
From nobody Mon Jan 11 13:32:25 2016
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To: "Olle E. Johansson" <oej@edvina.net>, Gonzalo Salgueiro <gsalguei@cisco.com>, "Vijay K. Gurbani" <vkg@bell-labs.com>, SIPCORE <sipcore@ietf.org>
From: "A. Jean Mahoney" <mahoney@nostrum.com>
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Date: Mon, 11 Jan 2016 15:31:58 -0600
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Subject: [sipcore] Going forward with draft-ietf-sipcore-dns-dual-stack
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Hi all,

As chairs, we are looking to see if can restart the work on this WG 
item. For the draft to go forward we need the authors and the working 
group to commit to working on and reviewing it.

There's outstanding feedback that needs to be incorporated into this 
expired draft:

https://mailarchive.ietf.org/arch/msg/sipcore/bvqJHA6kLHOE0GamFfeJjN450Qc
https://mailarchive.ietf.org/arch/msg/sipcore/LRdKLi9X2bn8bePrddQ4Feh5Ie0
https://mailarchive.ietf.org/arch/msg/sipcore/AgN1Os5ktSrCNLoymu2HzxDQrmo

Authors, can one of you commit to updating the draft with the above 
feedback?

Thanks!

Jean, as chair


From nobody Wed Jan 13 13:25:58 2016
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From: "A. Jean Mahoney" <mahoney@nostrum.com>
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Date: Wed, 13 Jan 2016 15:25:40 -0600
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Subject: Re: [sipcore] Going forward with draft-ietf-sipcore-dns-dual-stack
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Hi all,

Dale Worley will be taking up the pen on the draft with the authors' and 
chairs' thanks. The original authors will still be engaged and have 
offered assistance if needed.

In addition to the feedback below, Dale has identified some open issues 
and questions that he'll be bringing to the list soon.

Thanks!

Jean

On 1/11/16 3:31 PM, A. Jean Mahoney wrote:
> Hi all,
>
> As chairs, we are looking to see if can restart the work on this WG
> item. For the draft to go forward we need the authors and the working
> group to commit to working on and reviewing it.
>
> There's outstanding feedback that needs to be incorporated into this
> expired draft:
>
> https://mailarchive.ietf.org/arch/msg/sipcore/bvqJHA6kLHOE0GamFfeJjN450Qc
> https://mailarchive.ietf.org/arch/msg/sipcore/LRdKLi9X2bn8bePrddQ4Feh5Ie0
> https://mailarchive.ietf.org/arch/msg/sipcore/AgN1Os5ktSrCNLoymu2HzxDQrmo
>
> Authors, can one of you commit to updating the draft with the above
> feedback?
>
> Thanks!
>
> Jean, as chair
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Thu Jan 14 10:44:35 2016
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Subject: [sipcore] WG session for sipcore in Buenos Aires?
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Hi all,

Please think about sipcore topics (current and/or future) that we could 
discuss in a WG session in Buenos Aires, and bring them to the mailing 
list by the beginning of February.

Thanks!

Jean


From nobody Fri Jan 22 19:31:03 2016
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Subject: [sipcore] sipcore - New Meeting Session Request for IETF 95
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A new meeting session request has just been submitted by Adam Roach, a Chair of the sipcore working group.


---------------------------------------------------------
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):  30 Minutes
Number of Attendees: 100
Conflicts to Avoid: 
 First Priority: avtcore, perc, avtext, rtcweb, mmusic, stir, webpush, xrblock, netvc




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


From nobody Sun Jan 24 16:23:53 2016
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Subject: [sipcore] Going forward with draft-ietf-sipcore-dns-dual-stack
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I've been appointed the editor of draft-ietf-sipcore-dns-dual-stack.  In
the next few days, I'll be making updates in the next few days and
responding to outstanding reviews.

Dale


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Subject: [sipcore] draft-ietf-sipcore-dns-dual-stack: Simon Perreault's review
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I've edited in a number of Simon's changes.  However:

Simon Perreault <sperreault@jive.com> writes:
> - It's not clear to me whether Section 4.1 advocates parallel connections
> or not. Is Happy Eyeballs to be applied or not? Or are we not recommending
> one way or another? This needs to be made more explicit.

At this point, the relationship between this draft and Happy Eyeballs is
not clear to me.  My expectations of a combination of 3263 and "Happy
Eyeballs" are:

1. A client must look up both A and AAAA records for a destination in all
   situations where it looks up a destination.

2. A client whose initial attempt to reach a destination via IPv6 fails
   should quickly attempt to reach it via IPv4, regardless of whether the
   destination has declared that it prefers a second IPv6 address to its
   IPv4 address (RFC 6555 section 4 item 1).

3. A client must remember whether attempts to reach a destination via
   IPv6 have failed in the past, and if so, prefer IPv4 to reach it (RFC
   6555 section 4).

Item 1 does not seem to me to be normative.  RFC 3263 is a mess to read,
but any sensible integration of it with the RFCs it references seems to
me to imply that a SIP client (as RFC 3263 calls the sender) must look
up address records for all address families that it supports (in any of
the 3 situations in which it looks up address records).

Items 2 and 3 seem to me to be the essence of "Happy Eyeballs", and they
are normative changes to RFC 3263/2782, since when SRV records are
involved, the SRV records currently declare the order which the client
MUST use the specified addresses.

> - The guidance in Section 4.2 could benefit from stronger normative
> language:
> 
>    When indicating address family preference through SRV, IPv4-only and/
>    or IPv6-only clients should be prepared to handle SRV record sets
>    that don't resolve into an ip address in the address family used by
>    that client.  In such a case, the client should simply proceed to the
>    next priority and try the hostnames in the alternate address family.
> 
> Suggestion: s/should/MUST/g
> 
> And please remove the "and try the hostnames in the alternate address
> family." part, which makes no sense (a hostname cannot be "in an address
> family").

Actually, this paragraph should be moved out of section 4.2, because 4.2
is about how a client should indicate how it prefers to be contacted,
whereas this text is about *all* clients, that resolution may produce
addresses the client doesn't support, and it must be prepared to ignore
those addresses.

> minor
> =====
> 
> - Should this draft formally update RFC 3263?

Assuming that we're actually incorporating the Happy Eyeballs
processing, it updates RFC 3263:

- In regard to URIs that do not invoke SRV records, Happy Eyeballs
  processing provides more constraints on the sequence in which resolved
  addresses are used.  As a tightening of the requirements of RFC 3263,
  it is an update thereof.

- in regard to URIs that invoke SRV records, the sequence specified by
  the SRV records can be overridden.  This contravenes RFC 2782 as
  incorporated by RFC 3263, and so is an update.

However, Olle says:
> At one point it did, but the wg - especially Cullen Jennings - did not
> accept it. I personally still think it's needed.

How do we resolve this?

Dale


From nobody Sun Jan 24 17:09:18 2016
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Subject: [sipcore] draft-ietf-sipcore-dns-dual-stack:  branch parameters
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At the risk of opening a can of worms:

draft-ietf-sipcore-dns-dual-stack resembles RFC 3263 in that it
describes how to send a SIP request to a URI.  They provide a process
for resolving the URI into one or more network connection points
(address, port, and transport protocol), which the sender attempts to
contact in sequence until it successfully contacts the destination.

RFC 3263 states that the transmission to each address/port must have a
different Via branch parameter:

   If a failure occurs, the client SHOULD create a new request, which is
   identical to the previous, but has a different value of the Via
   branch ID than the previous (and therefore constitutes a new SIP
   transaction).  That request is sent to the next element in the list
   as specified by RFC 2782.

The nasty consequence of this is that if the destination receives one
request but its response (e.g., a 100 response) does not get back to the
sender in time, the sender can send another request to a different
address/port/transport that arrives at the same destination, but is
marked as being a different SIP transaction by the new branch parameter
value.  And so the destination processes this request as a different
branch of the SIP request.

I've seen this cause trouble in practice:  If proxy A routes to proxy B
using the URI <sip:example.com>, proxy A can attempt to contact proxy B
using UDP.  If it does not receive a response quickly, it can then
attempt to contact proxy B using TCP.  If there has been network
congestion, both the UDP datagram and the TCP SYN can arrive at proxy B
at the same time, and proxy B has no way to know that they are
"equivalent" and should be processed together.  Instead, proxy B will
forward one of them to the destination phone.  It will then forward the
other one to the destination phone, the phone rejects it with 482 (since
it's already ringing for another branch of this call), proxy B will then
sequentially-fork it to the phone's voice mail.  This leaves the
destination phone ringing and the caller's phone talking to the
destination user's voicemail.

A better user experience would be obtained if all copies of the request
were sent with the same branch parameter, so that the destination will
know to ignore all but the first copy that it received.

Dale


From nobody Sun Jan 24 23:12:14 2016
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Subject: Re: [sipcore] draft-ietf-sipcore-dns-dual-stack: Simon Perreault's review
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> On 25 Jan 2016, at 01:43, Dale R. Worley <worley@ariadne.com> wrote:
>=20
> I've edited in a number of Simon's changes.  However:
>=20
> Simon Perreault <sperreault@jive.com> writes:
>> - It's not clear to me whether Section 4.1 advocates parallel =
connections
>> or not. Is Happy Eyeballs to be applied or not? Or are we not =
recommending
>> one way or another? This needs to be made more explicit.
>=20
> At this point, the relationship between this draft and Happy Eyeballs =
is
> not clear to me.
This draft was meant to only clear up DNS lookups to prepare for another
draft that discusses Happy Eyeballs in SIP. But that one could not be =
done
if we did not have both A and AAAA records at the same time.

>  My expectations of a combination of 3263 and "Happy
> Eyeballs" are:
>=20
> 1. A client must look up both A and AAAA records for a destination in =
all
>   situations where it looks up a destination.
>=20
> 2. A client whose initial attempt to reach a destination via IPv6 =
fails
>   should quickly attempt to reach it via IPv4, regardless of whether =
the
>   destination has declared that it prefers a second IPv6 address to =
its
>   IPv4 address (RFC 6555 section 4 item 1).
The problem here is UDP. We=E2=80=99ve spent many hours discussing this.
For connection oriented protocols we can follow Happy Eyeballs,
but it doesn=E2=80=99t work for UDP. For UDP, we need a solution based=20=

on DNS. I think I have a presentation that outlines this somewhere.

>=20
> 3. A client must remember whether attempts to reach a destination via
>   IPv6 have failed in the past, and if so, prefer IPv4 to reach it =
(RFC
>   6555 section 4).
>=20
> Item 1 does not seem to me to be normative.  RFC 3263 is a mess to =
read,
> but any sensible integration of it with the RFCs it references seems =
to
> me to imply that a SIP client (as RFC 3263 calls the sender) must look
> up address records for all address families that it supports (in any =
of
> the 3 situations in which it looks up address records).
>=20
> Items 2 and 3 seem to me to be the essence of "Happy Eyeballs", and =
they
> are normative changes to RFC 3263/2782, since when SRV records are
> involved, the SRV records currently declare the order which the client
> MUST use the specified addresses.
>=20
>> - The guidance in Section 4.2 could benefit from stronger normative
>> language:
>>=20
>>   When indicating address family preference through SRV, IPv4-only =
and/
>>   or IPv6-only clients should be prepared to handle SRV record sets
>>   that don't resolve into an ip address in the address family used by
>>   that client.  In such a case, the client should simply proceed to =
the
>>   next priority and try the hostnames in the alternate address =
family.
>>=20
>> Suggestion: s/should/MUST/g
>>=20
>> And please remove the "and try the hostnames in the alternate address
>> family." part, which makes no sense (a hostname cannot be "in an =
address
>> family").
>=20
> Actually, this paragraph should be moved out of section 4.2, because =
4.2
> is about how a client should indicate how it prefers to be contacted,
> whereas this text is about *all* clients, that resolution may produce
> addresses the client doesn't support, and it must be prepared to =
ignore
> those addresses.
>=20
>> minor
>> =3D=3D=3D=3D=3D
>>=20
>> - Should this draft formally update RFC 3263?
>=20
> Assuming that we're actually incorporating the Happy Eyeballs
> processing, it updates RFC 3263:
>=20
> - In regard to URIs that do not invoke SRV records, Happy Eyeballs
>  processing provides more constraints on the sequence in which =
resolved
>  addresses are used.  As a tightening of the requirements of RFC 3263,
>  it is an update thereof.
>=20
> - in regard to URIs that invoke SRV records, the sequence specified by
>  the SRV records can be overridden.  This contravenes RFC 2782 as
>  incorporated by RFC 3263, and so is an update.
>=20
> However, Olle says:
>> At one point it did, but the wg - especially Cullen Jennings - did =
not
>> accept it. I personally still think it's needed.
>=20
> How do we resolve this?
If we write documents in order to guide developers, an update
would be easy to follow while browsing the RFCs.  And it is indeed
an update. The formal problem that I think worries people is that
if we update RFC 3263 the definition of =E2=80=9CSIP compliant=E2=80=9D =
will change
and that will affect many vendors. Personally I think that=E2=80=99s a =
good thing.


/O




From nobody Sun Jan 24 23:13:23 2016
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Subject: Re: [sipcore] draft-ietf-sipcore-dns-dual-stack: branch parameters
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> On 25 Jan 2016, at 02:09, Dale R. Worley <worley@ariadne.com> wrote:
> 
> At the risk of opening a can of worms:

> 
> draft-ietf-sipcore-dns-dual-stack resembles RFC 3263 in that it
> describes how to send a SIP request to a URI.  They provide a process
> for resolving the URI into one or more network connection points
> (address, port, and transport protocol), which the sender attempts to
> contact in sequence until it successfully contacts the destination.
> 
> RFC 3263 states that the transmission to each address/port must have a
> different Via branch parameter:
> 
>   If a failure occurs, the client SHOULD create a new request, which is
>   identical to the previous, but has a different value of the Via
>   branch ID than the previous (and therefore constitutes a new SIP
>   transaction).  That request is sent to the next element in the list
>   as specified by RFC 2782.
> 
> The nasty consequence of this is that if the destination receives one
> request but its response (e.g., a 100 response) does not get back to the
> sender in time, the sender can send another request to a different
> address/port/transport that arrives at the same destination, but is
> marked as being a different SIP transaction by the new branch parameter
> value.  And so the destination processes this request as a different
> branch of the SIP request.
> 
> I've seen this cause trouble in practice:  If proxy A routes to proxy B
> using the URI <sip:example.com>, proxy A can attempt to contact proxy B
> using UDP.  If it does not receive a response quickly, it can then
> attempt to contact proxy B using TCP.  If there has been network
> congestion, both the UDP datagram and the TCP SYN can arrive at proxy B
> at the same time, and proxy B has no way to know that they are
> "equivalent" and should be processed together.  Instead, proxy B will
> forward one of them to the destination phone.  It will then forward the
> other one to the destination phone, the phone rejects it with 482 (since
> it's already ringing for another branch of this call), proxy B will then
> sequentially-fork it to the phone's voice mail.  This leaves the
> destination phone ringing and the caller's phone talking to the
> destination user's voicemail.
> 
> A better user experience would be obtained if all copies of the request
> were sent with the same branch parameter, so that the destination will
> know to ignore all but the first copy that it received.

I am not sure this is related to the draft, but it does indicate that
RFC 3263 may need a generic overhaul.

/O


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Subject: [sipcore] sipcore - Update to a Meeting Session Request for IETF 95
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An update to a meeting session request has just been submitted by Adam Roach, a Chair of the sipcore working group.


---------------------------------------------------------
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: avtcore perc avtext rtcweb mmusic stir webpush xrblock netvc




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


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An update to a meeting session request has just been submitted by Adam Roach, a Chair of the sipcore working group.


---------------------------------------------------------
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: avtcore perc avtext rtcweb mmusic stir webpush xrblock netvc dispatch




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


From nobody Mon Jan 25 11:54:07 2016
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Subject: Re: [sipcore] draft-ietf-sipcore-dns-dual-stack: Simon Perreault's review
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"Olle E. Johansson" <oej@edvina.net> writes:
>> At this point, the relationship between this draft and Happy Eyeballs is
>> not clear to me.

> This draft was meant to only clear up DNS lookups to prepare for another
> draft that discusses Happy Eyeballs in SIP. But that one could not be done
> if we did not have both A and AAAA records at the same time.

If I understand you correctly:

The scope of this draft is quite narrow.  It is to update these two
passages of RFC 3263:

   4.2 Determining Port and IP Address

   [a host name with a port number:]
   If the TARGET was not a numeric IP address, but a port is present in
   the URI, the client performs an A or AAAA record lookup of the domain
   name.

   [...]

   [a host name without a port number, but no SRV records for the host name:]
   If no SRV records were found, the client performs an A or AAAA record
   lookup of the domain name.  The result will be a list of IP
   addresses, each of which can be contacted using the transport
   protocol determined previously, at the default port for that
   transport.  Processing then proceeds as described above for an
   explicit port once the A or AAAA records have been looked up.

to align with this passage of RFC 2782 (SRV records):

   Note that where this document refers to "address records", it means A
   RR's, AAAA RR's, or their most modern equivalent.

Is that correct?

Dale


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"Olle E. Johansson" <oej@edvina.net> writes:
> The formal problem that I think worries people is that if we update
> RFC 3263 the definition of "SIP compliant" will change and that will
> affect many vendors. Personally I think that's a good thing.

My understanding is that "SIP compliant", to be well-defined, must
always followed by a list of the RFCs that compliance is claimed for, so
the customers and vendors can always ask whether a device is compliant
with RFC 8123, or whatever it will be, as well as RFC 3263.  Certainly
this happened when "SIP" changed from RFC 2543 to RFC 3261.  But maybe
customers are acting differently these days.

Dale


From nobody Mon Jan 25 12:23:18 2016
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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"Penar, Russell" <Russell.Penar@centurylink.com> writes:
> I'm seeing non-ideal behavior from cpe and hoping this group can
> provide guidelines specific to the scenario below, and ultimately
> standardize ideal cpe behavior for the given context.  I haven't found
> any specifications defining the ideal behavior, and it was suggested
> during sipnoc to contact sipcore as an appropriate next step.

Unfortunately, Sipcore doesn't so much define ideal behavior as define
the behaviors that a UA (CPE) is permitted to have.  Within that
context, a UA can drop a call whenever it chooses.  In the use cases we
see, the general advice is "If the UA is behaving in a way that provides
a poor user experience, buy a different type of UA."  That usually gives
the UA manufacturers incentive to improve their UAs.

> Where a standing call is setup against the primary sbc; after a
> primary sbc mated pair failover, calls are at risk of being dropped if
> a tcp connection is not established back to the primary sbc.  This can
> be due to session audits from the platform, or more specific to my ask
> is a cpe re-Invite scenario.  For example, when cpe puts standing call
> on hold (after sbc failover), the sbc target for re-establishing the
> signaling tcp session is critical.
>
> I've seen cpe exhibit ideal behavior, where upon receipt of tcp rst
> the cpe connects back to the primary sbc.  In that ideal scenario, the
> call stays up, mid-call control doesn't fail, etc.  And I've seen
> non-ideal behavior where once the cpe attempts to put the call on
> hold, it reacts to the tcp rst by failing over to the secondary sbc
> and dropping the standing call.  Fwiw, I'm told the vendor exhibiting
> the ideal behavior only recently began to do so (reportedly from
> customer pressure to properly support geo-redundant setups as
> described above).

As a general rule, the failure of a TCP connection does not change the
state of any SIP call.  But if the UA wants to issue a re-INVITE for an
existing call, it will send an INVITE request using the "route set" of
the call.  The UA learns the route set during call set-up; the devices
on the path that the INVITE request takes will add Record-Route or
Contact headers containing addresses; the collection of these headers is
the route set and specifies the path which all further requests for that
call will take.

Since the SBC wants to see further requests for the call, it adds an
address for itself in either a Record-Route header or a Contact header
(depending on the technical details of the SBC).  If you want the calls
to fail over, it's critical that the address is one that routes to both
the primary and secondary SBCs -- it is a host name which DNS maps to
the addresses of both SBCs.

If the address is only for the SBC involved in the call setup, if that
SBC goes down and the UA attempts a re-INVITE, the UA can't send the
re-INVITE.  The UA may decide to tear down the call, but some UAs
don't -- instead, they assume that if the call truly fails, the user
will hang up and so they maintain the call until the user hangs up.

It's understood that if a UA attempts to send a request on a TCP
connection and the TCP connection produces some sort of error that the
UA should either attempt to reestablish the TCP connection or send the
request to an alternative destination (e.g., the other SBC).  This is
specified in RFC 3263 section 4.3:

   For SIP requests, failure occurs if the transaction layer reports a
   503 error response or a transport failure of some sort (generally,
   due to fatal ICMP errors in UDP or connection failures in TCP).
   Failure also occurs if the transaction layer times out without ever
   having received any response, provisional or final (i.e., timer B or
   timer F in RFC 3261 [1] fires).  If a failure occurs, the client
   SHOULD create a new request, which is identical to the previous, but
   has a different value of the Via branch ID than the previous (and
   therefore constitutes a new SIP transaction).  That request is sent
   to the next element in the list as specified by RFC 2782.

Dale


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From: "Penar, Russell" <Russell.Penar@centurylink.com>
To: "'Dale R. Worley'" <worley@ariadne.com>
Thread-Topic: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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References: <12C684AA1283B949BD80AC8273CF3A723F45A77F@PDDCWMBXEX504.ctl.intranet> (Russell.Penar@centurylink.com) <87a8nt9znz.fsf@hobgoblin.ariadne.com>
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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Thanks for the reply Dale, comments in line.
Regards,
Russ

-----Original Message-----
From: Dale R. Worley [mailto:worley@ariadne.com]
Sent: Monday, January 25, 2016 1:23 PM
To: Penar, Russell
Cc: sipcore@ietf.org
Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (b=
ehavior upon receipt of tcp rst)

"Penar, Russell" <Russell.Penar@centurylink.com> writes:
> I'm seeing non-ideal behavior from cpe and hoping this group can
> provide guidelines specific to the scenario below, and ultimately
> standardize ideal cpe behavior for the given context.  I haven't found
> any specifications defining the ideal behavior, and it was suggested
> during sipnoc to contact sipcore as an appropriate next step.

Unfortunately, Sipcore doesn't so much define ideal behavior as define the =
behaviors that a UA (CPE) is permitted to have.  Within that context, a UA =
can drop a call whenever it chooses.  In the use cases we see, the general =
advice is "If the UA is behaving in a way that provides a poor user experie=
nce, buy a different type of UA."  That usually gives the UA manufacturers =
incentive to improve their UAs.

RP - That's a fair stance, we have seen at least one vendor move to the "id=
eal" behavior presumably from customer pressure.  Would it also be fair to =
state Sipcore defines behavior that a UA is not permitted to have?  Perhaps=
 what I'm really trying to get here is a MUST NOT statement against connect=
ing to secondary targets upon receipt of a TCP RST.

> Where a standing call is setup against the primary sbc; after a
> primary sbc mated pair failover, calls are at risk of being dropped if
> a tcp connection is not established back to the primary sbc.  This can
> be due to session audits from the platform, or more specific to my ask
> is a cpe re-Invite scenario.  For example, when cpe puts standing call
> on hold (after sbc failover), the sbc target for re-establishing the
> signaling tcp session is critical.
>
> I've seen cpe exhibit ideal behavior, where upon receipt of tcp rst
> the cpe connects back to the primary sbc.  In that ideal scenario, the
> call stays up, mid-call control doesn't fail, etc.  And I've seen
> non-ideal behavior where once the cpe attempts to put the call on
> hold, it reacts to the tcp rst by failing over to the secondary sbc
> and dropping the standing call.  Fwiw, I'm told the vendor exhibiting
> the ideal behavior only recently began to do so (reportedly from
> customer pressure to properly support geo-redundant setups as
> described above).

As a general rule, the failure of a TCP connection does not change the stat=
e of any SIP call.  But if the UA wants to issue a re-INVITE for an existin=
g call, it will send an INVITE request using the "route set" of the call.  =
The UA learns the route set during call set-up; the devices on the path tha=
t the INVITE request takes will add Record-Route or Contact headers contain=
ing addresses; the collection of these headers is the route set and specifi=
es the path which all further requests for that call will take.

Since the SBC wants to see further requests for the call, it adds an addres=
s for itself in either a Record-Route header or a Contact header (depending=
 on the technical details of the SBC).  If you want the calls to fail over,=
 it's critical that the address is one that routes to both the primary and =
secondary SBCs -- it is a host name which DNS maps to the addresses of both=
 SBCs.

RP - We're using DNS SRV to support geo-redundant access SBCs.  The "route =
set" doesn't appear to be an issue, as the re-Invite does go to the correct=
 location (primary), but upon receipt of the TCP RST the UA immediately dro=
ps the call, sets up a new TCP connection to the secondary and Registers.  =
The UA gathers primary and secondary targets (for outbound proxy FQDN) upon=
 boot up and then accordingly based on TTL.

If the address is only for the SBC involved in the call setup, if that SBC =
goes down and the UA attempts a re-INVITE, the UA can't send the re-INVITE.=
  The UA may decide to tear down the call, but some UAs don't -- instead, t=
hey assume that if the call truly fails, the user will hang up and so they =
maintain the call until the user hangs up.

RP - I agree in principle when using SRV the SBC's Contact/R-R should have =
the SBC FQDN and not a specific IP address.  With use of outbound proxy, I =
haven't seen issues with UA's not failing over when Contact (for example) i=
s returned with only the SBC (HA pair IP address) involved in the call setu=
p.

It's understood that if a UA attempts to send a request on a TCP connection=
 and the TCP connection produces some sort of error that the UA should eith=
er attempt to reestablish the TCP connection or send the request to an alte=
rnative destination (e.g., the other SBC).  This is specified in RFC 3263 s=
ection 4.3:

   For SIP requests, failure occurs if the transaction layer reports a
   503 error response or a transport failure of some sort (generally,
   due to fatal ICMP errors in UDP or connection failures in TCP).
   Failure also occurs if the transaction layer times out without ever
   having received any response, provisional or final (i.e., timer B or
   timer F in RFC 3261 [1] fires).  If a failure occurs, the client
   SHOULD create a new request, which is identical to the previous, but
   has a different value of the Via branch ID than the previous (and
   therefore constitutes a new SIP transaction).  That request is sent
   to the next element in the list as specified by RFC 2782.

RP - Could we amend this section of 3263, to reflect a UA SHOULD (specific =
to context of TCP) try the current element where a SIP dialog is establishe=
d before trying the next element per 2782?

Dale

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From nobody Tue Jan 26 00:50:56 2016
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> On 25 Jan 2016, at 20:54, Dale R. Worley <worley@ariadne.com> wrote:
>=20
> "Olle E. Johansson" <oej@edvina.net> writes:
>>> At this point, the relationship between this draft and Happy =
Eyeballs is
>>> not clear to me.
>=20
>> This draft was meant to only clear up DNS lookups to prepare for =
another
>> draft that discusses Happy Eyeballs in SIP. But that one could not be =
done
>> if we did not have both A and AAAA records at the same time.
>=20
> If I understand you correctly:
>=20
> The scope of this draft is quite narrow.  It is to update these two
> passages of RFC 3263:
>=20
>   4.2 Determining Port and IP Address
>=20
>   [a host name with a port number:]
>   If the TARGET was not a numeric IP address, but a port is present in
>   the URI, the client performs an A or AAAA record lookup of the =
domain
>   name.
>=20
>   [...]
>=20
>   [a host name without a port number, but no SRV records for the host =
name:]
>   If no SRV records were found, the client performs an A or AAAA =
record
>   lookup of the domain name.  The result will be a list of IP
>   addresses, each of which can be contacted using the transport
>   protocol determined previously, at the default port for that
>   transport.  Processing then proceeds as described above for an
>   explicit port once the A or AAAA records have been looked up.
>=20
> to align with this passage of RFC 2782 (SRV records):
>=20
>   Note that where this document refers to "address records", it means =
A
>   RR's, AAAA RR's, or their most modern equivalent.
>=20
> Is that correct?

Yes, the one important point was to make sure that =E2=80=9CA or AAAA=E2=80=
=9D was redefined to clearly
mean =E2=80=9CA and AAAA=E2=80=9D records so that we would always have =
the possibility to
perform Happy Eyeballs. It=E2=80=99s mentioned in several places in RFC =
3263 if I remember=20
correctly and unfortunately copied to other RFCs.

The second part was using the DNS SRV to indicate the server manager=E2=80=
=99s
order of preference.

We=E2=80=99ve been discussing back and forth wheather the Happy Eyeballs =
part
should be part of this draft or not, but have decided multiple times to =
split it up.
It may or may not be the right way :-)

/O=


From nobody Tue Jan 26 00:52:16 2016
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> On 25 Jan 2016, at 20:58, Dale R. Worley <worley@ariadne.com> wrote:
>=20
> "Olle E. Johansson" <oej@edvina.net> writes:
>> The formal problem that I think worries people is that if we update
>> RFC 3263 the definition of "SIP compliant" will change and that will
>> affect many vendors. Personally I think that's a good thing.
>=20
> My understanding is that "SIP compliant", to be well-defined, must
> always followed by a list of the RFCs that compliance is claimed for, =
so
> the customers and vendors can always ask whether a device is compliant
> with RFC 8123, or whatever it will be, as well as RFC 3263.  Certainly
> this happened when "SIP" changed from RFC 2543 to RFC 3261.  But maybe
> customers are acting differently these days.

I am not sure, but RFC 3261 clearly depends on 3263. If we update 3263,
doesn=E2=80=99t that mean that 3261 depends on the new RFC too?

So in order to be 3261 SIP-compliant, which is quite often a =
requirement,
it would also mean being compliant with proper dual stack handling,
as defined by this new draft=E2=80=A6

/O=


From nobody Tue Jan 26 08:04:35 2016
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Subject: Re: [sipcore] draft-ietf-sipcore-dns-dual-stack: Simon Perreault's review
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On 1/26/16 3:52 AM, Olle E. Johansson wrote:
>
>> On 25 Jan 2016, at 20:58, Dale R. Worley <worley@ariadne.com> wrote:
>>
>> "Olle E. Johansson" <oej@edvina.net> writes:
>>> The formal problem that I think worries people is that if we update
>>> RFC 3263 the definition of "SIP compliant" will change and that will
>>> affect many vendors. Personally I think that's a good thing.
>>
>> My understanding is that "SIP compliant", to be well-defined, must
>> always followed by a list of the RFCs that compliance is claimed for, so
>> the customers and vendors can always ask whether a device is compliant
>> with RFC 8123, or whatever it will be, as well as RFC 3263.  Certainly
>> this happened when "SIP" changed from RFC 2543 to RFC 3261.  But maybe
>> customers are acting differently these days.
>
> I am not sure, but RFC 3261 clearly depends on 3263. If we update 3263,
> doesn’t that mean that 3261 depends on the new RFC too?
>
> So in order to be 3261 SIP-compliant, which is quite often a requirement,
> it would also mean being compliant with proper dual stack handling,
> as defined by this new draft…

IIUC, you cannot revise history. Something that was once 3261 compliant 
remains so, regardless of what revisions are made later. That is why we 
don't reissue revised RFCs using the original numbers.

Revising 3263 doesn't automatically force implementations of 3261 to 
implement that revision.

What it does do is make compliance to 3261 something that should, in 
itself, be considered insufficient. If we wanted to provide a single RFC 
number that people could claim compliance to that would reflect the 
current understanding of sufficiency then we would need to issue a 
revision to 3261.

	Thanks,
	Paul


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From: Brett Tate <brett@broadsoft.com>
References: <87fuxmbh3d.fsf@hobgoblin.ariadne.com>
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Date: Tue, 26 Jan 2016 11:14:23 -0500
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To: "Dale R. Worley" <worley@ariadne.com>, sipcore@ietf.org
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Subject: Re: [sipcore] draft-ietf-sipcore-dns-dual-stack: branch parameters
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Hi,

> A better user experience would be obtained if all copies of
> the request were sent with the same branch parameter, so that
> the destination will know to ignore all but the first copy
> that it received.

Although merged request handling can potentially be improved within SIP,
I'm not sure that the proposed improvement is better than current
behavior.  However, this draft will definitely cause more merged request
situations (and delays before advancing to reachable server).  I'm also
not sure if the proposed improvement actually addresses the real issue.
Are you experiencing a transaction correlation issue or just a lack of 1xx
to compliantly allow you to send CANCEL?

Changing the branch allows the sender to know which destination actually
worked.  Without that knowledge, it would not know the destination and
port for sending ACK-for-non-2xx and/or CANCEL (unless added something
else to the Via).

The requests can take different paths before merging.  Ignoring requests
from the different path can cause the path to appear non responsive
(although intermediary can send 100 response to reduce some of the side
effects).

Thanks,
Brett


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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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Russ,

Are you aware of RFC5626? It provides an explicit mechanism for 
redundant registration paths.

	Thanks,
	Paul

On 1/25/16 6:23 PM, Penar, Russell wrote:
> Thanks for the reply Dale, comments in line.
> Regards,
> Russ
>
> -----Original Message-----
> From: Dale R. Worley [mailto:worley@ariadne.com]
> Sent: Monday, January 25, 2016 1:23 PM
> To: Penar, Russell
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
>
> "Penar, Russell" <Russell.Penar@centurylink.com> writes:
>> I'm seeing non-ideal behavior from cpe and hoping this group can
>> provide guidelines specific to the scenario below, and ultimately
>> standardize ideal cpe behavior for the given context.  I haven't found
>> any specifications defining the ideal behavior, and it was suggested
>> during sipnoc to contact sipcore as an appropriate next step.
>
> Unfortunately, Sipcore doesn't so much define ideal behavior as define the behaviors that a UA (CPE) is permitted to have.  Within that context, a UA can drop a call whenever it chooses.  In the use cases we see, the general advice is "If the UA is behaving in a way that provides a poor user experience, buy a different type of UA."  That usually gives the UA manufacturers incentive to improve their UAs.
>
> RP - That's a fair stance, we have seen at least one vendor move to the "ideal" behavior presumably from customer pressure.  Would it also be fair to state Sipcore defines behavior that a UA is not permitted to have?  Perhaps what I'm really trying to get here is a MUST NOT statement against connecting to secondary targets upon receipt of a TCP RST.
>
>> Where a standing call is setup against the primary sbc; after a
>> primary sbc mated pair failover, calls are at risk of being dropped if
>> a tcp connection is not established back to the primary sbc.  This can
>> be due to session audits from the platform, or more specific to my ask
>> is a cpe re-Invite scenario.  For example, when cpe puts standing call
>> on hold (after sbc failover), the sbc target for re-establishing the
>> signaling tcp session is critical.
>>
>> I've seen cpe exhibit ideal behavior, where upon receipt of tcp rst
>> the cpe connects back to the primary sbc.  In that ideal scenario, the
>> call stays up, mid-call control doesn't fail, etc.  And I've seen
>> non-ideal behavior where once the cpe attempts to put the call on
>> hold, it reacts to the tcp rst by failing over to the secondary sbc
>> and dropping the standing call.  Fwiw, I'm told the vendor exhibiting
>> the ideal behavior only recently began to do so (reportedly from
>> customer pressure to properly support geo-redundant setups as
>> described above).
>
> As a general rule, the failure of a TCP connection does not change the state of any SIP call.  But if the UA wants to issue a re-INVITE for an existing call, it will send an INVITE request using the "route set" of the call.  The UA learns the route set during call set-up; the devices on the path that the INVITE request takes will add Record-Route or Contact headers containing addresses; the collection of these headers is the route set and specifies the path which all further requests for that call will take.
>
> Since the SBC wants to see further requests for the call, it adds an address for itself in either a Record-Route header or a Contact header (depending on the technical details of the SBC).  If you want the calls to fail over, it's critical that the address is one that routes to both the primary and secondary SBCs -- it is a host name which DNS maps to the addresses of both SBCs.
>
> RP - We're using DNS SRV to support geo-redundant access SBCs.  The "route set" doesn't appear to be an issue, as the re-Invite does go to the correct location (primary), but upon receipt of the TCP RST the UA immediately drops the call, sets up a new TCP connection to the secondary and Registers.  The UA gathers primary and secondary targets (for outbound proxy FQDN) upon boot up and then accordingly based on TTL.
>
> If the address is only for the SBC involved in the call setup, if that SBC goes down and the UA attempts a re-INVITE, the UA can't send the re-INVITE.  The UA may decide to tear down the call, but some UAs don't -- instead, they assume that if the call truly fails, the user will hang up and so they maintain the call until the user hangs up.
>
> RP - I agree in principle when using SRV the SBC's Contact/R-R should have the SBC FQDN and not a specific IP address.  With use of outbound proxy, I haven't seen issues with UA's not failing over when Contact (for example) is returned with only the SBC (HA pair IP address) involved in the call setup.
>
> It's understood that if a UA attempts to send a request on a TCP connection and the TCP connection produces some sort of error that the UA should either attempt to reestablish the TCP connection or send the request to an alternative destination (e.g., the other SBC).  This is specified in RFC 3263 section 4.3:
>
>     For SIP requests, failure occurs if the transaction layer reports a
>     503 error response or a transport failure of some sort (generally,
>     due to fatal ICMP errors in UDP or connection failures in TCP).
>     Failure also occurs if the transaction layer times out without ever
>     having received any response, provisional or final (i.e., timer B or
>     timer F in RFC 3261 [1] fires).  If a failure occurs, the client
>     SHOULD create a new request, which is identical to the previous, but
>     has a different value of the Via branch ID than the previous (and
>     therefore constitutes a new SIP transaction).  That request is sent
>     to the next element in the list as specified by RFC 2782.
>
> RP - Could we amend this section of 3263, to reflect a UA SHOULD (specific to context of TCP) try the current element where a SIP dialog is established before trying the next element per 2782?
>
> Dale
>
> This communication is the property of CenturyLink and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in error, please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.
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>


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From: Brett Tate <brett@broadsoft.com>
References: <87fuxmbh3d.fsf@hobgoblin.ariadne.com> 29e056ac5c7ce97a51afee4e75e6619b@mail.gmail.com
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Date: Tue, 26 Jan 2016 12:34:26 -0500
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To: "Dale R. Worley" <worley@ariadne.com>, sipcore@ietf.org
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Subject: Re: [sipcore] draft-ietf-sipcore-dns-dual-stack: branch parameters
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Hi,

Sorry about resend.  After reading the raised issue again, you can ignore
my question since I now understand the service interaction issue that you
were raising.


> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: Tuesday, January 26, 2016 11:14 AM
> To: 'Dale R. Worley'; 'sipcore@ietf.org'
> Subject: RE: [sipcore] draft-ietf-sipcore-dns-dual-stack: branch
parameters
>
> Hi,
>
> > A better user experience would be obtained if all copies of the
> > request were sent with the same branch parameter, so that the
> > destination will know to ignore all but the first copy that it
> > received.
>
> Although merged request handling can potentially be improved within SIP,
I'm
> not sure that the proposed improvement is better than current behavior.
> However, this draft will definitely cause more merged request situations
(and
> delays before advancing to reachable server).  I'm also not sure if the
> proposed improvement actually addresses the real issue.  Are you
experiencing
> a transaction correlation issue or just a lack of 1xx to compliantly
allow
> you to send CANCEL?
>
> Changing the branch allows the sender to know which destination actually
> worked.  Without that knowledge, it would not know the destination and
port
> for sending ACK-for-non-2xx and/or CANCEL (unless added something else
to the
> Via).
>
> The requests can take different paths before merging.  Ignoring requests
from
> the different path can cause the path to appear non responsive (although
> intermediary can send 100 response to reduce some of the side effects).
>
> Thanks,
> Brett


From nobody Tue Jan 26 12:39:38 2016
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From: "Penar, Russell" <Russell.Penar@centurylink.com>
To: "'pkyzivat@alum.mit.edu'" <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] sip/tcp with primary & secondary outbound 
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound
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Thank you Paul,
I am aware and somewhat familiar with 5626, largely for the Keep-Alive cont=
ent.
I'm missing the details in 5626 addressing the TCP/RST issue for an HA prox=
y as I described.

3.3 gets close with a possible solution, however in practice UAs I encounte=
r have a single outbound proxy field and don't support dual registration pa=
ths.  If they did, I believe my scenario is still an issue given most comme=
rcial proxies are stateful b2bua's (SBCs) and the second path won't recogni=
ze the session re-Invite as valid.
Regards,
Russ

-----Original Message-----
----------------------------------------------------------------------
Date: Tue, 26 Jan 2016 11:30:52 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
To: sipcore@ietf.org
Subject: Re: [sipcore] sip/tcp with primary & secondary outbound
        proxies (behavior upon receipt of tcp rst)
Message-ID: <56A79F3C.1060201@alum.mit.edu>
Content-Type: text/plain; charset=3Dwindows-1252; format=3Dflowed

Russ,

Are you aware of RFC5626? It provides an explicit mechanism for redundant r=
egistration paths.

        Thanks,
        Paul

On 1/25/16 6:23 PM, Penar, Russell wrote:
> Thanks for the reply Dale, comments in line.
> Regards,
> Russ
>
> -----Original Message-----
> From: Dale R. Worley [mailto:worley@ariadne.com]
> Sent: Monday, January 25, 2016 1:23 PM
> To: Penar, Russell
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] sip/tcp with primary & secondary outbound
> proxies (behavior upon receipt of tcp rst)
>
> "Penar, Russell" <Russell.Penar@centurylink.com> writes:
>> I'm seeing non-ideal behavior from cpe and hoping this group can
>> provide guidelines specific to the scenario below, and ultimately
>> standardize ideal cpe behavior for the given context.  I haven't
>> found any specifications defining the ideal behavior, and it was
>> suggested during sipnoc to contact sipcore as an appropriate next step.
>
> Unfortunately, Sipcore doesn't so much define ideal behavior as define th=
e behaviors that a UA (CPE) is permitted to have.  Within that context, a U=
A can drop a call whenever it chooses.  In the use cases we see, the genera=
l advice is "If the UA is behaving in a way that provides a poor user exper=
ience, buy a different type of UA."  That usually gives the UA manufacturer=
s incentive to improve their UAs.
>
> RP - That's a fair stance, we have seen at least one vendor move to the "=
ideal" behavior presumably from customer pressure.  Would it also be fair t=
o state Sipcore defines behavior that a UA is not permitted to have?  Perha=
ps what I'm really trying to get here is a MUST NOT statement against conne=
cting to secondary targets upon receipt of a TCP RST.
>
>> Where a standing call is setup against the primary sbc; after a
>> primary sbc mated pair failover, calls are at risk of being dropped
>> if a tcp connection is not established back to the primary sbc.  This
>> can be due to session audits from the platform, or more specific to
>> my ask is a cpe re-Invite scenario.  For example, when cpe puts
>> standing call on hold (after sbc failover), the sbc target for
>> re-establishing the signaling tcp session is critical.
>>
>> I've seen cpe exhibit ideal behavior, where upon receipt of tcp rst
>> the cpe connects back to the primary sbc.  In that ideal scenario,
>> the call stays up, mid-call control doesn't fail, etc.  And I've seen
>> non-ideal behavior where once the cpe attempts to put the call on
>> hold, it reacts to the tcp rst by failing over to the secondary sbc
>> and dropping the standing call.  Fwiw, I'm told the vendor exhibiting
>> the ideal behavior only recently began to do so (reportedly from
>> customer pressure to properly support geo-redundant setups as
>> described above).
>
> As a general rule, the failure of a TCP connection does not change the st=
ate of any SIP call.  But if the UA wants to issue a re-INVITE for an exist=
ing call, it will send an INVITE request using the "route set" of the call.=
  The UA learns the route set during call set-up; the devices on the path t=
hat the INVITE request takes will add Record-Route or Contact headers conta=
ining addresses; the collection of these headers is the route set and speci=
fies the path which all further requests for that call will take.
>
> Since the SBC wants to see further requests for the call, it adds an addr=
ess for itself in either a Record-Route header or a Contact header (dependi=
ng on the technical details of the SBC).  If you want the calls to fail ove=
r, it's critical that the address is one that routes to both the primary an=
d secondary SBCs -- it is a host name which DNS maps to the addresses of bo=
th SBCs.
>
> RP - We're using DNS SRV to support geo-redundant access SBCs.  The "rout=
e set" doesn't appear to be an issue, as the re-Invite does go to the corre=
ct location (primary), but upon receipt of the TCP RST the UA immediately d=
rops the call, sets up a new TCP connection to the secondary and Registers.=
  The UA gathers primary and secondary targets (for outbound proxy FQDN) up=
on boot up and then accordingly based on TTL.
>
> If the address is only for the SBC involved in the call setup, if that SB=
C goes down and the UA attempts a re-INVITE, the UA can't send the re-INVIT=
E.  The UA may decide to tear down the call, but some UAs don't -- instead,=
 they assume that if the call truly fails, the user will hang up and so the=
y maintain the call until the user hangs up.
>
> RP - I agree in principle when using SRV the SBC's Contact/R-R should hav=
e the SBC FQDN and not a specific IP address.  With use of outbound proxy, =
I haven't seen issues with UA's not failing over when Contact (for example)=
 is returned with only the SBC (HA pair IP address) involved in the call se=
tup.
>
> It's understood that if a UA attempts to send a request on a TCP connecti=
on and the TCP connection produces some sort of error that the UA should ei=
ther attempt to reestablish the TCP connection or send the request to an al=
ternative destination (e.g., the other SBC).  This is specified in RFC 3263=
 section 4.3:
>
>     For SIP requests, failure occurs if the transaction layer reports a
>     503 error response or a transport failure of some sort (generally,
>     due to fatal ICMP errors in UDP or connection failures in TCP).
>     Failure also occurs if the transaction layer times out without ever
>     having received any response, provisional or final (i.e., timer B or
>     timer F in RFC 3261 [1] fires).  If a failure occurs, the client
>     SHOULD create a new request, which is identical to the previous, but
>     has a different value of the Via branch ID than the previous (and
>     therefore constitutes a new SIP transaction).  That request is sent
>     to the next element in the list as specified by RFC 2782.
>
> RP - Could we amend this section of 3263, to reflect a UA SHOULD (specifi=
c to context of TCP) try the current element where a SIP dialog is establis=
hed before trying the next element per 2782?
>
> Dale
>
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>
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From: worley@ariadne.com (Dale R. Worley)
To: Brett Tate <brett@broadsoft.com>
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Subject: Re: [sipcore] draft-ietf-sipcore-dns-dual-stack: branch parameters
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Brett Tate <brett@broadsoft.com> writes:
> Changing the branch allows the sender to know which destination actually
> worked.

Yes, that's true.  I've been assuming without thinking about it that if
a 100 response arrives, the recipient can tell from the source
address/port of the 100 what destination/address port was used for the
request that is being responded to.  But of course, that's not true.

And that is the fatal flaw in my proposal, unless "added something else
to the Via", as you say.

Dale


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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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"Penar, Russell" <Russell.Penar@centurylink.com> writes:
> RP - That's a fair stance, we have seen at least one vendor move to
> the "ideal" behavior presumably from customer pressure.  Would it also
> be fair to state Sipcore defines behavior that a UA is not permitted
> to have?

It's true that Sipcore defines behavior that a UA is not permitted to
have, because that is the logical negation of the behavior that it is
required to have.

> Perhaps what I'm really trying to get here is a MUST NOT statement
> against connecting to secondary targets upon receipt of a TCP RST.

I don't think that there is any intention to make such a requirement.
In principle, a UA has to contact targets in the order specified by RFC
3263, e.g., the SRV records for the URI, but in practice -- given
transient network failures -- there can be no guarantee.

It's not clear to me what that matters to you.  You wrote:

> Most, if not all, vendor sbc mated pairs maintain signaling/media state,
> but not tcp state (reportedly doing so is significantly hard/expensive).

If the SBCs share call (signaling/media) state, then the re-INVITE
doesn't have to go to the same SBC that the original INVITE did.

On the other hand, if updates to a call must be made at the same SBC
that the call was established (that is, if they don't share signaling
state), then an SBC should Record-Route itself with a URI which routes
to itself only.  And in that case, if the UA's TCP connection to the SBC
fails, it will be forced to reestablish a TCP connection to that SBC,
because the URI in the route set only routes to that address.

> RP - We're using DNS SRV to support geo-redundant access SBCs.  The
> "route set" doesn't appear to be an issue, as the re-Invite does go to
> the correct location (primary), but upon receipt of the TCP RST the UA
> immediately drops the call, sets up a new TCP connection to the
> secondary and Registers.  The UA gathers primary and secondary targets
> (for outbound proxy FQDN) upon boot up and then accordingly based on
> TTL.

That's a strange UA; it seems to consider the secondary SBC to be a
different destination from the primary SBC, rather than being two
destination addresses for one destination URI.

> RP - Could we amend this section of 3263, to reflect a UA SHOULD
> (specific to context of TCP) try the current element where a SIP
> dialog is established before trying the next element per 2782?

I don't think there would be much support for that, at least not without
considerably more justification.

What I don't seem to be understanding is how the two SBCs are presented
to the UA.  You write as if they are redudant deeply in SIP, so that
they share dialog (signaling) state.  In that case, there is some URI
which maps to *both* of them, and the UA is told to register at that one
URI.  Of course, if the UA needs to update a call, it can send the
re-INVITE to either SBC, because both SBCs can manage the state of the
call.

But your failure mode sounds like the UA is told of the two SBCs
separately, and doesn't treat them as alternative addresses for the
"same" destination.  Of course, in that case, the UA doesn't know that
it can send re-INVITEs to the secondary SBC to update calls that were
placed through the primary SBC.

Dale


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From: "Penar, Russell" <Russell.Penar@centurylink.com>
To: "'Dale R. Worley'" <worley@ariadne.com>
Thread-Topic: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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Date: Wed, 27 Jan 2016 20:52:32 +0000
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References: <12C684AA1283B949BD80AC8273CF3A723F57F803@PDDCWMBXEX504.ctl.intranet> (Russell.Penar@centurylink.com) <87mvrr7owa.fsf@hobgoblin.ariadne.com>
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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Thank you Dale,
Apologies for any confusion, I worked to clear it up in line below.
Regards,
Russ

-----Original Message-----
From: Dale R. Worley [mailto:worley@ariadne.com]
Sent: Tuesday, January 26, 2016 7:11 PM
To: Penar, Russell
Cc: sipcore@ietf.org
Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (b=
ehavior upon receipt of tcp rst)

"Penar, Russell" <Russell.Penar@centurylink.com> writes:
> RP - That's a fair stance, we have seen at least one vendor move to
> the "ideal" behavior presumably from customer pressure.  Would it also
> be fair to state Sipcore defines behavior that a UA is not permitted
> to have?

It's true that Sipcore defines behavior that a UA is not permitted to have,=
 because that is the logical negation of the behavior that it is required t=
o have.

> Perhaps what I'm really trying to get here is a MUST NOT statement
> against connecting to secondary targets upon receipt of a TCP RST.

I don't think that there is any intention to make such a requirement.
In principle, a UA has to contact targets in the order specified by RFC 326=
3, e.g., the SRV records for the URI, but in practice -- given transient ne=
twork failures -- there can be no guarantee.

RP - It's a very specific case, where the single primary target in SRV cons=
ists of 2 SBCs in a Highly Available active/hot-standby configuration.  We =
want to steer UA makers away from taking a simple TCP RST as a reason to mo=
ve to the next target.  In doing so, they will force active calls to be dro=
pped.  I see value in trying a TCP connection with the primary before movin=
g to the next and dropping calls.

It's not clear to me what that matters to you.  You wrote:

> Most, if not all, vendor sbc mated pairs maintain signaling/media
> state, but not tcp state (reportedly doing so is significantly hard/expen=
sive).

If the SBCs share call (signaling/media) state, then the re-INVITE doesn't =
have to go to the same SBC that the original INVITE did.

On the other hand, if updates to a call must be made at the same SBC that t=
he call was established (that is, if they don't share signaling state), the=
n an SBC should Record-Route itself with a URI which routes to itself only.=
  And in that case, if the UA's TCP connection to the SBC fails, it will be=
 forced to reestablish a TCP connection to that SBC, because the URI in the=
 route set only routes to that address.

> RP - We're using DNS SRV to support geo-redundant access SBCs.  The
> "route set" doesn't appear to be an issue, as the re-Invite does go to
> the correct location (primary), but upon receipt of the TCP RST the UA
> immediately drops the call, sets up a new TCP connection to the
> secondary and Registers.  The UA gathers primary and secondary targets
> (for outbound proxy FQDN) upon boot up and then accordingly based on
> TTL.

That's a strange UA; it seems to consider the secondary SBC to be a differe=
nt destination from the primary SBC, rather than being two destination addr=
esses for one destination URI.
RP - In the context of how it gathers the targets it is a very typical UA, =
market leader type.

> RP - Could we amend this section of 3263, to reflect a UA SHOULD
> (specific to context of TCP) try the current element where a SIP
> dialog is established before trying the next element per 2782?

I don't think there would be much support for that, at least not without co=
nsiderably more justification.

RP - Yeah, I think I wasn't clear enough in my problem statement (rely too =
much on pictures in other forums).  This is a very common deployment model =
for medium to large VoIP providers, at least with respect to how the SBCs a=
re deployed.

What I don't seem to be understanding is how the two SBCs are presented to =
the UA.  You write as if they are redudant deeply in SIP, so that they shar=
e dialog (signaling) state.  In that case, there is some URI which maps to =
*both* of them, and the UA is told to register at that one URI.  Of course,=
 if the UA needs to update a call, it can send the re-INVITE to either SBC,=
 because both SBCs can manage the state of the call.

RP - Each target in DNS (e.g. SBC1- weighted primary target and SBC2- weigh=
ted secondary target) consists of a mated pair of SBCs.  The mated pair is =
deeply redundant in SIP, but not in TCP.  The lack of TCP redundancy is wha=
t leads us to the TCP RST scenario.  One of the mated pair has faulted or b=
een taken down for maintenance; RTP is still flowing over UDP via the now a=
ctive second mate in the pair, and given a new TCP connection the SIP chann=
el would be healthy as well.  So the UA is causing an unnecessary call drop=
 by considering the RST a complete primary target failure and moving on.

But your failure mode sounds like the UA is told of the two SBCs separately=
, and doesn't treat them as alternative addresses for the "same" destinatio=
n.  Of course, in that case, the UA doesn't know that it can send re-INVITE=
s to the secondary SBC to update calls that were placed through the primary=
 SBC.

RP - The destination is a single FQDN - e.g. sbc.carrier.com.  DNS resolves=
 that to two weighted targets, essentially a primary and failover scheme.  =
The UA is correct in assuming once it connects to the second target all bet=
s are off with any existing dialogs.

Dale

This communication is the property of CenturyLink and may contain confident=
ial or privileged information. Unauthorized use of this communication is st=
rictly prohibited and may be unlawful. If you have received this communicat=
ion in error, please immediately notify the sender by reply e-mail and dest=
roy all copies of the communication and any attachments.


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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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"Penar, Russell" <Russell.Penar@centurylink.com> writes:
> RP - It's a very specific case, where the single primary target in SRV
> consists of 2 SBCs in a Highly Available active/hot-standby
> configuration.  We want to steer UA makers away from taking a simple
> TCP RST as a reason to move to the next target.  In doing so, they
> will force active calls to be dropped.  I see value in trying a TCP
> connection with the primary before moving to the next and dropping
> calls.

If I understand you correctly, the UA is configured with a single FQDN,
which maps through SRV records to two SBCs.  The SBCs share signaling
and media state.

Initially, the UA creates a call by contacting one SBC -- from the SIP
point of view, it establishes a TCP connection to one of the addresses
associated with the FQDN for the proxy to which it is to send calls.

In the failure case, the first SBC goes down.  Since the SBCs share
media state, the media continues to flow and the call state of the
dialog is maintained at the UA, at the SBCs collectively, and at the
far-end UA.

Now, the UA attempts a re-INVITE.  It sends the request to the first
address for the Record-Route URI, which is the common FQDN of the two
SBCs.  Since the UA has a TCP connection to this address, it uses this
TCP connection.  This TCP send fails immediately with an RST.

At this point, the UA should attempt to resend the request, and it
should attempt to contact both addresses for the SBCs -- a new TCP
connection to the first SBC tried as the TCP connection may have been
lost due to a transient error, and the second SBC should be contacted as
it is the other address for the FQDN.  Since the SBCs share
signaling/media state, the dialog can be updates via either of them, so
it doesn't matter which SBC the UA sends the re-INVITE to.

It seems that what happens is that the UA drops the call and then tries
to contact the second SBC.

I see that there are two possible situations, which require different
solutions:

1. If the Record-Route is the FQDN for both SBCs, the UA must not
conclude that the dialog is broken until it has attempted to contact
every address for the Record-Route FQDN and has failed to reach all of
the addresses.

2. If the Record-Route is a FQDN or address for the primary SBC, then
the UA is behaving correctly, since the dialog can only be updated
through that SBC.  In that case, the fix is to change the URI that is
put in the Record-Route by the SBC.

Dale


From nobody Wed Jan 27 19:07:18 2016
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Subject: [sipcore] First draft of draft-ietf-sipcore-dns-dual-stack-03
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I've made a first draft of draft-ietf-sipcore-dns-dual-stack-03.  This
message is to invite people to look at it.  I think that it addresses
all the substantial issues that are outstanding and all that I've
discovered, but I've not compared the new text with the mailing list
messages to check that I've covered everything.

This draft is based on the idea that it does not specify Happy Eyeballs
procedures for SIP, but it cleans up a number of matters in RFC 3263 so
that specifying Happy Eyeballs can be done clearly.

These URLs can be used to retrieve the XML and text versions:
http://svn.resiprocate.org/rep/ietf-drafts/worley/draft-ietf-sipcore-dns-dual-stack-03.xml
http://svn.resiprocate.org/rep/ietf-drafts/worley/draft-ietf-sipcore-dns-dual-stack-03.txt

A summary of the changes is given in section 9.1:

   9.1.  Changes from draft-ietf-sipcore-dns-dual-stack-02 to draft-ietf-
         sipcore-dns-dual-stack-03

   Updated wording, punctuation, and capitalization in various places.

   Clarified that this draft does not document Happy Eyeballs for SIP,
   but is preparatory for it.

   Attempted to use "update" for text that is definitively a change to
   the preexisting text and "clarify" for text that is a more clear
   statement of the (presumed) intention of the preexisting text.

   Removed normative words from section 1, the introduction.

   Introduced definition of "address records" (taken from RFC 2782, SRV
   records) to allow the definition to expand to include any new address
   families automatically.

   Relocated the text requiring a client to ignore addresses that it
   discovers in address families it does not support from section 4.2
   (which describes why the situation would arise) to section 4.1 (which
   describes how clients look up RRs).

   Clarified the interaction with RFC 6157 (source and destination
   address selection in IPv6) to specify what must have been intended:
   The major sort of the destinations is the ordering determined by
   priority/weight in the SRV records; the addresses derived from a
   single SRV record's target are minorly sorted based on RFC 6157.

   Removed editor's name from the acknowledgments list.

Dale


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From: "Penar, Russell" <Russell.Penar@centurylink.com>
To: "'Dale R. Worley'" <worley@ariadne.com>
Thread-Topic: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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That is very close to 100% correct.  The minor but significant delta is DNS=
 target 1 is two SBCs acting as one, in a primary/hot-standby configuration=
. DNS target 2, is a separate mated-pair of SBCs. So as the UA sees things =
(and in reality), the two SBCs returned in DNS do not share state for signa=
ling and/or media.

I'm in complete agreement with your quote just below, as I believe it start=
s to get at the heart of the issue I'm raising.

"Now, the UA attempts a re-INVITE.  It sends the request to the first addre=
ss for the Record-Route URI, which is the common FQDN of the two SBCs.  Sin=
ce the UA has a TCP connection to this address, it uses this TCP connection=
.  This TCP send fails immediately with an RST.

At this point, the UA should attempt to resend the request, ... a new TCP c=
onnection to the first SBC tried as the TCP connection may have been lost d=
ue to a transient error.." // This last sentence is what I'm looking to enf=
orce, or at least suggest officially.  If existing RFCs already do so, I'd =
like to review with you and perhaps request to make the language clearer fo=
r UA manufacturers and consumer (for reference to said vendors).

Regards,
Russ

-----Original Message-----
From: Dale R. Worley [mailto:worley@ariadne.com]
Sent: Wednesday, January 27, 2016 6:42 PM
To: Penar, Russell
Cc: sipcore@ietf.org
Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (b=
ehavior upon receipt of tcp rst)

"Penar, Russell" <Russell.Penar@centurylink.com> writes:
> RP - It's a very specific case, where the single primary target in SRV
> consists of 2 SBCs in a Highly Available active/hot-standby
> configuration.  We want to steer UA makers away from taking a simple
> TCP RST as a reason to move to the next target.  In doing so, they
> will force active calls to be dropped.  I see value in trying a TCP
> connection with the primary before moving to the next and dropping
> calls.

If I understand you correctly, the UA is configured with a single FQDN, whi=
ch maps through SRV records to two SBCs.  The SBCs share signaling and medi=
a state.

Initially, the UA creates a call by contacting one SBC -- from the SIP poin=
t of view, it establishes a TCP connection to one of the addresses associat=
ed with the FQDN for the proxy to which it is to send calls.

In the failure case, the first SBC goes down.  Since the SBCs share media s=
tate, the media continues to flow and the call state of the dialog is maint=
ained at the UA, at the SBCs collectively, and at the far-end UA.

Now, the UA attempts a re-INVITE.  It sends the request to the first addres=
s for the Record-Route URI, which is the common FQDN of the two SBCs.  Sinc=
e the UA has a TCP connection to this address, it uses this TCP connection.=
  This TCP send fails immediately with an RST.

At this point, the UA should attempt to resend the request, and it should a=
ttempt to contact both addresses for the SBCs -- a new TCP connection to th=
e first SBC tried as the TCP connection may have been lost due to a transie=
nt error, and the second SBC should be contacted as it is the other address=
 for the FQDN.  Since the SBCs share signaling/media state, the dialog can =
be updates via either of them, so it doesn't matter which SBC the UA sends =
the re-INVITE to.

It seems that what happens is that the UA drops the call and then tries to =
contact the second SBC.

I see that there are two possible situations, which require different
solutions:

1. If the Record-Route is the FQDN for both SBCs, the UA must not conclude =
that the dialog is broken until it has attempted to contact every address f=
or the Record-Route FQDN and has failed to reach all of the addresses.

2. If the Record-Route is a FQDN or address for the primary SBC, then the U=
A is behaving correctly, since the dialog can only be updated through that =
SBC.  In that case, the fix is to change the URI that is put in the Record-=
Route by the SBC.

Dale

This communication is the property of CenturyLink and may contain confident=
ial or privileged information. Unauthorized use of this communication is st=
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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"Penar, Russell" <Russell.Penar at centurylink.com> writes:
> That is very close to 100% correct.  The minor but significant delta
> is DNS target 1 is two SBCs acting as one, in a primary/hot-standby
> configuration. DNS target 2, is a separate mated-pair of SBCs. So as
> the UA sees things (and in reality), the two SBCs returned in DNS do
> not share state for signaling and/or media.  [trimmed]

OK, now I understand the configuration:  There are two *pairs* of SBCs,
each pair of which shares state, but the two pairs do not.

I think the situation is covered by four rules which are
well-established:

1.  If a device attempts to send a request to a TCP address, and the
device has a previously-existing TCP connection to the address, and the
attempt to send the request on that connection fails, the device must
not assume that the target is unreachable, because the TCP connection
might have failed due to a transient problem in the past that is now
resolved.  The device may only conclude that the address is unreachable
(at this time) if the attempt to create and use a new TCP connection
fails.

2.  A device may not conclude that a URI is unreachable unless attempts
to contact all of its addresses fail.

3.  A device may not conclude that the signaling path of an existing
call has failed unless the Record-Route/Contact URI is unreachable.

4.  An intermediate device should add to an INVITE a Record-Route URI
that routes to itself and all other intermediate devices that can be
used to update the state of the call.

Rule 4 is not a standard, but is known to be the best practice.  In your
case, when a call goes through an SBC, the Record-Route URI should be
one that routes to both SBCs in that pair, but not to the SBCs in the
other pair.

Rule 3 is not a standard, and a UA may tear down a call any time it
desires to.  But clearly, unless the signaling path becomes unusable,
the UA shouldn't tear down the call.  And in some usages, it might be a
better user experience to keep the call up even if the signaling path
fails -- the user may prefer to have his on-hold request fail and just
keep on talking.

Rule 2 is what RFC 3263 specifies.

Rule 1 is the core of your complaint, that the UA concludes that a TCP
target is unreachable just because an old TCP connection to it isn't
usable.  I don't know if rule 1 is specified anywhere, but this sort of
thing is a known problem in any situation when using a persistent
connection, and the proper recovery from it (attempting to reestablish
the connection) is well-known as the way to recover from it.

Of course, the UA my decide to attempt first to contact the other
address for the Record-Route URI, that is, the second SBC of the pair,
on the grounds that the first address seems to be less reliable.  (The
UA has latitude in what order to contact the addresses for the URI.)  If
that is successful, by rules 2 and 3 it should never tear down the call,
since the UA has successfully contacted the Record-Route URI.

Dale


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From: worley@ariadne.com (Dale R. Worley)
To: Russell.Penar@centurylink.com
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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There seems to be another aspect of this:  Why does the UA send a request
along a TCP connection that should have been torn down already?

When the SBC shuts down, the best practice would be for it to close (TCP
FIN) all TCP connections it has open.  In worse practice, it would
force-close (TCP RST) all TCP connections it had open.

In either case, the FIN or RST reaches the UA, which causes its TCP
stack to mark the TCP connection closed.

When the UA next wants to send a request to the SBC, it should notice
that the TCP connection is already torn down and not attempt to send the
request on it.

Dale


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From: "Penar, Russell" <Russell.Penar@centurylink.com>
To: "'Dale R. Worley'" <worley@ariadne.com>
Thread-Topic: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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Date: Fri, 29 Jan 2016 22:36:47 +0000
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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Agreed in principle.  In practice, the SBC (multiple vendors) dies quietly =
and only when the mate is contacted is the RST sent to the UA.

While we agree on these informal best practices (including retrying primary=
 target upon receipt of RST), it seems vendors aren't getting the memo and =
ideally these practices/behaviors would be included in an RFC for easy refe=
rence and less back and forth with vendors who chose wrong up front.

Regards,
Russ

-----Original Message-----
From: Dale R. Worley [mailto:worley@ariadne.com]
Sent: Friday, January 29, 2016 9:07 AM
To: Penar, Russell
Cc: sipcore@ietf.org
Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (b=
ehavior upon receipt of tcp rst)

There seems to be another aspect of this:  Why does the UA send a request a=
long a TCP connection that should have been torn down already?

When the SBC shuts down, the best practice would be for it to close (TCP
FIN) all TCP connections it has open.  In worse practice, it would force-cl=
ose (TCP RST) all TCP connections it had open.

In either case, the FIN or RST reaches the UA, which causes its TCP stack t=
o mark the TCP connection closed.

When the UA next wants to send a request to the SBC, it should notice that =
the TCP connection is already torn down and not attempt to send the request=
 on it.

Dale

This communication is the property of CenturyLink and may contain confident=
ial or privileged information. Unauthorized use of this communication is st=
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Subject: Re: [sipcore] sip/tcp with primary & secondary outbound proxies (behavior upon receipt of tcp rst)
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"Penar, Russell" <Russell.Penar@centurylink.com> writes:
> While we agree on these informal best practices (including retrying
> primary target upon receipt of RST), it seems vendors aren't getting
> the memo and ideally these practices/behaviors would be included in an
> RFC for easy reference and less back and forth with vendors who chose
> wrong up front.

You could write an RFC covering what you want to see:
http://www.ietf.org/edu/documents/81IETF-New-Work.pdf

Dale

