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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session PEERing for Multimedia INTerconnect Working Group of the IETF.


	Title           : SPEERMINT Security Threats and Suggested Countermeasures
	Author(s)       : S. Niccolini, et al.
	Filename        : draft-ietf-speermint-voipthreats-01.txt
	Pages           : 24
	Date            : 2009-07-13

This memo presents the different security threats related to
SPEERMINT, classifying them into threats to the Lookup Function
(LUF), Location Routing Function (LRF), to the Signaling Function
(SF) and to the Media Function (MF).  The different instances of the
threats are briefly introduced inside the classification.  Finally,
the existing security solutions for SIP and RTP/RTCP are presented to
describe the countermeasures currently available for such threats.
Security requirements for SPEERMINT can be found in
draft-ietf-speermint-requirements.  The objective of this document is
to identify and enumerate SPEERMINT-specific threat vectors and to
give guidance for implementers on selecting appropriate
countermeasures.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-speermint-voipthreats-01.txt

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ftp://ftp.ietf.org/internet-drafts/

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From Jan.Seedorf@nw.neclab.eu  Mon Jul 13 06:38:21 2009
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From: "Jan Seedorf" <Jan.Seedorf@nw.neclab.eu>
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Subject: [Speermint] new version of draft-ietf-speermint-voipthreats submitted
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Dear all,

We have just submitted the -01 version of =
draft-ietf-speermint-voipthreats. We addressed all the comments received =
at IETF-73 and on the list, and made some editorial changes. We are =
looking forward to comments and most importantly to feedback from real =
implementers of SSP-peering (I know there are some of the subscribed to =
this list :)). Let us know if some threats or countermeasures are =
missing or if something is not clearly written in the document.

Kind regards,
Jan

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Jan Seedorf
Research Scientist
NEC Europe Ltd., NEC Laboratories Europe, Network Division=09
Kurfuerstenanlage 36, D-69115 Heidelberg
Tel.     +49 (0)6221 4342-221
Fax:     +49 (0)6221 4342-155
e-mail:  jan.seedorf@nw.neclab.eu=20
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NEC Europe Limited Registered Office: NEC House,
1 Victoria Road, London W3 6BL Registered in England 2832014=20


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> This message is in MIME format. Since your mail reader does not understand
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Has been posted at http://www.ietf.org/proceedings/75/agenda/speermint.txt

Our WG session will take place on Friday of IETF week.

Do not hesitate to suggest agenda changes.

Regards
Jason

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From Jan.Seedorf@nw.neclab.eu  Wed Jul 22 02:05:54 2009
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From: "Jan Seedorf" <Jan.Seedorf@nw.neclab.eu>
To: <speermint@ietf.org>
Subject: [Speermint] IETF-75 Agenda: link to draft-ietf-speermint-voipthreats is not working
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Dear chairs,

When checking out the agenda for IETF-75, I found that the link for our =
draft "draft-ietf-speermint-voipthreats" is not working. Can you please =
replace it with this one:

http://tools.ietf.org/html/draft-ietf-speermint-voipthreats-01

Thanks,
Jan

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Jan Seedorf
Research Scientist
NEC Europe Ltd., NEC Laboratories Europe, Network Division=09
Kurfuerstenanlage 36, D-69115 Heidelberg
Tel.     +49 (0)6221 4342-221
Fax:     +49 (0)6221 4342-155
e-mail:  jan.seedorf@nw.neclab.eu=20
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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
NEC Europe Limited Registered Office: NEC House,
1 Victoria Road, London W3 6BL Registered in England 2832014=20


From john.elwell@siemens-enterprise.com  Fri Jul 31 06:00:58 2009
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Date: Fri, 31 Jul 2009 14:00:34 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: speermint@ietf.org
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Thread-Topic: draft-hancock-sip-interconnect-guideline - offerless INVITE requests
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Subject: [Speermint] draft-hancock-sip-interconnect-guideline - offerless INVITE requests
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One of the points mentioned during this afternoon's presentation was
restricting the scope to basic 2-way call establishment **not** using
3PCC, which implies no support for offerless INVITE requests. I tried to
make a comment via Jabber, but it was not taken to the mic.

This would be an unacceptable limitation of the scope. Enterprises make
substantial use of 3PCC techniques, including third party call
establishment. This necessitates an offerless INVITE request to one of
the parties.

For example, I use my web browser to establish a call to you. The device
I wish to use is my cellphone, so it sends an offerless INVITE request
to, which will be routed to a PSTN gateway. The PSTN gateway returns an
SDP offer that can be used in the INVITE request towards you (who might
also be reachable via a PSTN gateway, or directly via SIP). If the
initial offerless INVITE request is carried inter-domain before reaching
the PSTN gateway, it needs to get through.

Note that this concern applies not just to enterprise-to-enterprise (if
the scope of the document were to be extended to include this), but to
any interconnect interface.

John

From gunnar.hellstrom@omnitor.se  Fri Jul 31 07:10:39 2009
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <speermint@ietf.org>
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Date: Fri, 31 Jul 2009 16:10:20 +0200
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Cc: d.hancock@cablelabs.com
Subject: [Speermint] Missing media in draft-hancock-sip-interconnect-guidelines-01.txt
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I find it very sad to limit the scope of this specification to audio only,
and ask to widen the scope at least to real-time conversational sessions
including real-time text, video and audio.

One motivation is that the work with emergency service calls in ECRIT
require a possibility to use real-time text, video and audio in emergency
calls. Such calls might need to be routed over a sip interconnection, and
thus need to get full support for its media. ( emergency calls are within
the scope list of this draft. )

There are a number of other valid reasons for this request.

Hoping that it can accepted, I give here a first set of edit proposals to
bring the draft up to proper multimedia status.

****************************************
In 1.1  change from
"1.1.  Scope

   The document focuses on the interworking procedures required to
   support basic telephone service, including the following
   capabilities:

   o  On-net to on-net calls
"
To
"
1.1.  Scope

   The document focuses on the interworking procedures required to
   support basic real-time conversational service, including the following
   capabilities:

   o  Video
   o  Real-time text
   o  On-net to on-net calls
"

In 5.1.1.  change from
"
5.1.1.  SDP Requirements

   SIP entities involved in session peering MUST support the SDP
   requirements defined in [RFC4566].  A SIP entity involved in session
   peering MUST include only one media (m=) descriptor in an SDP offer
   to a peer network.  If a SIP entity involved in session peering
   receives an SDP offer containing multiple media descriptors, it
   SHOULD act on the first audio descriptor with a non-zero port.
"
To
"
5.1.1.  SDP Requirements

   SIP entities involved in session peering MUST support the SDP
   requirements defined in [RFC4566].  A SIP entity involved in session
   peering MUST include only one media (m=) descriptor per desired media
   stream in an SDP offer to a peer network.  

   If a SIP entity involved in session peering receives an SDP offer
   containing multiple media descriptors, it MUST act on the media
   descriptors and include all of them in the same order in the
   response, including non-zero ports and zero ports for the offered
   media according to its capabilities as specified in [RFC 3264]
   Offer - Answer model. An offered session MUST NOT be rejected
   because it offers more media than the responding entity can handle.
"

These wew just the apparent initial edits after rapid browsing of the draft.
There is likely more to do to complete the topic.
-

/Gunnar


-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
On Behalf Of Internet-Drafts@ietf.org
Sent: Monday, July 13, 2009 9:00 PM
To: i-d-announce@ietf.org
Subject: I-D Action:draft-hancock-sip-interconnect-guidelines-01.txt 

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : draft-hancock-sip-interconnect-guidelines-01
	Author(s)       : D. Hancock, D. Malas
	Filename        : draft-hancock-sip-interconnect-guidelines-01.txt
	Pages           : 25
	Date            : 2009-07-13

As Session Initiation Protocol (SIP) peering becomes more widely accepted by
service providers the need to define an interconnect guideline becomes of
greater value.  This document takes into consideration the SIP and commonly
used SIP extensions, and it defines a fundamental set of requirements for
SIP Service Providers
(SSPs) to implement within their signaling functions (SFs) or Signaling Path
Border Elements (SBEs) for peering.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-hancock-sip-interconnect-guideline
s-01.txt

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

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implementation to automatically retrieve the ASCII version of the
Internet-Draft.


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Date: Sat, 1 Aug 2009 16:57:58 +1000
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From: Barry Dingle <btdingle@gmail.com>
To: speermint@ietf.org, Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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Cc: d.hancock@cablelabs.com
Subject: Re: [Speermint] Missing media in draft-hancock-sip-interconnect-guidelines-01.txt
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--001485f1a1bc922efa04700f0ec2
Content-Type: text/plain; charset=UTF-8
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Hi,

I support the comments of Gunnar Hellstrom regarding the need for  strong
support of real-time text and video conversational services in addition to
voice in the SIP-Interconnect Guidelines.

RFC5012 (ECRIT Requirements) in Section 5 has a requirement Re4 which
states:

Re4.  Multi-mode communication:  IP-based emergency calls MUST support
multiple communication modes, including, for example,
      audio, video, and text.

      Motivation: Within the PSTN, voice and text telephony (often
called TTY or text-phone in North America) are the only commonly
      supported media.  Emergency calling must support a variety of
media.  Such media should include voice, conversational text (RFC
      4103 [RFC4103]), instant messaging, and video.

SPEERMINT must support these services at the Interconnect point as part of
support of access to Emergency Services if it is to be taken seriously as an
alternative Interconnect for public telecommunications based on IP.

/Barry

Barry Dingle
Fellow of University of Melbourne, Electrical and Electronic Eng., Australia

Fixed - +61(0)3-9725-3937    Mob - +61(0)41-911-7578
> Supporter of Linux + Open Source software


On Sat, Aug 1, 2009 at 12:10 AM, Gunnar Hellstrom <
gunnar.hellstrom@omnitor.se> wrote:

> I find it very sad to limit the scope of this specification to audio only,
> and ask to widen the scope at least to real-time conversational sessions
> including real-time text, video and audio.
>
> One motivation is that the work with emergency service calls in ECRIT
> require a possibility to use real-time text, video and audio in emergency
> calls. Such calls might need to be routed over a sip interconnection, and
> thus need to get full support for its media. ( emergency calls are within
> the scope list of this draft. )
>
> There are a number of other valid reasons for this request.
>
> Hoping that it can accepted, I give here a first set of edit proposals to
> bring the draft up to proper multimedia status.
>
> ****************************************
> In 1.1  change from
> "1.1.  Scope
>
>   The document focuses on the interworking procedures required to
>   support basic telephone service, including the following
>   capabilities:
>
>   o  On-net to on-net calls
> "
> To
> "
> 1.1.  Scope
>
>   The document focuses on the interworking procedures required to
>   support basic real-time conversational service, including the following
>   capabilities:
>
>   o  Video
>   o  Real-time text
>   o  On-net to on-net calls
> "
>
> In 5.1.1.  change from
> "
> 5.1.1.  SDP Requirements
>
>   SIP entities involved in session peering MUST support the SDP
>   requirements defined in [RFC4566].  A SIP entity involved in session
>   peering MUST include only one media (m=) descriptor in an SDP offer
>   to a peer network.  If a SIP entity involved in session peering
>   receives an SDP offer containing multiple media descriptors, it
>   SHOULD act on the first audio descriptor with a non-zero port.
> "
> To
> "
> 5.1.1.  SDP Requirements
>
>   SIP entities involved in session peering MUST support the SDP
>   requirements defined in [RFC4566].  A SIP entity involved in session
>   peering MUST include only one media (m=) descriptor per desired media
>   stream in an SDP offer to a peer network.
>
>   If a SIP entity involved in session peering receives an SDP offer
>   containing multiple media descriptors, it MUST act on the media
>   descriptors and include all of them in the same order in the
>   response, including non-zero ports and zero ports for the offered
>   media according to its capabilities as specified in [RFC 3264]
>   Offer - Answer model. An offered session MUST NOT be rejected
>   because it offers more media than the responding entity can handle.
> "
>
> These wew just the apparent initial edits after rapid browsing of the
> draft.
> There is likely more to do to complete the topic.
> -
>
> /Gunnar
>
>
> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
> On Behalf Of Internet-Drafts@ietf.org
> Sent: Monday, July 13, 2009 9:00 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-hancock-sip-interconnect-guidelines-01.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>        Title           : draft-hancock-sip-interconnect-guidelines-01
>        Author(s)       : D. Hancock, D. Malas
>        Filename        : draft-hancock-sip-interconnect-guidelines-01.txt
>        Pages           : 25
>        Date            : 2009-07-13
>
> As Session Initiation Protocol (SIP) peering becomes more widely accepted
> by
> service providers the need to define an interconnect guideline becomes of
> greater value.  This document takes into consideration the SIP and commonly
> used SIP extensions, and it defines a fundamental set of requirements for
> SIP Service Providers
> (SSPs) to implement within their signaling functions (SFs) or Signaling
> Path
> Border Elements (SBEs) for peering.
>
> A URL for this Internet-Draft is:
>
> http://www.ietf.org/internet-drafts/draft-hancock-sip-interconnect-guideline
> s-01.txt<http://www.ietf.org/internet-drafts/draft-hancock-sip-interconnect-guideline%0As-01.txt>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www.ietf.org/mailman/listinfo/speermint
>

--001485f1a1bc922efa04700f0ec2
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Hi,<br><br>I support the comments of Gunnar Hellstrom regarding the need fo=
r=C2=A0 strong  support of real-time text and video conversational services=
 in addition to voice in the SIP-Interconnect Guidelines. <br><br>RFC5012 (=
ECRIT Requirements) in Section 5 has a requirement Re4 which states:<br>
<br><pre>Re4.  Multi-mode communication:  IP-based emergency calls MUST sup=
port multiple communication modes, including, for example,<br>      audio, =
video, and text.<br><br>      Motivation: Within the PSTN, voice and text t=
elephony (often called TTY or text-phone in North America) are the only com=
monly<br>
      supported media.  Emergency calling must support a variety of media. =
 Such media should include voice, conversational text (RFC<br>      4103 [R=
FC4103]), instant messaging, and video.<br></pre>SPEERMINT must support the=
se services at the Interconnect point as part of support of access to Emerg=
ency Services if it is to be taken seriously as an alternative Interconnect=
 for public telecommunications based on IP.<br>
<br>/Barry<br><br>Barry Dingle<br>Fellow of University of Melbourne, Electr=
ical and Electronic Eng., Australia <br>Fixed - +61(0)3-9725-3937 =C2=A0 =
=C2=A0Mob - +61(0)41-911-7578<br>&gt; Supporter of Linux + Open Source soft=
ware<br>

<br><br><div class=3D"gmail_quote">On Sat, Aug 1, 2009 at 12:10 AM, Gunnar =
Hellstrom <span dir=3D"ltr">&lt;<a href=3D"mailto:gunnar.hellstrom@omnitor.=
se">gunnar.hellstrom@omnitor.se</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin=
: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I find it very sad to limit the scope of this specification to audio only,<=
br>
and ask to widen the scope at least to real-time conversational sessions<br=
>
including real-time text, video and audio.<br>
<br>
One motivation is that the work with emergency service calls in ECRIT<br>
require a possibility to use real-time text, video and audio in emergency<b=
r>
calls. Such calls might need to be routed over a sip interconnection, and<b=
r>
thus need to get full support for its media. ( emergency calls are within<b=
r>
the scope list of this draft. )<br>
<br>
There are a number of other valid reasons for this request.<br>
<br>
Hoping that it can accepted, I give here a first set of edit proposals to<b=
r>
bring the draft up to proper multimedia status.<br>
<br>
****************************************<br>
In 1.1 =C2=A0change from<br>
&quot;1.1. =C2=A0Scope<br>
<br>
 =C2=A0 The document focuses on the interworking procedures required to<br>
 =C2=A0 support basic telephone service, including the following<br>
 =C2=A0 capabilities:<br>
<br>
 =C2=A0 o =C2=A0On-net to on-net calls<br>
&quot;<br>
To<br>
&quot;<br>
1.1. =C2=A0Scope<br>
<br>
 =C2=A0 The document focuses on the interworking procedures required to<br>
 =C2=A0 support basic real-time conversational service, including the follo=
wing<br>
 =C2=A0 capabilities:<br>
<br>
 =C2=A0 o =C2=A0Video<br>
 =C2=A0 o =C2=A0Real-time text<br>
 =C2=A0 o =C2=A0On-net to on-net calls<br>
&quot;<br>
<br>
In 5.1.1. =C2=A0change from<br>
&quot;<br>
5.1.1. =C2=A0SDP Requirements<br>
<br>
 =C2=A0 SIP entities involved in session peering MUST support the SDP<br>
 =C2=A0 requirements defined in [RFC4566]. =C2=A0A SIP entity involved in s=
ession<br>
 =C2=A0 peering MUST include only one media (m=3D) descriptor in an SDP off=
er<br>
 =C2=A0 to a peer network. =C2=A0If a SIP entity involved in session peerin=
g<br>
 =C2=A0 receives an SDP offer containing multiple media descriptors, it<br>
 =C2=A0 SHOULD act on the first audio descriptor with a non-zero port.<br>
&quot;<br>
To<br>
&quot;<br>
5.1.1. =C2=A0SDP Requirements<br>
<br>
 =C2=A0 SIP entities involved in session peering MUST support the SDP<br>
 =C2=A0 requirements defined in [RFC4566]. =C2=A0A SIP entity involved in s=
ession<br>
 =C2=A0 peering MUST include only one media (m=3D) descriptor per desired m=
edia<br>
 =C2=A0 stream in an SDP offer to a peer network.<br>
<br>
 =C2=A0 If a SIP entity involved in session peering receives an SDP offer<b=
r>
 =C2=A0 containing multiple media descriptors, it MUST act on the media<br>
 =C2=A0 descriptors and include all of them in the same order in the<br>
 =C2=A0 response, including non-zero ports and zero ports for the offered<b=
r>
 =C2=A0 media according to its capabilities as specified in [RFC 3264]<br>
 =C2=A0 Offer - Answer model. An offered session MUST NOT be rejected<br>
 =C2=A0 because it offers more media than the responding entity can handle.=
<br>
&quot;<br>
<br>
These wew just the apparent initial edits after rapid browsing of the draft=
.<br>
There is likely more to do to complete the topic.<br>
-<br>
<br>
/Gunnar<br>
<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:i-d-announce-bounces@ietf.org">i-d-announce-bounces=
@ietf.org</a> [mailto:<a href=3D"mailto:i-d-announce-bounces@ietf.org">i-d-=
announce-bounces@ietf.org</a>]<br>
On Behalf Of <a href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ie=
tf.org</a><br>
Sent: Monday, July 13, 2009 9:00 PM<br>
To: <a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><br>
Subject: I-D Action:draft-hancock-sip-interconnect-guidelines-01.txt<br>
<br>
A New Internet-Draft is available from the on-line Internet-Drafts<br>
directories.<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : draf=
t-hancock-sip-interconnect-guidelines-01<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Author(s) =C2=A0 =C2=A0 =C2=A0 : D. Hancock, D.=
 Malas<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draft-han=
cock-sip-interconnect-guidelines-01.txt<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 25<b=
r>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 2009-07-13<br>
<br>
As Session Initiation Protocol (SIP) peering becomes more widely accepted b=
y<br>
service providers the need to define an interconnect guideline becomes of<b=
r>
greater value. =C2=A0This document takes into consideration the SIP and com=
monly<br>
used SIP extensions, and it defines a fundamental set of requirements for<b=
r>
SIP Service Providers<br>
(SSPs) to implement within their signaling functions (SFs) or Signaling Pat=
h<br>
Border Elements (SBEs) for peering.<br>
<br>
A URL for this Internet-Draft is:<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-hancock-sip-interconne=
ct-guideline%0As-01.txt" target=3D"_blank">http://www.ietf.org/internet-dra=
fts/draft-hancock-sip-interconnect-guideline<br>
s-01.txt</a><br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
Below is the data which will enable a MIME compliant mail reader<br>
implementation to automatically retrieve the ASCII version of the<br>
Internet-Draft.<br>
<br>
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