocol
>>>>     Area                : Real-time Applications and Infrastructure
>>>>     Stream              : IETF
>>>>     Verifying Party     : IESG
>>>>


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Subject: Re: [sipcore] [Technical Errata Reported] RFC3515 (4652)
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(copying sipcore)

On 30 Mar 2016, at 12:12, Brett Tate wrote:

> Hi,
>
> My proposed resolution is similar to what occurred within RFC 3325 errata
> (3744 and 3894).
>
>> -----Original Message-----
>> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
>> Sent: Wednesday, March 30, 2016 1:09 PM
>> To: rsparks@dynamicsoft.com; ben@nostrum.com; alissa@cooperw.in;
>> dean.willis@softarmor.com; drage@alcatel-lucent.com
>> Cc: brett@broadsoft.com; rfc-editor@rfc-editor.org
>> Subject: [Technical Errata Reported] RFC3515 (4652)
>>
>> The following errata report has been submitted for RFC3515, "The Session
>> Initiation Protocol (SIP) Refer Method".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=3515&eid=4652
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Brett Tate <brett@broadsoft.com>
>>
>> Section: 2.1
>>
>> Original Text
>> -------------
>> The Refer-To header field MAY be encrypted as part of end-to-end
> encryption.
>>
>> Corrected Text
>> --------------
>> If the URI contains a comma, question mark or semicolon, the URI MUST be
>> enclosed in angle brackets (< and >).
>>
>> The Refer-To header field MAY be encrypted as part of end-to-end
> encryption.
>>
>> Notes
>> -----
>> If addr-spec is used when there are parameters, it is ambiguous if the
>> parameters are URI parameters or header parameters.  For consistency
> with RFC
>> 3261 section 20, the same bracket rule is indicated even if comma and
>> question mark do not cause an issue.
>>
>> Instructions:
>> -------------
>> This erratum is currently posted as "Reported". If necessary, please use
>> "Reply All" to discuss whether it should be verified or rejected. When a
>> decision is reached, the verifying party (IESG) can log in to change the
>> status and edit the report, if necessary.
>>
>> --------------------------------------
>> RFC3515 (draft-ietf-sip-refer-07)
>> --------------------------------------
>> Title               : The Session Initiation Protocol (SIP) Refer Method
>> Publication Date    : April 2003
>> Author(s)           : R. Sparks
>> Category            : PROPOSED STANDARD
>> Source              : Session Initiation Protocol
>> Area                : Real-time Applications and Infrastructure
>> Stream              : IETF
>> Verifying Party     : IESG


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Subject: Re: [sipcore] [Technical Errata Reported] RFC5318 (4651)
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(copying sipcore)

On 30 Mar 2016, at 11:12, Gonzalo Camarillo wrote:

> Hi Brett,
>
> I have the same comment as on the other document. I would talk about the
> name-addr form instead. Let's agree on the text we want to use for the
> three documents, since it is the same issue.
>
> Cheers,
>
> Gonzalo
>
> On 29/03/2016 7:06 PM, Brett Tate wrote:
>> Hi,
>>
>> My proposed resolution is similar to what occurred within RFC 3325 errata
>> (3744 and 3894).
>>
>>> -----Original Message-----
>>> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
>>> Sent: Tuesday, March 29, 2016 12:02 PM
>>> To: Jani.Hautakorpi@ericsson.com; Gonzalo.Camarillo@ericsson.com;
>>> iesg@ietf.org
>>> Cc: brett@broadsoft.com; rfc-editor@rfc-editor.org
>>> Subject: [Technical Errata Reported] RFC5318 (4651)
>>>
>>> The following errata report has been submitted for RFC5318, "The Session
>>> Initiation Protocol (SIP) P-Refused-URI-List Private-Header (P-Header)".
>>>
>>> --------------------------------------
>>> You may review the report below and at:
>>> http://www.rfc-editor.org/errata_search.php?rfc=5318&eid=4651
>>>
>>> --------------------------------------
>>> Type: Technical
>>> Reported by: Brett Tate <brett@broadsoft.com>
>>>
>>> Section: 5
>>>
>>> Original Text
>>> -------------
>>> The members P-Header parameter MUST contain a cid-url, which is defined
>> in
>>> RFC 2392 [2].
>>>
>>> Corrected Text
>>> --------------
>>> The members P-Header parameter MUST contain a cid-url, which is defined
>> in
>>> RFC 2392 [2].
>>>
>>> If the URI contains a comma, question mark or semicolon, the URI MUST be
>>> enclosed in angle brackets (< and >).
>>>
>>> Notes
>>> -----
>>> Comma and semicolon can cause decode ambiguities when the header
>> contains
>>> addr-spec values instead of name-addr values.  For consistency with RFC
>> 3261
>>> section 20, the same bracket rule is indicated to resolve the ambiguity.
>>>
>>> Instructions:
>>> -------------
>>> This erratum is currently posted as "Reported". If necessary, please use
>>> "Reply All" to discuss whether it should be verified or rejected. When a
>>> decision is reached, the verifying party (IESG) can log in to change the
>>> status and edit the report, if necessary.
>>>
>>> --------------------------------------
>>> RFC5318 (draft-hautakorpi-sipping-uri-list-handling-refused-03)
>>> --------------------------------------
>>> Title               : The Session Initiation Protocol (SIP)
>> P-Refused-URI-
>>> List Private-Header (P-Header)
>>> Publication Date    : December 2008
>>> Author(s)           : J. Hautakorpi, G. Camarillo
>>> Category            : INFORMATIONAL
>>> Source              : IETF - NON WORKING GROUP
>>> Area                : N/A
>>> Stream              : IETF
>>> Verifying Party     : IESG


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From: Robert Sparks <rjsparks@nostrum.com>
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Date: Wed, 30 Mar 2016 16:40:01 -0500
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Cc: Ben Campbell <ben@nostrum.com>
Subject: [sipcore] name-addr vs addr-spec requirements (was Re: [Technical Errata Reported] RFC5318 (4651))
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This cluster of errata (4650, 4651, 4652) are on the wrong edge of 
making a technical change from my perspective.

It's a good technical change to make, and it _should_ have been stated 
as such in the documents the errata are filed against, but it is still a 
technical change.

I would prefer the work to clarify the documents be done with a draft 
that goes through IETF LC rather than use errata.

That draft could solve the problem this cluster points to much more 
generally, updating 3261 to require that _anywhere_ name-addr/addr-spec 
are used as alternates, name-addr MUST be used if comma, question mark, 
or semicolon appear. The text could leave an "unless explicitly 
specified otherwise" out.
That draft could have a section capturing the places where we failed to 
say that so far, and update the relevant RFCs (the ones these errata 
point to) as well.

RjS

On 3/30/16 3:06 PM, Ben Campbell wrote:
> (copying sipcore)
>
> On 30 Mar 2016, at 11:12, Gonzalo Camarillo wrote:
>
>> Hi Brett,
>>
>> I have the same comment as on the other document. I would talk about the
>> name-addr form instead. Let's agree on the text we want to use for the
>> three documents, since it is the same issue.
>>
>> Cheers,
>>
>> Gonzalo
>>
>> On 29/03/2016 7:06 PM, Brett Tate wrote:
>>> Hi,
>>>
>>> My proposed resolution is similar to what occurred within RFC 3325 errata
>>> (3744 and 3894).
>>>
>>>> -----Original Message-----
>>>> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
>>>> Sent: Tuesday, March 29, 2016 12:02 PM
>>>> To: Jani.Hautakorpi@ericsson.com; Gonzalo.Camarillo@ericsson.com;
>>>> iesg@ietf.org
>>>> Cc: brett@broadsoft.com; rfc-editor@rfc-editor.org
>>>> Subject: [Technical Errata Reported] RFC5318 (4651)
>>>>
>>>> The following errata report has been submitted for RFC5318, "The Session
>>>> Initiation Protocol (SIP) P-Refused-URI-List Private-Header (P-Header)".
>>>>
>>>> --------------------------------------
>>>> You may review the report below and at:
>>>> http://www.rfc-editor.org/errata_search.php?rfc=5318&eid=4651
>>>>
>>>> --------------------------------------
>>>> Type: Technical
>>>> Reported by: Brett Tate <brett@broadsoft.com>
>>>>
>>>> Section: 5
>>>>
>>>> Original Text
>>>> -------------
>>>> The members P-Header parameter MUST contain a cid-url, which is defined
>>> in
>>>> RFC 2392 [2].
>>>>
>>>> Corrected Text
>>>> --------------
>>>> The members P-Header parameter MUST contain a cid-url, which is defined
>>> in
>>>> RFC 2392 [2].
>>>>
>>>> If the URI contains a comma, question mark or semicolon, the URI MUST be
>>>> enclosed in angle brackets (< and >).
>>>>
>>>> Notes
>>>> -----
>>>> Comma and semicolon can cause decode ambiguities when the header
>>> contains
>>>> addr-spec values instead of name-addr values.  For consistency with RFC
>>> 3261
>>>> section 20, the same bracket rule is indicated to resolve the ambiguity.
>>>>
>>>> Instructions:
>>>> -------------
>>>> This erratum is currently posted as "Reported". If necessary, please use
>>>> "Reply All" to discuss whether it should be verified or rejected. When a
>>>> decision is reached, the verifying party (IESG) can log in to change the
>>>> status and edit the report, if necessary.
>>>>
>>>> --------------------------------------
>>>> RFC5318 (draft-hautakorpi-sipping-uri-list-handling-refused-03)
>>>> --------------------------------------
>>>> Title               : The Session Initiation Protocol (SIP)
>>> P-Refused-URI-
>>>> List Private-Header (P-Header)
>>>> Publication Date    : December 2008
>>>> Author(s)           : J. Hautakorpi, G. Camarillo
>>>> Category            : INFORMATIONAL
>>>> Source              : IETF - NON WORKING GROUP
>>>> Area                : N/A
>>>> Stream              : IETF
>>>> Verifying Party     : IESG
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] [Technical Errata Reported] RFC5002 (4649)
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Copying sipcore since similar errata being discussed.

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: Tuesday, March 29, 2016 10:59 AM
> To: 'RFC Errata System'; 'Gonzalo.Camarillo@ericsson.com';
> 'German.Blanco@ericsson.com'; 'iesg@ietf.org'
> Subject: RE: [Technical Errata Reported] RFC5002 (4649)
>
> Hi,
>
> My proposed resolution is similar to what occurred within RFC 3325
errata
> (3744 and 3894).
>
> > -----Original Message-----
> > From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
> > Sent: Tuesday, March 29, 2016 10:55 AM
> > To: Gonzalo.Camarillo@ericsson.com; German.Blanco@ericsson.com;
> > iesg@ietf.org
> > Cc: brett@broadsoft.com; rfc-editor@rfc-editor.org
> > Subject: [Technical Errata Reported] RFC5002 (4649)
> >
> > The following errata report has been submitted for RFC5002, "The
> > Session Initiation Protocol (SIP) P-Profile-Key Private Header
(P-Header)".
> >
> > --------------------------------------
> > You may review the report below and at:
> > http://www.rfc-editor.org/errata_search.php?rfc=5002&eid=4649
> >
> > --------------------------------------
> > Type: Technical
> > Reported by: Brett Tate <brett@broadsoft.com>
> >
> > Section: 5
> >
> > Original Text
> > -------------
> > The following is an example of a P-Profile-Key header field that
> > contains a Wildcarded Public Service Identity:
> >
> > Corrected Text
> > --------------
> > If the URI contains a comma, question mark or semicolon, the URI MUST
> > be enclosed in angle brackets (< and >).
> >
> > The following is an example of a P-Profile-Key header field that
> > contains a Wildcarded Public Service Identity:
> >
> > Notes
> > -----
> > If addr-spec is used when there are parameters, it is ambiguous if the
> > parameters are URI parameters or header parameters.  For consistency
> > with RFC
> > 3261 section 20, the same bracket rule is indicated even if comma and
> > question mark do not cause an issue.
> >
> > Instructions:
> > -------------
> > This erratum is currently posted as "Reported". If necessary, please
> > use "Reply All" to discuss whether it should be verified or rejected.
> > When a decision is reached, the verifying party (IESG) can log in to
> > change the status and edit the report, if necessary.
> >
> > --------------------------------------
> > RFC5002 (draft-camarillo-sipping-profile-key-02)
> > --------------------------------------
> > Title               : The Session Initiation Protocol (SIP)
P-Profile-Key
> > Private Header (P-Header)
> > Publication Date    : August 2007
> > Author(s)           : G. Camarillo, G. Blanco
> > Category            : INFORMATIONAL
> > Source              : IETF - NON WORKING GROUP
> > Area                : N/A
> > Stream              : IETF
> > Verifying Party     : IESG


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Subject: Re: [sipcore] [Technical Errata Reported] RFC5502 (4648)
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Copying sipcore since similar errata being discussed.

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: Tuesday, March 29, 2016 10:23 AM
> To: 'RFC Errata System'; 'HansErik.van.Elburg@ericsson.com';
'iesg@ietf.org'
> Subject: RE: [Technical Errata Reported] RFC5502 (4648)
>
> Hi,
>
> My proposed resolution is similar to what occurred within RFC 3325
errata
> (3744 and 3894).
>
> > -----Original Message-----
> > From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
> > Sent: Tuesday, March 29, 2016 9:54 AM
> > To: HansErik.van.Elburg@ericsson.com; iesg@ietf.org
> > Cc: brett@broadsoft.com; rfc-editor@rfc-editor.org
> > Subject: [Technical Errata Reported] RFC5502 (4648)
> >
> > The following errata report has been submitted for RFC5502, "The SIP
> > P- Served-User Private-Header (P-Header) for the 3GPP IP Multimedia
> > (IM) Core Network (CN) Subsystem".
> >
> > --------------------------------------
> > You may review the report below and at:
> > http://www.rfc-editor.org/errata_search.php?rfc=5502&eid=4648
> >
> > --------------------------------------
> > Type: Technical
> > Reported by: Brett Tate <brett@broadsoft.com>
> >
> > Section: 6
> >
> > Original Text
> > -------------
> > EQUAL, HCOLON, SEMI, name-addr, addr-spec, and generic-param are
> > defined in RFC 3261 [2].
> >
> >
> > Corrected Text
> > --------------
> > EQUAL, HCOLON, SEMI, name-addr, addr-spec, and generic-param are
> > defined in RFC 3261 [2].
> >
> > If the URI contains a comma, question mark or semicolon, the URI MUST
> > be enclosed in angle brackets (< and >).
> >
> > Notes
> > -----
> > If addr-spec is used when there are parameters, it is ambiguous if the
> > parameters are URI parameters or served-user-param.  For consistency
> > with RFC
> > 3261 section 20, the same bracket rule is indicated even if comma and
> > question mark do not cause an issue.
> >
> > Instructions:
> > -------------
> > This erratum is currently posted as "Reported". If necessary, please
> > use "Reply All" to discuss whether it should be verified or rejected.
> > When a decision is reached, the verifying party (IESG) can log in to
> > change the status and edit the report, if necessary.
> >
> > --------------------------------------
> > RFC5502 (draft-vanelburg-sipping-served-user-08)
> > --------------------------------------
> > Title               : The SIP P-Served-User Private-Header (P-Header)
for
> the
> > 3GPP IP Multimedia (IM) Core Network (CN) Subsystem
> > Publication Date    : April 2009
> > Author(s)           : J. van Elburg
> > Category            : INFORMATIONAL
> > Source              : IETF - NON WORKING GROUP
> > Area                : N/A
> > Stream              : IETF
> > Verifying Party     : IESG


From nobody Thu Mar 31 07:07:37 2016
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To: sipcore@ietf.org
References: <20160329135428.51F55180004@rfc-editor.org> <8ebfaf07d50da3c1dafb4083e5d2862b@mail.gmail.com>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
Message-ID: <56FD2F21.2070609@alum.mit.edu>
Date: Thu, 31 Mar 2016 10:07:29 -0400
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Subject: Re: [sipcore] [Technical Errata Reported] RFCs 3515, 5502, 5360, ...
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ISTM that it would be good to discuss all of these together, and perhaps 
solicit pointers to any others that need a similar fix.

	Thanks,
	Paul

On 3/31/16 7:36 AM, Brett Tate wrote:
> Copying sipcore since similar errata being discussed.
>
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com]
>> Sent: Tuesday, March 29, 2016 10:23 AM
>> To: 'RFC Errata System'; 'HansErik.van.Elburg@ericsson.com';
> 'iesg@ietf.org'
>> Subject: RE: [Technical Errata Reported] RFC5502 (4648)
>>
>> Hi,
>>
>> My proposed resolution is similar to what occurred within RFC 3325
> errata
>> (3744 and 3894).
>>
>>> -----Original Message-----
>>> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
>>> Sent: Tuesday, March 29, 2016 9:54 AM
>>> To: HansErik.van.Elburg@ericsson.com; iesg@ietf.org
>>> Cc: brett@broadsoft.com; rfc-editor@rfc-editor.org
>>> Subject: [Technical Errata Reported] RFC5502 (4648)
>>>
>>> The following errata report has been submitted for RFC5502, "The SIP
>>> P- Served-User Private-Header (P-Header) for the 3GPP IP Multimedia
>>> (IM) Core Network (CN) Subsystem".
>>>
>>> --------------------------------------
>>> You may review the report below and at:
>>> http://www.rfc-editor.org/errata_search.php?rfc=5502&eid=4648
>>>
>>> --------------------------------------
>>> Type: Technical
>>> Reported by: Brett Tate <brett@broadsoft.com>
>>>
>>> Section: 6
>>>
>>> Original Text
>>> -------------
>>> EQUAL, HCOLON, SEMI, name-addr, addr-spec, and generic-param are
>>> defined in RFC 3261 [2].
>>>
>>>
>>> Corrected Text
>>> --------------
>>> EQUAL, HCOLON, SEMI, name-addr, addr-spec, and generic-param are
>>> defined in RFC 3261 [2].
>>>
>>> If the URI contains a comma, question mark or semicolon, the URI MUST
>>> be enclosed in angle brackets (< and >).
>>>
>>> Notes
>>> -----
>>> If addr-spec is used when there are parameters, it is ambiguous if the
>>> parameters are URI parameters or served-user-param.  For consistency
>>> with RFC
>>> 3261 section 20, the same bracket rule is indicated even if comma and
>>> question mark do not cause an issue.
>>>
>>> Instructions:
>>> -------------
>>> This erratum is currently posted as "Reported". If necessary, please
>>> use "Reply All" to discuss whether it should be verified or rejected.
>>> When a decision is reached, the verifying party (IESG) can log in to
>>> change the status and edit the report, if necessary.
>>>
>>> --------------------------------------
>>> RFC5502 (draft-vanelburg-sipping-served-user-08)
>>> --------------------------------------
>>> Title               : The SIP P-Served-User Private-Header (P-Header)
> for
>> the
>>> 3GPP IP Multimedia (IM) Core Network (CN) Subsystem
>>> Publication Date    : April 2009
>>> Author(s)           : J. van Elburg
>>> Category            : INFORMATIONAL
>>> Source              : IETF - NON WORKING GROUP
>>> Area                : N/A
>>> Stream              : IETF
>>> Verifying Party     : IESG
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From nobody Thu Mar 31 07:47:04 2016
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From: Brett Tate <brett@broadsoft.com>
References: <20160329135428.51F55180004@rfc-editor.org> <8ebfaf07d50da3c1dafb4083e5d2862b@mail.gmail.com> <56FD2F21.2070609@alum.mit.edu>
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Date: Thu, 31 Mar 2016 10:46:46 -0400
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, sipcore@ietf.org
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Archived-At: <http://mailarchive.ietf.org/arch/msg/sipcore/dqT-rqcTW4eOYtx6KTv4FOz6cWE>
Subject: Re: [sipcore] [Technical Errata Reported] RFCs 3515, 5502, 5360, ...
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Hi,

I agree; that was why I sent the following email in October 2013 while
sipcore was working on similar errata for RFC 3325.  I used that email and
aspects of the RFC 3325 errata to produce the recent errata for RFC 3515,
RFC 5002, RFC 5318, RFC 5360, and RFC 5502.  I have not check any headers
register with IANA since October 2013.

http://www.ietf.org/mail-archive/web/sipcore/current/msg05731.html

Thanks,
Brett

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul
Kyzivat
> Sent: Thursday, March 31, 2016 10:07 AM
> To: sipcore@ietf.org
> Subject: Re: [sipcore] [Technical Errata Reported] RFCs 3515, 5502,
5360, ...
>
> ISTM that it would be good to discuss all of these together, and perhaps
> solicit pointers to any others that need a similar fix.
>
> 	Thanks,
> 	Paul
>
> On 3/31/16 7:36 AM, Brett Tate wrote:
> > Copying sipcore since similar errata being discussed.
> >
> >> -----Original Message-----
> >> From: Brett Tate [mailto:brett@broadsoft.com]
> >> Sent: Tuesday, March 29, 2016 10:23 AM
> >> To: 'RFC Errata System'; 'HansErik.van.Elburg@ericsson.com';
> > 'iesg@ietf.org'
> >> Subject: RE: [Technical Errata Reported] RFC5502 (4648)
> >>
> >> Hi,
> >>
> >> My proposed resolution is similar to what occurred within RFC 3325
> > errata
> >> (3744 and 3894).
> >>
> >>> -----Original Message-----
> >>> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
> >>> Sent: Tuesday, March 29, 2016 9:54 AM
> >>> To: HansErik.van.Elburg@ericsson.com; iesg@ietf.org
> >>> Cc: brett@broadsoft.com; rfc-editor@rfc-editor.org
> >>> Subject: [Technical Errata Reported] RFC5502 (4648)
> >>>
> >>> The following errata report has been submitted for RFC5502, "The SIP
> >>> P- Served-User Private-Header (P-Header) for the 3GPP IP Multimedia
> >>> (IM) Core Network (CN) Subsystem".
> >>>
> >>> --------------------------------------
> >>> You may review the report below and at:
> >>> http://www.rfc-editor.org/errata_search.php?rfc=5502&eid=4648
> >>>
> >>> --------------------------------------
> >>> Type: Technical
> >>> Reported by: Brett Tate <brett@broadsoft.com>
> >>>
> >>> Section: 6
> >>>
> >>> Original Text
> >>> -------------
> >>> EQUAL, HCOLON, SEMI, name-addr, addr-spec, and generic-param are
> >>> defined in RFC 3261 [2].
> >>>
> >>>
> >>> Corrected Text
> >>> --------------
> >>> EQUAL, HCOLON, SEMI, name-addr, addr-spec, and generic-param are
> >>> defined in RFC 3261 [2].
> >>>
> >>> If the URI contains a comma, question mark or semicolon, the URI
> >>> MUST be enclosed in angle brackets (< and >).
> >>>
> >>> Notes
> >>> -----
> >>> If addr-spec is used when there are parameters, it is ambiguous if
> >>> the parameters are URI parameters or served-user-param.  For
> >>> consistency with RFC
> >>> 3261 section 20, the same bracket rule is indicated even if comma
> >>> and question mark do not cause an issue.
> >>>
> >>> Instructions:
> >>> -------------
> >>> This erratum is currently posted as "Reported". If necessary, please
> >>> use "Reply All" to discuss whether it should be verified or
rejected.
> >>> When a decision is reached, the verifying party (IESG) can log in to
> >>> change the status and edit the report, if necessary.
> >>>
> >>> --------------------------------------
> >>> RFC5502 (draft-vanelburg-sipping-served-user-08)
> >>> --------------------------------------
> >>> Title               : The SIP P-Served-User Private-Header
(P-Header)
> > for
> >> the
> >>> 3GPP IP Multimedia (IM) Core Network (CN) Subsystem
> >>> Publication Date    : April 2009
> >>> Author(s)           : J. van Elburg
> >>> Category            : INFORMATIONAL
> >>> Source              : IETF - NON WORKING GROUP
> >>> Area                : N/A
> >>> Stream              : IETF
> >>> Verifying Party     : IESG
> >
> > _______________________________________________
> > 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 Mar 31 14:48:33 2016
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To: Brett Tate <brett@broadsoft.com>
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Sender: worley@ariadne.com (Dale R. Worley)
Date: Thu, 31 Mar 2016 17:48:27 -0400
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] [Technical Errata Reported] RFCs 3515, 5502, 5360, ...
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Brett Tate <brett@broadsoft.com> writes:
> I agree; that was why I sent the following email in October 2013 while
> sipcore was working on similar errata for RFC 3325.  I used that email and
> aspects of the RFC 3325 errata to produce the recent errata for RFC 3515,
> RFC 5002, RFC 5318, RFC 5360, and RFC 5502.  I have not check any headers
> register with IANA since October 2013.
>
> http://www.ietf.org/mail-archive/web/sipcore/current/msg05731.html

Why don't we put in an erratum that edits RFC 3261 section 20.10.  The
old text is:

   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, semicolon, or question
   mark.  There may or may not be LWS between the display-name and the
   "<".

The new text adds:

   In any header field whose value syntax is 
       (name-addr / addr-spec) *(SEMI generic-param)
   or
       (name-addr / addr-spec) *(SEMI generic-param)
           *(COMMA (name-addr / addr-spec) *(SEMI generic-param))
   the "addr-spec" alternative may not be used if its value would
   contain a comma, semicolon, or question mark.

In a sense, that's what everybody *thinks* RFC 3261 requires.

Dale


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From: Robert Sparks <rjsparks@nostrum.com>
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Subject: Re: [sipcore] [Technical Errata Reported] RFCs 3515, 5502, 5360, ...
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Again, I don't think errata is the right way to skin this.
<http://mailarchive.ietf.org/arch/msg/sipcore/OZN1Eey88Ksex8fia20fCZH_8qE>

RjS

On 3/31/16 4:48 PM, Dale R. Worley wrote:
> Brett Tate <brett@broadsoft.com> writes:
>> I agree; that was why I sent the following email in October 2013 while
>> sipcore was working on similar errata for RFC 3325.  I used that email and
>> aspects of the RFC 3325 errata to produce the recent errata for RFC 3515,
>> RFC 5002, RFC 5318, RFC 5360, and RFC 5502.  I have not check any headers
>> register with IANA since October 2013.
>>
>> http://www.ietf.org/mail-archive/web/sipcore/current/msg05731.html
> Why don't we put in an erratum that edits RFC 3261 section 20.10.  The
> old text is:
>
>     Even if the "display-name" is empty, the "name-addr" form MUST be
>     used if the "addr-spec" contains a comma, semicolon, or question
>     mark.  There may or may not be LWS between the display-name and the
>     "<".
>
> The new text adds:
>
>     In any header field whose value syntax is
>         (name-addr / addr-spec) *(SEMI generic-param)
>     or
>         (name-addr / addr-spec) *(SEMI generic-param)
>             *(COMMA (name-addr / addr-spec) *(SEMI generic-param))
>     the "addr-spec" alternative may not be used if its value would
>     contain a comma, semicolon, or question mark.
>
> In a sense, that's what everybody *thinks* RFC 3261 requires.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

