Requirements for Multiple Address of Record (AOR) Reachability Information in the Session Initiation Protocol (SIP)
RFC 5947
| Document | Type | RFC - Informational (September 2010; No errata) | |
|---|---|---|---|
| Authors | John Elwell , Hadriel Kaplan | ||
| Last updated | 2015-10-14 | ||
| Replaces | draft-elwell-martini-reqs | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html pdf htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | (None) | |
| Document shepherd | No shepherd assigned | ||
| IESG | IESG state | RFC 5947 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Gonzalo Camarillo | ||
| Send notices to | (None) | ||
Internet Engineering Task Force (IETF) J. Elwell
Request for Comments: 5947 Siemens Enterprise Communications
Category: Informational H. Kaplan
ISSN: 2070-1721 Acme Packet
September 2010
Requirements for Multiple Address of Record (AOR) Reachability
Information in the Session Initiation Protocol (SIP)
Abstract
This document states requirements for a standardized SIP registration
mechanism for multiple addresses of record (AORs), the mechanism
being suitable for deployment by SIP service providers on a large
scale in support of small to medium sized Private Branch Exchanges
(PBXs). The requirements are for a solution that can, as a minimum,
support AORs based on E.164 numbers.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc5947.
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Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction ....................................................3
2. Terminology .....................................................3
3. Problem Statement ...............................................4
3.1. Issues with the REGISTER Transaction .......................5
3.1.1. Mishandling by SIP-Aware Middleboxes ................5
3.1.2. REGISTER Response Growth ............................6
3.1.3. Illegal "Wildcard" Syntax ...........................6
3.2. Issues with Routing Inbound Requests to the SIP-PBX ........6
3.2.1. Loss of Target Information ..........................6
3.2.2. Inconsistent Placement of Target URI
Information in Inbound Requests .....................6
3.2.3. Request-URI Misrouting ..............................7
3.3. Policy-Related Issues ......................................7
3.3.1. Authorization Policy Mismatches .....................7
3.3.2. PAI or PPI URI Mismatches ...........................7
4. Requirements ....................................................8
5. Desirables .....................................................10
6. Non-Requirements ...............................................11
7. Security considerations ........................................11
8. Acknowledgements ...............................................12
9. References .....................................................12
9.1. Normative References ......................................12
9.2. Informative References ....................................12
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1. Introduction
The Session Initiation Protocol (SIP) [RFC3261], together with its
extensions, supports multiple means of obtaining the connection
information necessary to deliver out-of-dialog SIP requests to their
intended targets. When a SIP proxy needs to send a request to a
target address of record (AOR) within its domain, it can use a
location service to obtain the registered Contact Universal Resource
Identifiers (URIs), together with any associated path information
[RFC3327], and build a route set to reach each target user agent
(UA). The SIP REGISTER method can be used to register Contact URIs
and path information. SIP-outbound [RFC5626] enhances this mechanism
to cater to UAs behind Network Address Translators (NATs) and
firewalls. When an entity needs to send a request to a target for
which it is not authoritative, the entity can follow [RFC3263]
procedures for using the Domain Name System (DNS) to obtain the next-
hop connection information.
In practice, many small- and medium-sized businesses use a SIP
Private Branch Exchange (SIP-PBX) that is authoritative for tens or
hundreds of SIP AORs. This SIP-PBX acts as a registrar/proxy for
these AORs for users hosted by the SIP-PBX. A SIP Service Provider
(SSP) provides SIP peering/trunking capability to the SIP-PBX. The
SIP-PBX needs to be reachable from the SSP so that the SSP can handle
inbound out-of-dialog SIP requests targeted at these AORs, routing
these requests to the SIP-PBX for onward delivery to registered UAs.
Experience has shown that existing mechanisms are not always
sufficient to support SIP-PBXs for small/medium businesses. In
particular, RFC 3263 procedures are generally inappropriate, except
for some larger SIP-PBXs. In current deployments, mechanisms for the
dynamic provision of reachability information based on the SIP
REGISTER method are commonly used. However, these mechanisms vary in
detail, leading to interoperability issues between SIP-PBXs and SSPs,
and the need for equipment to support different variants. A more
detailed statement of the problem is given in Section 3.
This document states requirements for a standardized SIP registration
mechanism for multiple AORs, the mechanism being suitable for
deployment by SSPs on a large scale in support of small- to medium-
sized SIP-PBXs. The requirements are for a solution that can, as a
minimum, support AORs based on E.164 numbers.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
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The terms address of record (AOR), proxy, REGISTER, registrar,
request, response, and user agent (UA) are to be interpreted as
described in [RFC3261].
3. Problem Statement
A number of other standards organizations have addressed the issue of
a SIP-PBX registering with its SSP, notably ETSI [ETSI_TS_182025] and
3rd Generation Partnership Project (3GPP) [3GPP.24.229]. Also,
various SSPs have produced proprietary specifications for use with
their own offerings. The reader is encouraged to review the
documents produced by those organizations.
A short summary of the general concept is as follows. The figure
below illustrates the scope of the problem.
+----+
| UA |----+
+----+ | - - - - - - - - - - - - - -
| : SCOPE OF PROBLEM :
+----+ | : :
| UA |--+ | +---------+ +---------+
+----+ | | | | | |
| +------| | | |
+----+ +--------| SIP-PBX |------------------| SSP |
| UA |-----------| | | |
+----+ | | | |
+---------+ +---------+
: :
----------------> : ------------------> :
UAs register with : SIP-PBX registers with :
SIP-PBX on behalf of : SSP once on behalf of :
individual AORs : multiple AORs :
: :
: <------------------ :
<---------------- : SSP delivers inbound :
SIP-PBX forwards : requests to SIP-PBX :
inbound requests : :
to appropriate UAs : :
- - - - - - - - - - - - - -
In virtually all models, the SIP-PBX generates a SIP REGISTER request
using a mutually agreed-upon SIP AOR -- typically based on the SIP-
PBX's main attendant-/reception-desk number. The AOR is often in the
domain of the SSP, and both the To and From URIs used for the
REGISTER request identify that AOR. In all respects, it appears on
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the wire as a "normal" SIP REGISTER request, as if from a typical
user's UA. However, it generally implicitly registers other AORs
associated with the SIP-PBX.
For both 3GPP and ETSI mechanisms, the 200 OK response to the
REGISTER request, sent after a successful authentication challenge,
contains a P-Associated-URI (PAU) [RFC3455] header field listing the
other SIP URIs or TEL URIs (i.e., phone numbers) of the SIP-PBX,
which are implicitly registered AORs. The registered reachability
information from the REGISTER request will be used to reach not only
the single explicitly registered AOR but also each of the implicitly
registered AORs. In order to reduce the number of PAU entries, a
"wildcard" syntax model is defined [3GPP.23.003], which uses a
regular expression syntax in the user field of the URI to express
multiple AORs in a compressed manner.
For routing requests for any of the explicitly or implicitly
registered AORs from the SSP to the SIP-PBX, the Request-URI is
typically replaced with the registered Contact URI. In the case of
3GPP and ETSI, the SSP has the option of using loose routing instead,
and inserting the registered Contact URI as a loose route Route
header field value, while leaving the Request-URI alone. This
decision is made based upon manually provisioned information in the
registrar's database (i.e., the Home Subscriber Server (HSS)).
3.1. Issues with the REGISTER Transaction
3.1.1. Mishandling by SIP-Aware Middleboxes
None of the currently available mechanisms indicate that the REGISTER
request or response is any different from a "normal" REGISTER request
or response. This has caused issues when SIP-aware middleboxes
between the SIP-PBX and the registrar serve both SIP-PBXs and normal
UAs yet need to apply different policies to the two cases.
Furthermore, some middleboxes expect the registrar to follow normal
[RFC3261] procedures of Request-URI replacement with the registered
Contact URI for routing subsequent requests to the SIP-PBX. If the
registrar adopts a different practice for requests to SIP-PBXs, this
can cause the middlebox to fail to route such requests correctly,
because there is no indication that the registration was any
different.
Lastly, lack of an indication of implicit registration makes
troubleshooting more difficult because the on-the-wire messages are
indistinguishable from "normal" registrations. Note that even the
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existence of a PAU header field in the response does not indicate
that implicit registration for a SIP-PBX has occurred, since the PAU
header field is also used for normal UAs with multiple identities.
3.1.2. REGISTER Response Growth
If a SIP-PBX represents many AORs, the PAU list in the response can
grow the SIP message size beyond the limits for UDP.
3.1.3. Illegal "Wildcard" Syntax
The current syntax for "wildcarded" PAUs is illegal for TEL URIs,
based on the ABNF rules for TEL URIs in [RFC3966].
3.2. Issues with Routing Inbound Requests to the SIP-PBX
3.2.1. Loss of Target Information
If the proxy-registrar follows [RFC3261] for registration resolution
of requests targeted at one of the SIP-PBX's AORs, and thus replaces
the Request-URI with the registered Contact URI, it is not clear
which AOR is the intended target of the request. The To URI, for
example, may not contain the intended target AOR if the request was
forwarded/retargeted prior to reaching the proxy-registrar. Some
middleboxes between the registrar and SIP-PBX will overwrite the
Request-URI specifically to try to fix this issue. In some cases, a
P-Called-Party-ID header field [RFC3455] will contain the intended
target AOR; and in some cases, the History-Info header field
[RFC4244] will contain it. The SIP-PBX needs to know where to look
to find the required information and, in the case of History-Info,
needs to identify the particular element containing the required
information.
3.2.2. Inconsistent Placement of Target URI Information in Inbound
Requests
Even when all information needed by the SIP-PBX is provided, in some
deployments, inbound SIP requests from the SSP have the registered
SIP-PBX Contact URI in the Request-URI, whereas in other deployments
inbound SIP requests have the intended target SIP-PBX user (AOR) in
the Request-URI and the Contact URI in the Route header field. There
are other variants, too. Interoperability problems arise when a SIP-
PBX designed or configured for one variant attempts to interwork with
an SSP designed or configured for another variant.
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3.2.3. Request-URI Misrouting
Although many SIP-PBXs support registration with an SSP, they do not
consider themselves authoritative for the explicitly or implicitly
registered AORs if the domain portion still identifies the SSP's
domain. They expect the domain portion to be their own IP Address,
Fully Qualified Domain Name (FQDN), or domain. Currently,
middleboxes have to fix that issue.
3.3. Policy-Related Issues
The following are largely policy matters for the SSP, but it should
be noted the policies described below will not work in some
situations. A mechanism for solving the SIP-PBX registration problem
will not solve these policy issues directly, although when specifying
the mechanism, the opportunity can be taken to highlight the impact
of such policies.
3.3.1. Authorization Policy Mismatches
Many SSPs perform a first-order level of authorization for requests
from the SIP-PBX by checking the URI in the From, P-Asserted-
Identity (PAI), or P-Preferred-Identity (PPI) [RFC3325] header field
for one matching either an explicitly or implicitly registered AOR
for the same Contact URI and/or Layer-3 IP Address. However, some
SIP-PBXs change the Contact URI they use for non-REGISTER requests to
be different from the one they explicitly registered. For example,
they change the user portion of the Contact URI, or even the host
portion. This is particularly true for a SIP-PBX operating as a
proxy and forwarding the Contact URI from the UA behind the SIP-PBX
(the SIP-PBX typically being identified in a Record-Route header
field), rather than acting as a Back-to-Back User Agent (B2BUA) and
substituting its own Contact URI. This can cause an SSP to fail to
find an AOR corresponding to the Contact URI for non-REGISTER
requests, resulting in the SSP rejecting such requests or asserting
its own PAI value, rather than asserting a value based on received
header fields.
3.3.2. PAI or PPI URI Mismatches
Some SSPs expect the PAI or PPI URI in SIP requests received from the
SIP-PBX to match one of the explicitly or implicitly registered AORs,
whereas some SIP-PBXs generate the URIs using their local IP Address,
hostname, or domain name. Some SSPs expect the PAI or PPI URI in SIP
requests received from the SIP-PBX to be the explicitly registered
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AOR only, as it is the main billing number, instead of the implicitly
registered AOR of the calling party. In either case, this can result
in the SSP rejecting requests with values that do not match, or
asserting its own PAI value.
Again, these are policy matters for the SSP, but drawbacks should be
noted. For example, rejection of requests can rule out requests from
sources beyond the SIP-PBX (e.g., calls forwarded by the SIP-PBX),
unless the SIP-PBX changes the PAI or PPI URI to a value acceptable
to the SSP (in which case it will no longer identify the calling
user). If the SSP changes the PAI or PPI URI, again the request will
fail to identify the calling user.
4. Requirements
The following are requirements of the mechanism.
REQ1: The mechanism MUST allow a SIP-PBX to enter into a trunking
arrangement with an SSP whereby the two parties have agreed
on a set of telephone numbers assigned to the SIP-PBX.
REQ2: The mechanism MUST allow a set of assigned telephone numbers
to comprise E.164 numbers, which can be in contiguous ranges,
discrete, or in any combination of the two.
REQ3: The mechanism MUST allow a SIP-PBX to register reachability
information with its SSP, in order to enable the SSP to route
to the SIP-PBX inbound requests targeted at assigned
telephone numbers.
REQ4: The mechanism MUST allow UAs attached to a SIP-PBX to
register with the SIP-PBX for AORs based on assigned
telephone numbers, in order to receive requests targeted at
those telephone numbers, without needing to involve the SSP
in the registration process.
REQ5: The mechanism MUST allow a SIP-PBX to handle requests
originating at its own UAs and targeted at its assigned
telephone numbers, without routing those requests to the SSP.
REQ6: The mechanism MUST allow a SIP-PBX to receive requests to its
assigned telephone numbers originating outside the SIP-PBX
and arriving via the SSP, so that the SIP-PBX can route those
requests onwards to its UAs, as it would for internal
requests to those telephone numbers.
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REQ7: The mechanism MUST provide a means whereby a SIP-PBX knows at
which of its assigned telephone numbers an inbound request
from its SSP is targeted.
REQ8: The mechanism MUST provide a means of avoiding problems due
to one side using the mechanism and the other side not.
In other words, the mechanism is required to avoid the
situation where one side believes it is using the
mechanism and the other side believes it is not, e.g., the
SIP-PBX believes it is performing the registration of
multiple telephone numbers, but the SSP believes a single
AOR is being registered.
REQ9: The mechanism MUST observe SIP backwards compatibility
principles.
In other words, the mechanism is required to provide a
graceful means of recovery or fall-back if either side
does not support the mechanism. For example, this might
involve the use of an option tag.
REQ10: The mechanism MUST work in the presence of a sequence of
intermediate SIP entities on the SIP-PBX-to-SSP interface
(i.e., between the SIP-PBX and the SSP's domain proxy), where
those intermediate SIP entities indicated during registration
a need to be on the path of inbound requests to the SIP-PBX.
These intermediate SIP entities can be edge proxies,
session border controllers, etc.
REQ11: The mechanism MUST work when a SIP-PBX obtains its IP address
dynamically.
REQ12: The mechanism MUST work without requiring the SIP-PBX to have
a domain name or the ability to publish its domain name in
the DNS.
REQ13: For a given SIP-PBX and its SSP, there MUST be no impact on
other domains, which are expected to be able to use normal
RFC 3263 procedures to route requests, including requests
needing to be routed via the SSP in order to reach the SIP-
PBX.
REQ14: The mechanism MUST be able to operate over a transport that
provides end-to-end integrity protection and confidentiality
between the SIP-PBX and the SSP, e.g., using Transport Layer
Security (TLS) as specified in [RFC3261].
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REQ15: The mechanism MUST support authentication of the SIP-PBX by
the SSP and vice versa, e.g., using SIP digest authentication
plus TLS server authentication as specified in [RFC3261].
REQ16: The mechanism MUST allow the SIP-PBX to provide its UAs with
public or temporary Globally Routable UA URIs (GRUUs)
[RFC5627].
REQ17: The mechanism MUST work over any existing transport specified
for SIP, including UDP.
REQ18: Documentation MUST give guidance or warnings about how
authorization policies may be affected by the mechanism, to
address the problems described in Section 3.3.
REQ19: The mechanism MUST be extensible to allow a set of assigned
telephone numbers to comprise local numbers as specified in
[RFC3966], which can be in contiguous ranges, discrete, or in
any combination of the two.
REQ20: The mechanism MUST be extensible to allow a set of
arbitrarily assigned SIP URI's as specified in [RFC3261], as
opposed to just telephone numbers, without requiring a
complete change of mechanism as compared to that used for
telephone numbers.
5. Desirables
The following are desirable properties of the mechanism.
In addition to the desirables below, the general aim is to require
only relatively modest changes to a substantial population of
existing SSP and SIP-PBX implementations, in order to encourage a
fast market adoption of the standardized mechanism. Ease of market
adoption is paramount here. Many SIP-PBXs and SSPs have implemented
mechanisms based on the REGISTER method, and the need for substantial
changes to those implementations will discourage convergence on a
single standard in the foreseeable future.
DES1: The mechanism SHOULD allow an SSP to exploit its mechanisms
for providing SIP service to normal UAs in order to provide a
SIP trunking service to SIP-PBXs.
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DES2: The mechanism SHOULD scale to SIP-PBXs of several thousand
assigned telephone numbers.
This will probably preclude any mechanism involving a
separate REGISTER transaction per assigned telephone
number.
In practice, the mechanism is more likely to be used on
SIP-PBXs with up to a few hundred telephone numbers, but it
is impossible to give a precise limit, and hence the desire
to be able to support several thousand.
DES3: The mechanism SHOULD scale to support several thousand SIP-
PBXs on a single SSP.
6. Non-Requirements
The means by which a third domain can route a request to the SSP for
onward delivery to the SIP-PBX is outside the scope of this work.
This is related to REQ13, which requires normal routing based on RFC
3263 be used.
Provisioning is outside the scope of this work. In particular, an
SSP will need to assign a set of telephone numbers to a SIP-PBX, and
a SIP-PBX will need to be aware of the set of assigned numbers and
allocate those numbers to its users. Automated means for a SIP-PBX
to obtain, from its SSP, the set of assigned telephone numbers is
considered to be a provisioning topic.
7. Security considerations
The security of signaling between the SIP-PBX and the SSP is
important. Some of the requirements above already address this.
In particular, it is important that an entity acting as a SIP-PBX
cannot register with an SSP and receive inbound requests to which it
is not entitled. The SSP is assumed to have procedures for ensuring
that a SIP-PBX is entitled to use a set of E.164 telephone numbers
prior to entering into agreement with that SIP-PBX for using those
telephone numbers with this mechanism. Furthermore, by
authenticating the SIP-PBX when it provides reachability information,
the SSP can be sure that it delivers inbound requests only to the
correct destination.
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8. Acknowledgements
The contents of the document have been compiled from extensive
discussions within the MARTINI WG, the individuals concerned being
too numerous to mention.
9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263,
June 2002.
9.2. Informative References
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
Extensions to the Session Initiation Protocol (SIP) for
Asserted Identity within Trusted Networks", RFC 3325,
November 2002.
[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol
(SIP) Extension Header Field for Registering Non-Adjacent
Contacts", RFC 3327, December 2002.
[RFC3455] Garcia-Martin, M., Henrikson, E., and D. Mills, "Private
Header (P-Header) Extensions to the Session Initiation
Protocol (SIP) for the 3rd-Generation Partnership Project
(3GPP)", RFC 3455, January 2003.
[RFC3966] Schulzrinne, H., "The tel URI for Telephone Numbers",
RFC 3966, December 2004.
[RFC4244] Barnes, M., "An Extension to the Session Initiation
Protocol (SIP) for Request History Information", RFC 4244,
November 2005.
[RFC5626] Jennings, C., Mahy, R., and F. Audet, "Managing Client-
Initiated Connections in the Session Initiation Protocol
(SIP)", RFC 5626, October 2009.
Elwell & Kaplan Informational [Page 12]
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[RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUUs) in the Session Initiation Protocol
(SIP)", RFC 5627, October 2009.
[3GPP.23.003]
3GPP, "Numbering, addressing and identification", 3GPP
TS 23.003 3.15.0, October 2006.
[3GPP.24.229]
3GPP, "IP multimedia call control protocol based on
Session Initiation Protocol (SIP) and Session Description
Protocol (SDP); Stage 3", 3GPP TS 24.229 10.0.0,
June 2010.
[ETSI_TS_182025]
ETSI TS 182 025, "Telecommunications and Internet
converged Services and Protocols for Advanced Networking
(TISPAN); Business trunking; Architecture and functional
description".
Authors' Addresses
John Elwell
Siemens Enterprise Communications
EMail: john.elwell@siemens-enterprise.com
Hadriel Kaplan
Acme Packet
71 Third Ave.
Burlington, MA 01803
USA
EMail: hkaplan@acmepacket.com
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