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IETF RFC 6140
Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP)
Last modified on Thursday, March 3rd, 2011
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Internet Engineering Task Force (IETF) A.B. Roach
Request for Comments: 6140 Tekelec
Updates: 3680 March 2011
Category: Standards Track
ISSN: 2070-1721
Registration for Multiple Phone Numbers
in the Session Initiation Protocol (SIP)
 Abstract
This document defines a mechanism by which a Session Initiation
Protocol (SIP) server acting as a traditional Private Branch Exchange
(PBX) can register with a SIP Service Provider (SSP) to receive phone
calls for SIP User Agents (UAs). In order to function properly, this
mechanism requires that each of the Addresses of Record (AORs)
registered in bulk map to a unique set of contacts. This requirement
is satisfied by AORs representing phone numbers regardless of the
domain, since phone numbers are fully qualified and globally unique.
This document therefore focuses on this use case.
Status of This Memo
This is an Internet Standards Track document.
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). Further information on
Internet Standards is available in 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/RFC 6140.
Copyright Notice
Copyright (c) 2011 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
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RFC 6140 Globally Identifiable Number Routing March 2011
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. Constraints .....................................................3
3. Terminology and Conventions .....................................4
4. Mechanism Overview ..............................................5
5. Registering for Multiple Phone Numbers ..........................5
5.1. SIP-PBX Behavior ...........................................5
5.2. Registrar Behavior .........................................6
5.3. SIP URI "user" Parameter Handling ..........................8
6. SSP Processing of Inbound Requests ..............................8
7. Interaction with Other Mechanisms ...............................9
7.1. Globally Routable User Agent URIs (GRUU) ...................9
7.1.1. Public GRUUs ........................................9
7.1.2. Temporary GRUUs ....................................11
7.2. Registration Event Package ................................16
7.2.1. SIP-PBX Aggregate Registration State ...............16
7.2.2. Individual AOR Registration State ..................16
7.3. Client-Initiated (Outbound) Connections ...................18
7.4. Non-Adjacent Contact Registration (Path) and
Service-Route Discovery ...................................19
8. Examples .......................................................20
8.1. Usage Scenario: Basic Registration ........................20
8.2. Usage Scenario: Using Path to Control Request URI .........22
9. IANA Considerations ............................................24
9.1. New SIP Option Tag ........................................24
9.2. New SIP URI Parameters ....................................25
9.2.1. 'bnc' SIP URI Parameter ............................25
9.2.2. 'sg' SIP URI Parameter .............................25
9.3. New SIP Header Field Parameter ............................25
10. Security Considerations .......................................25
11. Acknowledgements ..............................................28
12. References ....................................................28
12.1. Normative References .....................................28
12.2. Informative References ...................................29
Appendix A. Requirements Analysis .................................31
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1. Introduction
The Session Initiation Protocol (SIP) is an application-layer control
(signaling) protocol for creating, modifying, and terminating
sessions with one or more participants. One of SIP's primary
functions is providing rendezvous between users. By design, these
rendezvous have been provided through a combination of the server
look-up procedures defined in RFC 3263 [4] and the registrar
procedures described in RFC 3261 [3].
The intention of the original protocol design was that any user's AOR
(Address of Record) would be handled by the authority indicated by
the hostport portion of the AOR. The users would register individual
reachability information with this authority, which would then route
incoming requests accordingly.
In actual deployments, some SIP servers have been deployed in
architectures that, for various reasons, have requirements to provide
dynamic routing information for large blocks of AORs, where all of
the AORs in the block were to be handled by the same server. For
purposes of efficiency, many of these deployments do not wish to
maintain separate registrations for each of the AORs in the block.
Thus, an alternate mechanism to provide dynamic routing information
for blocks of AORs is desirable.
Although the use of SIP REGISTER request messages to update
reachability information for multiple users simultaneously is
somewhat beyond the original semantics defined for REGISTER requests
by RFC 3261 [3], this approach has seen significant deployment in
certain environments. In particular, deployments in which small to
medium SIP-PBX servers are addressed using E.164 numbers have used
this mechanism to avoid the need to maintain DNS entries or static IP
addresses for the SIP-PBX servers.
In recognition of the momentum that REGISTER-based approaches have
seen in deployments, this document defines a REGISTER-based approach.
Since E.164-addressed UAs are very common today in SIP-PBX
environments, and since SIP URIs in which the user portion is an
E.164 number are always globally unique, regardless of the domain,
this document focuses on registration of SIP URIs in which the user
portion is an E.164 number.
2. Constraints
Within the problem space that has been established for this work,
several constraints shape our solution. These are defined in the
MARTINI requirements document [22] and are analyzed in Appendix A.
In terms of impact to the solution at hand, the following two
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constraints have the most profound effect: (1) The SIP-PBX cannot be
assumed to be assigned a static IP address; and (2) No DNS entry can
be relied upon to consistently resolve to the IP address of the SIP-
PBX.
3. Terminology and Conventions
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 RFC 2119 [2].
Further, the term "SSP" is meant as an acronym for a "SIP Service
Provider," while the term "SIP-PBX" is used to indicate a SIP Private
Branch Exchange.
Indented portions of the document, such as this one, form non-
normative, explanatory sections of the document.
Although SIP is a text-based protocol, some of the examples in this
document cannot be unambiguously rendered without additional markup
due to the constraints placed on the formatting of RFCs. This
document uses the <allOneLine/> markup convention established in RFC
4475 [17] to avoid ambiguity and meet the RFC layout requirements.
For the sake of completeness, the text defining this markup (Section
2.1 of RFC 4475 [17]) is reproduced in its entirety below:
Several of these examples contain unfolded lines longer than 72
characters. These are captured between <allOneLine/> tags. The
single unfolded line is reconstructed by directly concatenating
all lines appearing between the tags (discarding any line feeds or
carriage returns). There will be no whitespace at the end of
lines. Any whitespace appearing at a fold-point will appear at
the beginning of a line.
The following represent the same string of bits:
Header-name: first value, reallylongsecondvalue, third value
<allOneLine>
Header-name: first value,
reallylongsecondvalue
, third value
</allOneLine>
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<allOneLine>
Header-name: first value,
reallylong
second
value,
third value
</allOneLine>
Note that this is NOT SIP header-line folding, where different
strings of bits have equivalent meaning.
4. Mechanism Overview
The overall mechanism is achieved using a REGISTER request with a
specially formatted Contact URI. This document also defines an
option tag that can be used to ensure that a registrar and any
intermediaries understand the mechanism described herein.
The Contact URI itself is tagged with a URI parameter to indicate
that it actually represents multiple phone-number-associated
contacts.
We also define some lightweight extensions to the Globally Routable
UA URIs (GRUU) mechanism defined by RFC 5627 [20] to allow the use of
public and temporary GRUUs assigned by the SSP.
Aside from these extensions, the REGISTER request itself is processed
by a registrar in the same way as normal registrations: by updating
its location service with additional AOR-to-Contact bindings.
Note that the list of AORs associated with a SIP-PBX is a matter of
local provisioning at the SSP and the SIP-PBX. The mechanism defined
in this document does not provide any means to detect or recover from
provisioning mismatches (although the registration event package can
be used as a standardized means for auditing such AORs; see
Section 7.2.1).
5. Registering for Multiple Phone Numbers
5.1. SIP-PBX Behavior
To register for multiple AORs, the SIP-PBX sends a REGISTER request
to the SSP. This REGISTER request varies from a typical REGISTER
request in two important ways. First, it MUST contain an option tag
of "gin" in both a "Require" header field and a "Proxy-Require"
header field. (The option tag "gin" is an acronym for "generate
implicit numbers".) Second, in at least one "Contact" header field,
it MUST include a Contact URI that contains the URI parameter "bnc"
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(which stands for "bulk number contact") and has no user portion
(hence no "@" symbol). A URI with a "bnc" parameter MUST NOT contain
a user portion. Except for the SIP URI "user" parameter, this URI
MAY contain any other parameters that the SIP-PBX desires. These
parameters will be echoed back by the SSP in any requests bound for
the SIP-PBX.
Because of the constraints discussed in Section 2, the host portion
of the Contact URI will generally contain an IP address, although
nothing in this mechanism enforces or relies upon that fact. If the
SIP-PBX operator chooses to maintain DNS entries that resolve to the
IP address of his SIP-PBX via RFC 3263 resolution procedures, then
this mechanism works just fine with domain names in the "Contact"
header field.
The "bnc" URI parameter indicates that special interpretation of the
Contact URI is necessary: instead of indicating the insertion of a
single Contact URI into the location service, it indicates that
multiple URIs (one for each associated AOR) should be inserted.
Any SIP-PBX implementing the registration mechanism defined in this
document MUST also support the path mechanism defined by RFC 3327
[10], and MUST include a 'path' option tag in the "Supported" header
field of the REGISTER request (which is a stronger requirement than
imposed by the path mechanism itself). This behavior is necessary
because proxies between the SIP-PBX and the registrar may need to
insert "Path" header field values in the REGISTER request for this
document's mechanism to function properly, and, per RFC 3327 [10],
they can only do so if the User Agent Client (UAC) inserted the
option tag in the "Supported" header field. In accordance with the
procedures defined in RFC 3327 [10], the SIP-PBX is allowed to ignore
the "Path" header fields returned in the REGISTER response.
5.2. Registrar Behavior
The registrar, upon receipt of a REGISTER request containing at least
one "Contact" header field with a "bnc" parameter, will use the value
in the "To" header field to identify the SIP-PBX for which
registration is being requested. It then authenticates the SIP-PBX
(e.g., using SIP digest authentication, mutual Transport Layer
Security (TLS) [18], or some other authentication mechanism). After
the SIP-PBX is authenticated, the registrar updates its location
service with a unique AOR-to-Contact mapping for each of the AORs
associated with the SIP-PBX. Semantically, each of these mappings
will be treated as a unique row in the location service. The actual
implementation may, of course, perform internal optimizations to
reduce the amount of memory used to store such information.
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For each of these unique rows, the AOR will be in the format that the
SSP expects to receive from external parties (e.g.,
"sip:+12145550102@ssp.example.com"). The corresponding contact will
be formed by adding to the REGISTER request's Contact URI a user
portion containing the fully qualified, E.164-formatted number
(including the preceding "+" symbol) and removing the "bnc"
parameter. Aside from the initial "+" symbol, this E.164-formatted
number MUST consist exclusively of digits from 0 through 9 and
explicitly MUST NOT contain any visual separator symbols (e.g., "-",
".", "(", or ")"). For example, if the "Contact" header field
contains the URI <sip:198.51.100.3:5060;bnc>, then the contact value
associated with the aforementioned AOR will be
<sip:+12145550102@198.51.100.3:5060>.
Although the SSP treats this registration as a number of discrete
rows for the purpose of re-targeting incoming requests, the renewal,
expiration, and removal of these rows is bound to the registered
contact. In particular, this means that REGISTER requests that
attempt to de-register a single AOR that has been implicitly
registered MUST NOT remove that AOR from the bulk registration. In
this circumstance, the registrar simply acts as if the UA attempted
to unregister a contact that wasn't actually registered (e.g., return
the list of presently registered contacts in a success response). A
further implication of this property is that an individual extension
that is implicitly registered may also be explicitly registered using
a normal, non-bulk registration (subject to SSP policy). If such a
registration exists, it is refreshed independently of the bulk
registration and is not removed when the bulk registration is
removed.
A registrar that receives a REGISTER request containing a Contact URI
with both a "bnc" parameter and a user portion MUST NOT send a 200-
class (Success) response. If no other error is applicable, the
registrar can use a 400 (Bad Request) response to indicate this error
condition.
Note that the preceding paragraph is talking about the user
portion of a URI:
sip:+12145550100@example.com
^^^^^^^^^^^^
A registrar compliant with this document MUST support the path
mechanism defined in RFC 3327 [10]. The rationale for support of
this mechanism is given in Section 5.1.
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Aside from the "bnc" parameter, all URI parameters present on the
Contact URI in the REGISTER request MUST be copied to the contact
value stored in the location service.
If the SSP servers perform processing based on User Agent
Capabilities (as defined in RFC 3840 [13]), they will treat any
feature tags present on a "Contact" header field with a "bnc"
parameter in its URI as applicable to all of the resulting AOR-to-
Contact mappings. Similarly, any option tags present on the REGISTER
request that indicate special handling for any subsequent requests
are also applicable to all of the AOR-to-Contact mappings.
5.3. SIP URI "user" Parameter Handling
This document does not modify the behavior specified in RFC 3261 [3]
for inclusion of the "user" parameter on Request URIs. However, to
avoid any ambiguity in handling at the SIP-PBX, the following
normative behavior is imposed on its interactions with the SSP.
When a SIP-PBX registers with an SSP using a Contact URI containing a
"bnc" parameter, that Contact URI MUST NOT include a "user"
parameter. A registrar that receives a REGISTER request containing a
Contact URI with both a "bnc" parameter and a "user" parameter MUST
NOT send a 200-class (success) response. If no other error is
applicable, the registrar can use a 400 (Bad Request) response to
indicate this error condition.
Note that the preceding paragraph is talking about the "user"
parameter of a URI:
sip:+12145550100@example.com;user=phone
^^^^^^^^^^
When a SIP-PBX receives a request from an SSP, and the Request URI
contains a user portion corresponding to an AOR registered using a
Contact URI containing a "bnc" parameter, then the SIP-PBX MUST NOT
reject the request (or otherwise cause the request to fail) due to
the absence, presence, or value of a "user" parameter on the Request
URI.
6. SSP Processing of Inbound Requests
In general, after processing the AOR-to-Contact mapping described in
the preceding section, the SSP proxy/registrar (or equivalent entity)
performs traditional proxy/registrar behavior, based on the mapping.
For any inbound SIP requests whose AOR indicates an E.164 number
assigned to one of the SSP's customers, this will generally involve
setting the target set to the registered contacts associated with
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that AOR and performing request forwarding as described in Section
16.6 of RFC 3261 [3]. An SSP using the mechanism defined in this
document MUST perform such processing for inbound INVITE requests and
SUBSCRIBE requests to the "reg" event package (see Section 7.2.2) and
SHOULD perform such processing for all other method types, including
unrecognized SIP methods.
7. Interaction with Other Mechanisms
The following sections describe the means by which this mechanism
interacts with relevant REGISTER-related extensions currently defined
by the IETF.
7.1. Globally Routable User Agent URIs (GRUU)
To enable advanced services to work with UAs behind a SIP-PBX, it is
important that the GRUU mechanism defined by RFC 5627 [20] work
correctly with the mechanism defined by this document -- that is,
that user agents served by the SIP-PBX can acquire and use GRUUs for
their own use.
7.1.1. Public GRUUs
Support of public GRUUs is OPTIONAL in SSPs and SIP-PBXes.
When a SIP-PBX registers a Bulk Number Contact (a contact with a
"bnc" parameter), and also invokes GRUU procedures for that contact
during registration, then the SSP will assign a public GRUU to the
SIP-PBX in the normal fashion. Because the URI being registered
contains a "bnc" parameter, the GRUU will also contain a "bnc"
parameter. In particular, this means that the GRUU will not contain
a user portion.
When a UA registers a contact with the SIP-PBX using GRUU procedures,
the SIP-PBX provides to the UA a public GRUU formed by adding an "sg"
parameter to the GRUU parameter it received from the SSP. This "sg"
parameter contains a disambiguation token that the SIP-PBX can use to
route inbound requests to the proper UA.
So, for example, when the SIP-PBX registers with the following
"Contact" header field:
Contact: <sip:198.51.100.3;bnc>;
+sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
the SSP may choose to respond with a "Contact" header field that
looks like this:
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<allOneLine>
Contact: <sip:198.51.100.3;bnc>;
pub-gruu="sip:ssp.example.com;bnc;gr=urn:
uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6";
+sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
;expires=7200
</allOneLine>
When its own UAs register using GRUU procedures, the SIP-PBX can then
add whatever device identifier it feels appropriate in an "sg"
parameter and present this value to its own UAs. For example, assume
the UA associated with the AOR "+12145550102" sent the following
"Contact" header field in its REGISTER request:
Contact: <sip:line-1@10.20.1.17>;
+sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>"
The SIP-PBX will add an "sg" parameter to the pub-gruu it received
from the SSP with a token that uniquely identifies the device
(possibly the URN itself; possibly some other identifier), insert a
user portion containing the fully qualified E.164 number associated
with the UA, and return the result to the UA as its public GRUU. The
resulting "Contact" header field sent from the SIP-PBX to the
registering UA would look something like this:
<allOneLine>
Contact: <sip:line-1@10.20.1.17>;
pub-gruu="sip:+12145550102@ssp.example.com;gr=urn:
uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6;sg=00:05:03:5e:70:a6";
+sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>"
;expires=3600
</allOneLine>
When an incoming request arrives at the SSP for a GRUU corresponding
to a bulk number contact ("bnc"), the SSP performs slightly different
processing for the GRUU than it would for a URI without a "bnc"
parameter. When the GRUU is re-targeted to the registered bulk
number contact, the SSP MUST copy the "sg" parameter from the GRUU to
the new target. The SIP-PBX can then use this "sg" parameter to
determine to which user agent the request should be routed. For
example, the first line of an INVITE request that has been re-
targeted to the SIP-PBX for the UA shown above would look like this:
INVITE sip:+12145550102@198.51.100.3;sg=00:05:03:5e:70:a6 SIP/2.0
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7.1.2. Temporary GRUUs
In order to provide support for privacy, the SSP SHOULD implement the
temporary GRUU mechanism described in this section. Reasons for not
doing so would include systems with an alternative privacy mechanism
that maintains the integrity of public GRUUs (i.e., if public GRUUs
are anonymized, then the anonymizer function would need to be capable
of providing -- as the anonymized URI -- a globally routable URI that
routes back only to the target identified by the original public
GRUU).
Temporary GRUUs are used to provide anonymity for the party creating
and sharing the GRUU. Being able to correlate two temporary GRUUs as
having originated from behind the same SIP-PBX violates this
principle of anonymity. Consequently, rather than relying upon a
single, invariant identifier for the SIP-PBX in its UA's temporary
GRUUs, we define a mechanism whereby the SSP provides the SIP-PBX
with sufficient information for the SIP-PBX to mint unique temporary
GRUUs. These GRUUs have the property that the SSP can correlate them
to the proper SIP-PBX, but no other party can do so. To achieve this
goal, we use a slight modification of the procedure described in
Appendix A.2 of RFC 5627 [20].
The SIP-PBX needs to be able to construct a temp-gruu in a way that
the SSP can decode. In order to ensure that the SSP can decode
GRUUs, we need to standardize the algorithm for creation of temp-
gruus at the SIP-PBX. This allows the SSP to reverse the algorithm
in order to identify the registration entry that corresponds to the
GRUU.
It is equally important that no party other than the SSP be capable
of decoding a temporary GRUU, including other SIP-PBXes serviced by
the SSP. To achieve this property, an SSP that supports temporary
GRUUs MUST create and store an asymmetric key pair: {K_e1,K_e2}.
K_e1 is kept secret by the SSP, while K_e2 is shared with the SIP-
PBXes via provisioning.
All base64 encoding discussed in the following sections MUST use the
character set and encoding defined in Section 4 of RFC 4648 [8],
except that any trailing "=" characters are discarded on encoding and
added as necessary to decode.
The following sections make use of the term "HMAC-SHA256-80" to
describe a particular Hashed Message Authentication Code (HMAC)
algorithm. In this document, HMAC-SHA256-80 is defined as the
application of the SHA-256 [24] secure hashing algorithm, truncating
the results to 80 bits by discarding the trailing (least-significant)
bits.
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7.1.2.1. Generation of "temp-gruu-cookie" by the SSP
An SSP that supports temporary GRUUs MUST include a "temp-gruu-
cookie" parameter on all "Contact" header fields containing a "bnc"
parameter in a 200-class REGISTER response. This "temp-gruu-cookie"
MUST have the following properties:
1. It can be used by the SSP to uniquely identify the registration
to which it corresponds.
2. It is encoded using base64. This allows the SIP-PBX to decode it
into as compact a form as possible for use in its calculations.
3. It is of a fixed length. This allows for its extraction once the
SIP-PBX has concatenated a distinguisher onto it.
4. The temp-gruu-cookie MUST NOT be forgeable by any party. In
other words, the SSP needs to be able to examine the cookie and
validate that it was generated by the SSP.
5. The temp-gruu-cookie MUST be invariant during the course of a
registration, including any refreshes to that registration. This
property is important, as it allows the SIP-PBX to examine the
temp-gruu-cookie to determine whether the temp-gruus it has
issued to its UAs are still valid.
The above properties can be met using the following algorithm, which
is non-normative. Implementors may chose to implement any algorithm
of their choosing for generation of the temp-gruu-cookie, as long as
it fulfills the five properties listed above.
The registrar maintains a counter, I. This counter is 48 bits
long and initialized to zero. This counter is persistently
stored, using a back-end database or similar technique. When the
registrar creates the first temporary GRUU for a particular SIP-
PBX and instance ID (as defined by [20]), the registrar notes the
current value of the counter, I_i, and increments the counter in
the database. The registrar then maps I_i to the contact and
instance ID using the database, a persistent hash-map, or similar
technology. If the registration expires such that there are no
longer any contacts with that particular instance ID bound to the
GRUU, the registrar removes the mapping. Similarly, if the
temporary GRUUs are invalidated due to a change in Call-ID, the
registrar removes the current mapping from I_i to the AOR and
instance ID, notes the current value of the counter I_j, and
stores a mapping from I_j to the contact containing a "bnc"
parameter and instance ID. Based on these rules, the hash-map
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will contain a single mapping for each contact containing a "bnc"
parameter and instance ID for which there is a currently valid
registration.
The registrar maintains a symmetric key SK_a, which is regenerated
every time the counter rolls over or is reset. When the counter
rolls over or is reset, the registrar remembers the old value of
SK_a for a while. To generate a temp-gruu-cookie, the registrar
computes:
SA = HMAC(SK_a, I_i)
temp-gruu-cookie = base64enc(I_i || SA)
where || denotes concatenation. "HMAC" represents any suitably
strong HMAC algorithm; see RFC 2104 [1] for a discussion of HMAC
algorithms. One suitable HMAC algorithm for this purpose is HMAC-
SHA256-80.
7.1.2.2. Generation of temp-gruu by the SIP-PBX
According to Section 3.2 of RFC 5627 [20], every registration refresh
generates a new temp-gruu that is valid for as long as the contact
remains registered. This property is both critical for the privacy
properties of temp-gruu and is expected by UAs that implement the
temp-gruu procedures. Nothing in this document should be construed
as changing this fundamental temp-gruu property in any way. SIP-
PBXes that implement temporary GRUUs MUST generate a new temp-gruu
according to the procedures in this section for every registration or
registration refresh from GRUU-supporting UAs attached to the SIP-
PBX.
Similarly, if the registration that a SIP-PBX has with its SSP
expires or is terminated, then the temp-gruu cookie it maintains with
the SSP will change. This change will invalidate all the temp-gruus
the SIP-PBX has issued to its UAs. If the SIP-PBX tracks this
information (e.g., to include <temp-gruu> elements in registration
event bodies, as described in RFC 5628 [9]), it can determine that
previously issued temp-gruus are invalid by observing a change in the
temp-gruu-cookie provided to it by the SSP.
A SIP-PBX that issues temporary GRUUs to its UAs MUST maintain an
HMAC key: PK_a. This value is used to validate that incoming GRUUs
were generated by the SIP-PBX.
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To generate a new temporary GRUU for use by its own UAs, the SIP-PBX
MUST generate a random distinguisher value: D. The length of this
value is up to implementors, but it MUST be long enough to prevent
collisions among all the temporary GRUUs issued by the SIP-PBX. A
size of 80 bits or longer is RECOMMENDED. See RFC 4086 [16] for
further considerations on the generation of random numbers in a
security context. After generating the distinguisher D, the SIP-PBX
MUST calculate:
M = base64dec(SSP-cookie) || D
E = RSA-Encrypt(K_e2, M)
PA = HMAC(PK_a, E)
Temp-Gruu-userpart = "tgruu." || base64(E) || "." || base64(PA)
where || denotes concatenation. "HMAC" represents any suitably
strong HMAC algorithm; see RFC 2104 [1] for a discussion of HMAC
algorithms. One suitable HMAC algorithm for this purpose is HMAC-
SHA256-80.
Finally, the SIP-PBX adds a "gr" parameter to the temporary GRUU that
can be used to uniquely identify the UA registration record to which
the GRUU corresponds. The means of generation of the "gr" parameter
are left to the implementor, as long as they satisfy the properties
of a GRUU as described in RFC 5627 [20].
One valid approach for generation of the "gr" parameter is
calculation of "E" and "A" as described in Appendix A.2 of RFC
5627 [20] and forming the "gr" parameter as:
gr = base64enc(E) || base64enc(A)
Using this procedure may result in a temporary GRUU returned to the
registering UA by the SIP-PBX that looks similar to this:
<allOneLine>
Contact: <sip:line-1@10.20.1.17>
;temp-gruu="sip:tgruu.MQyaRiLEd78RtaWkcP7N8Q.5qVbsasdo2pkKw@
ssp.example.com;gr=YZGSCjKD42ccxO08pA7HwAM4XNDIlMSL0HlA"
;+sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>"
;expires=3600
</allOneLine>
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7.1.2.3. Decoding of temp-gruu by the SSP
When the SSP proxy receives a request in which the user part begins
with "tgruu.", it extracts the remaining portion and splits it at the
"." character into E' and PA'. It discards PA'. It then computes E
by performing a base64 decode of E'. Next, it computes:
M = RSA-Decrypt(K_e1, E)
The SSP proxy extracts the fixed-length temp-gruu-cookie information
from the beginning of this M and discards the remainder (which will
be the distinguisher added by the SIP-PBX). It then validates this
temp-gruu-cookie. If valid, it uses it to locate the corresponding
SIP-PBX registration record and routes the message appropriately.
If the non-normative, exemplary algorithm described in
Section 7.1.2.1 is used to generate the temp-gruu-cookie, then
this identification is performed by splitting the temp-gruu-cookie
information into its 48-bit counter I and 80-bit HMAC. It
validates that the HMAC matches the counter I and then uses
counter I to locate the SIP-PBX registration record in its map.
If the counter has rolled over or reset, this computation is
performed with the current and previous SK_a.
7.1.2.4. Decoding of temp-gruu by the SIP-PBX
When the SIP-PBX receives a request in which the user part begins
with "tgruu.", it extracts the remaining portion and splits it at the
"." character into E' and PA'. It then computes E and PA by
performing a base64 decode of E' and PA', respectively. Next, it
computes:
PAc = HMAC(PK_a, E)
where HMAC is the HMAC algorithm used for the steps in
Section 7.1.2.2. If this computed value for PAc does not match the
value of PA extracted from the GRUU, then the GRUU is rejected as
invalid.
The SIP-PBX then uses the value of the "gr" parameter to locate the
UA registration to which the GRUU corresponds, and routes the message
accordingly.
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7.2. Registration Event Package
Neither the SSP nor the SIP-PBX is required to support the
registration event package defined by RFC 3680 [12]. However, if
they do support the registration event package, they MUST conform to
the behavior described in this section and its subsections.
As this mechanism inherently deals with REGISTER transaction
behavior, it is imperative to consider its impact on the registration
event package defined by RFC 3680 [12]. In practice, there will be
two main use cases for subscribing to registration data: learning
about the overall registration state for the SIP-PBX and learning
about the registration state for a single SIP-PBX AOR.
7.2.1. SIP-PBX Aggregate Registration State
If the SIP-PBX (or another interested and authorized party) wishes to
monitor or audit the registration state for all of the AORs currently
registered to that SIP-PBX, it can subscribe to the SIP registration
event package at the SIP-PBX's main URI -- that is, the URI used in
the "To" header field of the REGISTER request.
The NOTIFY messages for such a subscription will contain a body that
contains one record for each AOR associated with the SIP-PBX. The
AORs will be in the format expected to be received by the SSP (e.g.,
"sip:+12145550105@ssp.example.com"), and the contacts will correspond
to the mapped contact created by the registration (e.g.,
"sip:+12145550105@98.51.100.3").
In particular, the "bnc" parameter is forbidden from appearing in the
body of a reg-event NOTIFY request unless the subscriber has
indicated knowledge of the semantics of the "bnc" parameter. The
means for indicating this support are out of scope of this document.
Because the SSP does not necessarily know which GRUUs have been
issued by the SIP-PBX to its associated UAs, these records will not
generally contain the <temp-gruu> or <pub-gruu> elements defined in
RFC 5628 [9]. This information can be learned, if necessary, by
subscribing to the individual AOR registration state, as described in
Section 7.2.2.
7.2.2. Individual AOR Registration State
As described in Section 6, the SSP will generally re-target all
requests addressed to an AOR owned by a SIP-PBX to that SIP-PBX
according to the mapping established at registration time. Although
policy at the SSP may override this generally expected behavior,
proper behavior of the registration event package requires that all
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"reg" event SUBSCRIBE requests are processed by the SIP-PBX. As a
consequence, the requirements on an SSP for processing registration
event package SUBSCRIBE requests are not left to policy.
If the SSP receives a SUBSCRIBE request for the registration event
package with a Request URI that indicates an AOR registered via the
"Bulk Number Contact" mechanism defined in this document, then the
SSP MUST proxy that SUBSCRIBE to the SIP-PBX in the same way that it
would proxy an INVITE bound for that AOR, unless the SSP has and can
maintain a copy of complete, accurate, and up-to-date information
from the SIP-PBX (e.g., through an active back-end subscription).
If the Request URI in a SUBSCRIBE request for the registration event
package indicates a contact that is registered by more than one SIP-
PBX, then the SSP proxy will fork the SUBSCRIBE request to all the
applicable SIP-PBXes. Similarly, if the Request URI corresponds to a
contact that is both implicitly registered by a SIP-PBX and
explicitly registered directly with the SSP proxy, then the SSP proxy
will semantically fork the SUBSCRIBE request to the applicable SIP-
PBX or SIP-PBXes and to the registrar function (which will respond
with registration data corresponding to the explicit registrations at
the SSP). The forking in both of these cases can be avoided if the
SSP has and can maintain a copy of up-to-date information from the
PBXes.
Section 4.9 of RFC 3680 [12] indicates that "a subscriber MUST NOT
create multiple dialogs as a result of a single [registration event]
subscription request". Consequently, subscribers who are not aware
of the extension described by this document will accept only one
dialog in response to such requests. In the case described in the
preceding paragraph, this behavior will result in such clients
receiving accurate but incomplete information about the registration
state of an AOR. As an explicit change to the normative behavior of
RFC 3680, this document stipulates that subscribers to the
registration event package MAY create multiple dialogs as the result
of a single subscription request. This will allow subscribers to
create a complete view of an AOR's registration state.
Defining the behavior as described above is important, since the reg-
event subscriber is interested in finding out about the comprehensive
list of devices associated with the AOR. Only the SIP-PBX will have
authoritative access to this information. For example, if the user
has registered multiple UAs with differing capabilities, the SSP will
not know about the devices or their capabilities. By contrast, the
SIP-PBX will.
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If the SIP-PBX is not registered with the SSP when a registration
event subscription for a contact that would be implicitly registered
if the SIP-PBX were registered is received, then the SSP SHOULD
accept the subscription and indicate that the user is not currently
registered. Once the associated SIP-PBX is registered, the SSP
SHOULD use the subscription migration mechanism defined in RFC 3265
[5] to migrate the subscription to the SIP-PBX.
When a SIP-PBX receives a registration event subscription addressed
to an AOR that has been registered using the bulk registration
mechanism described in this document, then each resulting
registration information document SHOULD contain an 'aor' attribute
in its <registration/> element that corresponds to the AOR at the
SSP.
For example, consider a SIP-PBX that has registered with an SSP
that has a domain of "ssp.example.com". The SIP-PBX used a
Contact URI of "sip:198.51.100.3:5060;bnc". After such
registration is complete, a registration event subscription
arriving at the SSP with a Request URI of
"sip:+12145550102@ssp.example.com" will be re-targeted to the SIP-
PBX, with a Request URI of "sip:+12145550102@198.51.100.3:5060".
The resulting registration document created by the SIP-PBX would
contain a <registration/> element with an "aor" attribute of
"sip:+12145550102@ssp.example.com".
This behavior ensures that subscribers external to the system (and
unaware of GIN (generate implicit numbers) procedures) will be
able to find the relevant information in the registration document
(since they will be looking for the publicly visible AOR, not the
address used for sending information from the SSP to the SIP-PBX).
A SIP-PBX that supports both GRUU procedures and the registration
event packages SHOULD implement the extension defined in RFC 5628
[9].
7.3. Client-Initiated (Outbound) Connections
RFC 5626 [19] defines a mechanism that allows UAs to establish long-
lived TCP connections or UDP associations with a proxy in a way that
allows bidirectional traffic between the proxy and the UA. This
behavior is particularly important in the presence of NATs, and
whenever TLS [18] security is required. Neither the SSP nor the SIP-
PBX is required to support client-initiated connections.
Generally, the outbound mechanism works with the solution defined in
this document, without any modifications. Implementors should note
that the instance ID used between the SIP-PBX and the SSP's registrar
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identifies the SIP-PBX itself, and not any of the UAs registered with
the SIP-PBX. As a consequence, any attempts to use caller
preferences (defined in RFC 3841 [14]) to target a specific instance
are likely to fail. This shouldn't be an issue, as the preferred
mechanism for targeting specific instances of a user agent is GRUU
(see Section 7.1).
7.4. Non-Adjacent Contact Registration (Path) and Service-Route
Discovery
RFC 3327 [10] defines a means by which a registrar and its associated
proxy can be informed of a route that is to be used between the proxy
and the registered user agent. The scope of the route created by a
"Path" header field is contact specific; if an AOR has multiple
contacts associated with it, the routes associated with each contact
may be different from each other. Support for non-adjacent contact
registration is required in all SSPs and SIP-PBXes implementing the
multiple-AOR-registration protocol described in this document.
At registration time, any proxies between the user agent and the
registrar may add themselves to the "Path" header field. By doing
so, they request that any requests destined to the user agent as a
result of the associated registration include them as part of the
Route towards the user agent. Although the path mechanism does
deliver the final path value to the registering UA, UAs typically
ignore the value of the path.
To provide similar functionality in the opposite direction -- that
is, to establish a route for requests sent by a registering UA -- RFC
3608 [11] defines a means by which a UA can be informed of a route
that is to be used by the UA to route all outbound requests
associated with the AOR used in the registration. This information
is scoped to the AOR within the UA, and is not specific to the
contact (or contacts) in the REGISTER request. Support of service
route discovery is OPTIONAL in SSPs and SIP-PBXes.
The registrar unilaterally generates the values of the service route
using whatever local policy it wishes to apply. Although it is
common to use the "Path" and/or "Route" header field information in
the request in composing the service route, registrar behavior is not
constrained in any way that requires it to do so.
In considering the interaction between these mechanisms and the
registration of multiple AORs in a single request, implementors of
proxies, registrars, and intermediaries must keep in mind the
following issues, which stem from the fact that GIN effectively
registers multiple AORs and multiple contacts.
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First, all location service records that result from expanding a
single Contact URI containing a "bnc" parameter will necessarily
share a single path. Proxies will be unable to make policy decisions
on a contact-by-contact basis regarding whether to include themselves
in the path. Second, and similarly, all AORs on the SIP-PBX that are
registered with a common REGISTER request will be forced to share a
common service route.
One interesting technique that the path and service route mechanisms
enable is the inclusion of a token or cookie in the user portion of
the service route or path entries. This token or cookie may convey
information to proxies about the identity, capabilities, and/or
policies associated with the user. Since this information will be
shared among several AORs and several contacts when multiple AOR
registration is employed, care should be taken to ensure that doing
so is acceptable for all AORs and all contacts registered in a single
REGISTER request.
8. Examples
Note that the following examples elide any steps related to
authentication. This is done for the sake of clarity. Actual
deployments will need to provide a level of authentication
appropriate to their system.
8.1. Usage Scenario: Basic Registration
This example shows the message flows for a basic bulk REGISTER
transaction, followed by an INVITE addressed to one of the registered
UAs. Example messages are shown after the sequence diagram.
Internet SSP SIP-PBX
| | |
| |(1) REGISTER |
| |Contact:<sip:198.51.100.3;bnc> |
| |<--------------------------------|
| | |
| |(2) 200 OK |
| |-------------------------------->|
| | |
|(3) INVITE | |
|sip:+12145550105@ssp.example.com| |
|------------------------------->| |
| | |
| |(4) INVITE |
| |sip:+12145550105@198.51.100.3 |
| |-------------------------------->|
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(1) The SIP-PBX registers with the SSP for a range of AORs.
REGISTER sip:ssp.example.com SIP/2.0
Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7
Max-Forwards: 70
To: <sip:pbx@ssp.example.com>
From: <sip:pbx@ssp.example.com>;tag=a23589
Call-ID: 843817637684230@998sdasdh09
CSeq: 1826 REGISTER
Proxy-Require: gin
Require: gin
Supported: path
Contact: <sip:198.51.100.3:5060;bnc>
Expires: 7200
Content-Length: 0
(3) The SSP receives a request for an AOR assigned
to the SIP-PBX.
INVITE sip:+12145550105@ssp.example.com SIP/2.0
Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
Max-Forwards: 69
To: <sip:2145550105@some-other-place.example.net>
From: <sip:gsmith@example.org>;tag=456248
Call-ID: f7aecbfc374d557baf72d6352e1fbcd4
CSeq: 24762 INVITE
Contact: <sip:line-1@192.0.2.178:2081>
Content-Type: application/sdp
Content-Length: ...
<sdp body here>
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(4) The SSP re-targets the incoming request according to the
information received from the SIP-PBX at registration time.
INVITE sip:+12145550105@198.51.100.3 SIP/2.0
Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50
Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
Max-Forwards: 68
To: <sip:2145550105@some-other-place.example.net>
From: <sip:gsmith@example.org>;tag=456248
Call-ID: f7aecbfc374d557baf72d6352e1fbcd4
CSeq: 24762 INVITE
Contact: <sip:line-1@192.0.2.178:2081>
Content-Type: application/sdp
Content-Length: ...
<sdp body here>
8.2. Usage Scenario: Using Path to Control Request URI
This example shows a bulk REGISTER transaction with the SSP making
use of the "Path" header field extension [10]. This allows the SSP
to designate a domain on the incoming Request URI that does not
necessarily resolve to the SIP-PBX when the SSP applies RFC 3263
procedures to it.
Internet SSP SIP-PBX
| | |
| |(1) REGISTER |
| |Path:<sip:pbx@198.51.100.3;lr> |
| |Contact:<sip:pbx.example;bnc> |
| |<--------------------------------|
| | |
| |(2) 200 OK |
| |-------------------------------->|
| | |
|(3) INVITE | |
|sip:+12145550105@ssp.example.com| |
|------------------------------->| |
| | |
| |(4) INVITE |
| |sip:+12145550105@pbx.example |
| |Route:<sip:pbx@198.51.100.3;lr> |
| |-------------------------------->|
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(1) The SIP-PBX registers with the SSP for a range of AORs.
It includes the form of the URI it expects to receive in the
Request URI in its "Contact" header field, and it includes
information that routes to the SIP-PBX in the "Path" header
field.
REGISTER sip:ssp.example.com SIP/2.0
Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7
Max-Forwards: 70
To: <sip:pbx@ssp.example.com>
From: <sip:pbx@ssp.example.com>;tag=a23589
Call-ID: 326983936836068@998sdasdh09
CSeq: 1826 REGISTER
Proxy-Require: gin
Require: gin
Supported: path
Path: <sip:pbx@198.51.100.3:5060;lr>
Contact: <sip:pbx.example;bnc>
Expires: 7200
Content-Length: 0
(3) The SSP receives a request for an AOR assigned
to the SIP-PBX.
INVITE sip:+12145550105@ssp.example.com SIP/2.0
Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
Max-Forwards: 69
To: <sip:2145550105@some-other-place.example.net>
From: <sip:gsmith@example.org>;tag=456248
Call-ID: 7ca24b9679ffe9aff87036a105e30d9b
CSeq: 24762 INVITE
Contact: <sip:line-1@192.0.2.178:2081>
Content-Type: application/sdp
Content-Length: ...
<sdp body here>
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(4) The SSP re-targets the incoming request according to the
information received from the SIP-PBX at registration time.
Per the normal processing associated with "Path", it
will insert the "Path" value indicated by the SIP-PBX at
registration time in a "Route" header field, and
set the Request URI to the registered contact.
INVITE sip:+12145550105@pbx.example SIP/2.0
Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50
Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
Route: <sip:pbx@198.51.100.3:5060;lr>
Max-Forwards: 68
To: <sip:2145550105@some-other-place.example.net>
From: <sip:gsmith@example.org>;tag=456248
Call-ID: 7ca24b9679ffe9aff87036a105e30d9b
CSeq: 24762 INVITE
Contact: <sip:line-1@192.0.2.178:2081>
Content-Type: application/sdp
Content-Length: ...
<sdp body here>
9. IANA Considerations
This document registers a new SIP option tag to indicate support for
the mechanism it defines, two new SIP URI parameters, and a "Contact"
header field parameter. The process governing registration of these
protocol elements is outlined in RFC 5727 [21].
9.1. New SIP Option Tag
This section defines a new SIP option tag per the guidelines in
Section 27.1 of RFC 3261 [3].
Name: gin
Description: This option tag is used to identify the extension that
provides registration for Multiple Phone Numbers in SIP. When
present in a "Require" or "Proxy-Require" header field of a
REGISTER request, it indicates that support for this extension is
required of registrars and proxies, respectively, that are a party
to the registration transaction.
Reference: RFC 6140
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9.2. New SIP URI Parameters
This specification defines two new SIP URI parameters, as per the
registry created by RFC 3969 [7].
9.2.1. 'bnc' SIP URI Parameter
Parameter Name: bnc
Predefined Values: No (no values are allowed)
Reference: RFC 6140
9.2.2. 'sg' SIP URI Parameter
Parameter Name: sg
Predefined Values: No
Reference: RFC 6140
9.3. New SIP Header Field Parameter
This section defines a new SIP header field parameter per the
registry created by RFC 3968 [6].
Header field: Contact
Parameter name: temp-gruu-cookie
Predefined values: No
Reference: RFC 6140
10. Security Considerations
The change proposed for the mechanism described in this document
takes the unprecedented step of extending the previously defined
REGISTER method to apply to more than one AOR. In general, this kind
of change has the potential to cause problems at intermediaries --
such as proxies -- that are party to the REGISTER transaction. In
particular, such intermediaries may attempt to apply policy to the
user indicated in the "To" header field (i.e., the SIP-PBX's
identity), without any knowledge of the multiple AORs that are being
implicitly registered.
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The mechanism defined by this document solves this issue by adding an
option tag to a "Proxy-Require" header field in such REGISTER
requests. Proxies that are unaware of this mechanism will not
process the requests, preventing them from misapplying policy.
Proxies that process requests with this option tag are clearly aware
of the nature of the REGISTER request and can make reasonable policy
decisions.
As noted in Section 7.4, intermediaries need to take care if they use
a policy token in the path and service route mechanisms, as doing so
will cause them to apply the same policy to all users serviced by the
same SIP-PBX. This may frequently be the correct behavior, but
circumstances can arise in which differentiation of user policy is
required.
Section 7.4 also notes that these techniques use a token or cookie in
the "Path" and/or "Service-Route" header values, and that this value
will be shared among all AORs associated with a single registration.
Because this information will be visible to user agents under certain
conditions, proxy designers using this mechanism in conjunction with
the techniques described in this document need to take care that
doing so does not leak sensitive information.
One of the key properties of the outbound client connection mechanism
discussed in Section 7.3 is the assurance that a single connection is
associated with a single user and cannot be hijacked by other users.
With the mechanism defined in this document, such connections
necessarily become shared between users. However, the only entity in
a position to hijack calls as a consequence is the SIP-PBX itself.
Because the SIP-PBX acts as a registrar for all the potentially
affected users, it already has the ability to redirect any such
communications as it sees fit. In other words, the SIP-PBX must be
trusted to handle calls in an appropriate fashion, and the use of the
outbound connection mechanism introduces no additional
vulnerabilities.
The ability to learn the identity and registration state of every
user on the PBX (as described in Section 7.2.1) is invaluable for
diagnostic and administrative purposes. For example, this allows the
SIP-PBX to determine whether all its extensions are properly
registered with the SSP. However, this information can also be
highly sensitive, as many organizations may not wish to make their
entire list of phone numbers available to external entities.
Consequently, SSP servers are advised to use explicit (i.e., white-
list) and configurable policies regarding who can access this
information, with very conservative defaults (e.g., an empty access
list or an access list consisting only of the PBX itself).
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The procedure for the generation of temporary GRUUs requires the use
of an HMAC to detect any tampering that external entities may attempt
to perform on the contents of a temporary GRUU. The mention of HMAC-
SHA256-80 in Section 7.1.2 is intended solely as an example of a
suitable HMAC algorithm. Since all HMACs used in this document are
generated and consumed by the same entity, the choice of an actual
HMAC algorithm is entirely up to an implementation, provided that the
cryptographic properties are sufficient to prevent third parties from
spoofing GRUU-related information.
The procedure for the generation of temporary GRUUs also requires the
use of RSA keys. The selection of the proper key length for such
keys requires careful analysis, taking into consideration the current
and foreseeable speed of processing for the period of time during
which GRUUs must remain anonymous, as well as emerging cryptographic
analysis methods. The most recent guidance from RSA Laboratories
[25] suggests a key length of 2048 bits for data that needs
protection through the year 2030, and a length of 3072 bits
thereafter.
Similarly, implementors are warned to take precautionary measures to
prevent unauthorized disclosure of the private key used in GRUU
generation. Any such disclosure would result in the ability to
correlate temporary GRUUs to each other and, potentially, to their
associated PBXes.
Further, the use of RSA decryption when processing GRUUs received
from arbitrary parties can be exploited by Denial-of-Service (DoS)
attackers to amplify the impact of an attack: because of the presence
of a cryptographic operation in the processing of such messages, the
CPU load may be marginally higher when the attacker uses (valid or
invalid) temporary GRUUs in the messages employed by such an attack.
Normal DoS mitigation techniques, such as rate-limiting processing of
received messages, should help to reduce the impact of this issue as
well.
Finally, good security practices should be followed regarding the
duration an RSA key is used. For implementors, this means that
systems MUST include an easy way to update the public key provided to
the SIP-PBX. To avoid immediately invalidating all currently issued
temporary GRUUs, the SSP servers SHOULD keep the retired RSA key
around for a grace period before discarding it. If decryption fails
based on the new RSA key, then the SSP server can attempt to use the
retired key instead. By contrast, the SIP-PBXes MUST discard the
retired public key immediately and exclusively use the new public
key.
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RFC 6140 Globally Identifiable Number Routing March 2011
11. Acknowledgements
This document represents the hard work of many people in the IETF
MARTINI working group and the IETF RAI community as a whole.
Particular thanks are owed to John Elwell for his requirements
analysis of the mechanism described in this document, to Dean Willis
for his analysis of the interaction between this mechanism and the
"Path" and "Service-Route" extensions, to Cullen Jennings for his
analysis of the interaction between this mechanism and the SIP
Outbound extension, and to Richard Barnes for his detailed security
analysis of the GRUU construction algorithm. Thanks to Eric
Rescorla, whose text in the appendix of RFC 5627 was lifted directly
to provide substantial portions of Section 7.1.2. Finally, thanks to
Bernard Aboba, Francois Audet, Brian Carpenter, John Elwell, David
Hancock, Ted Hardie, Martien Huysmans, Cullen Jennings, Alan
Johnston, Hadriel Kaplan, Paul Kyzivat, and Radia Perlman for their
careful reviews of and constructive feedback on this document.
12. References
12.1. Normative References
[1] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-Hashing
for Message Authentication", RFC 2104, February 1997.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] 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.
[4] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
(SIP): Locating SIP Servers", RFC 3263, June 2002.
[5] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[6] Camarillo, G., "The Internet Assigned Number Authority (IANA)
Header Field Parameter Registry for the Session Initiation
Protocol (SIP)", BCP 98, RFC 3968, December 2004.
[7] Camarillo, G., "The Internet Assigned Number Authority (IANA)
Uniform Resource Identifier (URI) Parameter Registry for the
Session Initiation Protocol (SIP)", BCP 99, RFC 3969,
December 2004.
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[8] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings",
RFC 4648, October 2006.
[9] Kyzivat, P., "Registration Event Package Extension for Session
Initiation Protocol (SIP) Globally Routable User Agent URIs
(GRUUs)", RFC 5628, October 2009.
12.2. Informative References
[10] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Registering Non-Adjacent Contacts",
RFC 3327, December 2002.
[11] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Service Route Discovery During
Registration", RFC 3608, October 2003.
[12] Rosenberg, J., "A Session Initiation Protocol (SIP) Event
Package for Registrations", RFC 3680, March 2004.
[13] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating
User Agent Capabilities in the Session Initiation Protocol
(SIP)", RFC 3840, August 2004.
[14] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
Preferences for the Session Initiation Protocol (SIP)",
RFC 3841, August 2004.
[15] Schulzrinne, H., "The tel URI for Telephone Numbers", RFC 3966,
December 2004.
[16] Eastlake, D., Schiller, J., and S. Crocker, "Randomness
Requirements for Security", BCP 106, RFC 4086, June 2005.
[17] Sparks, R., Hawrylyshen, A., Johnston, A., Rosenberg, J., and
H. Schulzrinne, "Session Initiation Protocol (SIP) Torture Test
Messages", RFC 4475, May 2006.
[18] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS)
Protocol Version 1.2", RFC 5246, August 2008.
[19] Jennings, C., Mahy, R., and F. Audet, "Managing Client-
Initiated Connections in the Session Initiation Protocol
(SIP)", RFC 5626, October 2009.
[20] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUUs) in the Session Initiation Protocol (SIP)",
RFC 5627, October 2009.
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[21] Peterson, J., Jennings, C., and R. Sparks, "Change Process for
the Session Initiation Protocol (SIP) and the Real-time
Applications and Infrastructure Area", BCP 67, RFC 5727,
March 2010.
[22] Elwell, J. and H. Kaplan, "Requirements for Multiple Address of
Record (AOR) Reachability Information in the Session Initiation
Protocol (SIP)", RFC 5947, September 2010.
[23] Kaplan, H., "GIN with Literal AORs for SIP in SSPs (GLASS)",
Work in Progress, November 2010.
[24] National Institute of Standards and Technology, "Secure Hash
Standard (SHS)", FIPS PUB 180-3, October 2008, <http://
csrc.nist.gov/publications/fips/fips180-3/fips180-3_final.pdf>.
[25] Kaliski, B., "TWIRL and RSA Key Size", May 2003.
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RFC 6140 Globally Identifiable Number Routing March 2011
Appendix A. Requirements Analysis
The document "Requirements for Multiple Address of Record (AOR)
Reachability Information in the Session Initiation Protocol (SIP)"
[22] contains a list of requirements and desired properties for a
mechanism to register multiple AORs with a single SIP transaction.
This section evaluates those requirements against the mechanism
described in this document.
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.
The requirement is satisfied.
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.
The requirement is satisfied. The Direct Inward Dialing (DID)
numbers associated with a registration are established by
bilateral agreement between the SSP and the SIP-PBX; they are not
part of the mechanism described in this document.
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.
The requirement is satisfied.
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.
The requirement is satisfied; in the presumed architecture, SIP-
PBX UAs register with the SIP-PBX and require no interaction with
the SSP.
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.
The requirement is satisfied; SIP-PBXes may recognize their own
DID numbers and GRUUs, and perform on-SIP-PBX routing without
sending the requests to the SSP.
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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.
The requirement is satisfied
REQ7 - The mechanism MUST provide a means whereby a SIP-PBX knows
which of its assigned telephone numbers an inbound request from its
SSP is targeted at.
The requirement is satisfied. For ordinary calls and calls using
public GRUUs, the DID number is indicated in the user portion of
the Request URI. For calls using Temp GRUUs constructed with the
mechanism described in Section 7.1.2, the "gr" parameter provides
a correlation token the SIP-PBX can use to identify to which UA
the call should be routed.
REQ8 - The mechanism MUST provide a means of avoiding problems due to
one side using the mechanism and the other side not.
The requirement is satisfied through the 'gin' option tag and the
'bnc' Contact URI parameter.
REQ9 - The mechanism MUST observe SIP backwards compatibility
principles.
The requirement is satisfied through the 'gin' 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.
The requirement is satisfied through the use of the path mechanism
defined in RFC 3327 [10]
REQ11 - The mechanism MUST work when a SIP-PBX obtains its IP address
dynamically.
The requirement is satisfied by allowing the SIP-PBX to use an IP
address in the Bulk Number Contact URI contained in a REGISTER
"Contact" header field.
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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.
The requirement is satisfied by allowing the SIP-PBX to use an IP
address in the Bulk Number Contact URI contained in a REGISTER
"Contact" header field.
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.
The requirement is satisfied by allowing the domain name in the
Request URI used by external entities to resolve to the SSP's
servers via normal RFC 3263 resolution procedures.
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 TLS as specified in [3].
The requirement is satisfied; nothing in the proposed mechanism
prevents the use of TLS between the SSP and the SIP-PBX.
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 [3].
The requirement is satisfied; SIP-PBXes may employ either SIP
digest authentication or mutually authenticated TLS for
authentication purposes.
REQ16 - The mechanism MUST allow the SIP-PBX to provide its UAs with
public or temporary Globally Routable UA URIs (GRUUs) [20].
The requirement is satisfied via the mechanisms detailed in
Section 7.1.
REQ17 - The mechanism MUST work over any existing transport specified
for SIP, including UDP.
The requirement is satisfied to the extent that UDP can be used
for REGISTER requests in general. The application of certain
extensions and/or network topologies may exceed UDP MTU sizes, but
such issues arise both with and without the mechanism described in
this document. This document does not exacerbate such issues.
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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 [of RFC 5947].
These issues are addressed at length in Section 10, as well as
summarized in Section 7.4.
REQ19 - The mechanism MUST be extensible to allow a set of assigned
telephone numbers to comprise local numbers as specified in RFC 3966
[15], which can be in contiguous ranges, discrete, or in any
combination of the two.
Assignment of telephone numbers to a registration is performed by
the SSP's registrar, which is not precluded from assigning local
numbers in any combination it desires.
REQ20 - The mechanism MUST be extensible to allow a set of
arbitrarily assigned SIP URI's as specified in RFC 3261 [3], as
opposed to just telephone numbers, without requiring a complete
change of mechanism as compared to that used for telephone numbers.
The mechanism is extensible in such a fashion, as demonstrated by
the document "GIN with Literal AoRs for SIP in SSPs (GLASS)" [23].
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-PBXes.
The desired property is satisfied; the routing mechanism described
in this document is identical to the routing performed for singly
registered AORs.
DES2 - The mechanism SHOULD scale to SIP-PBXes of several thousand
assigned telephone numbers.
The desired property is satisfied; nothing in this document
precludes DID number pools of arbitrary size.
DES3 - The mechanism SHOULD scale to support several thousand SIP-
PBX's on a single SSP.
The desired property is satisfied; nothing in this document
precludes an arbitrary number of SIP-PBXes from attaching to a
single SSP.
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RFC 6140 Globally Identifiable Number Routing March 2011
Author's Address
Adam Roach
Tekelec
17210 Campbell Rd.
Suite 250
Dallas, TX 75252
US
EMail: adam@nostrum.com
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Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP)
RFC TOTAL SIZE: 79372 bytes
PUBLICATION DATE: Thursday, March 3rd, 2011
LEGAL RIGHTS: The IETF Trust (see BCP 78)
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