[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits] [IPR]
Versions: (draft-jennings-sipping-outbound)
00 01 02 03 04 05 06 07 08 09 10 11
12 13 14 15 16 17 18 19 20 RFC 5626
Network Working Group C. Jennings, Ed.
Internet-Draft Cisco Systems
Updates: 3261,3327 (if approved) R. Mahy, Ed.
Intended status: Standards Track Plantronics
Expires: July 11, 2007 January 7, 2007
Managing Client Initiated Connections in the Session Initiation Protocol
(SIP)
draft-ietf-sip-outbound-07
Status of this Memo
By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware
have been or will be disclosed, and any of which he or she becomes
aware will be disclosed, in accordance with Section 6 of BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
This Internet-Draft will expire on July 11, 2007.
Copyright Notice
Copyright (C) The Internet Society (2007).
Abstract
The Session Initiation Protocol (SIP) allows proxy servers to
initiate TCP connections and send asynchronous UDP datagrams to User
Agents in order to deliver requests. However, many practical
considerations, such as the existence of firewalls and Network
Address Translators (NATs), prevent servers from connecting to User
Agents in this way. This specification defines behaviors for User
Jennings & Mahy Expires July 11, 2007 [Page 1]
Internet-Draft Client Initiated Connections in SIP January 2007
Agents, registrars and proxy servers that allow requests to be
delivered on existing connections established by the User Agent. It
also defines keep alive behaviors needed to keep NAT bindings open
and specifies the usage of multiple connections.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions and Terminology . . . . . . . . . . . . . . . . . 4
2.1. Definitions . . . . . . . . . . . . . . . . . . . . . . . 5
3. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Summary of Mechanism . . . . . . . . . . . . . . . . . . . 5
3.2. Single Registrar and UA . . . . . . . . . . . . . . . . . 6
3.3. Multiple Connections from a User Agent . . . . . . . . . . 7
3.4. Edge Proxies . . . . . . . . . . . . . . . . . . . . . . . 9
3.5. Keepalive Technique . . . . . . . . . . . . . . . . . . . 11
4. User Agent Mechanisms . . . . . . . . . . . . . . . . . . . . 12
4.1. Instance ID Creation . . . . . . . . . . . . . . . . . . . 12
4.2. Initial Registrations . . . . . . . . . . . . . . . . . . 13
4.2.1. Registration by Other Instances . . . . . . . . . . . 15
4.3. Sending Requests . . . . . . . . . . . . . . . . . . . . . 15
4.4. Detecting Flow Failure . . . . . . . . . . . . . . . . . . 15
4.4.1. Keepalive with TCP KEEPALIVE . . . . . . . . . . . . . 16
4.4.2. Keepalive with STUN . . . . . . . . . . . . . . . . . 16
4.4.3. Flow Recovery . . . . . . . . . . . . . . . . . . . . 16
5. Edge Proxy Mechanisms . . . . . . . . . . . . . . . . . . . . 18
5.1. Processing Register Requests . . . . . . . . . . . . . . . 18
5.2. Generating Flow Tokens . . . . . . . . . . . . . . . . . . 18
5.3. Forwarding Requests . . . . . . . . . . . . . . . . . . . 19
5.4. Edge Proxy Keepalive Handling . . . . . . . . . . . . . . 20
6. Registrar Mechanisms: Processing REGISTER Requests . . . . . . 20
7. Authoritative Proxy Mechanisms: Forwarding Requests . . . . . 22
8. STUN Keepalive Processing . . . . . . . . . . . . . . . . . . 23
8.1. Explicit Probes . . . . . . . . . . . . . . . . . . . . . 25
8.2. Use with Sigcomp . . . . . . . . . . . . . . . . . . . . . 25
9. Example Message Flow . . . . . . . . . . . . . . . . . . . . . 26
10. Grammar . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
11. Definition of 430 Flow Failed response code . . . . . . . . . 30
12. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 30
12.1. Contact Header Field . . . . . . . . . . . . . . . . . . . 30
12.2. SIP/SIPS URI Parameters . . . . . . . . . . . . . . . . . 31
12.3. SIP Option Tag . . . . . . . . . . . . . . . . . . . . . . 31
12.4. Response Code . . . . . . . . . . . . . . . . . . . . . . 31
12.5. Media Feature Tag . . . . . . . . . . . . . . . . . . . . 31
13. Security Considerations . . . . . . . . . . . . . . . . . . . 32
14. Operational Notes on Transports . . . . . . . . . . . . . . . 33
15. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 34
Jennings & Mahy Expires July 11, 2007 [Page 2]
Internet-Draft Client Initiated Connections in SIP January 2007
16. Changes . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
16.1. Changes from 06 Version . . . . . . . . . . . . . . . . . 34
16.2. Changes from 05 Version . . . . . . . . . . . . . . . . . 34
16.3. Changes from 04 Version . . . . . . . . . . . . . . . . . 35
16.4. Changes from 03 Version . . . . . . . . . . . . . . . . . 36
16.5. Changes from 02 Version . . . . . . . . . . . . . . . . . 37
16.6. Changes from 01 Version . . . . . . . . . . . . . . . . . 37
16.7. Changes from 00 Version . . . . . . . . . . . . . . . . . 37
17. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 37
Appendix A. Default Flow Registration Backoff Times . . . . . . . 38
18. References . . . . . . . . . . . . . . . . . . . . . . . . . . 38
18.1. Normative References . . . . . . . . . . . . . . . . . . . 38
18.2. Informative References . . . . . . . . . . . . . . . . . . 39
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 40
Intellectual Property and Copyright Statements . . . . . . . . . . 42
Jennings & Mahy Expires July 11, 2007 [Page 3]
Internet-Draft Client Initiated Connections in SIP January 2007
1. Introduction
There are many environments for SIP [1] deployments in which the User
Agent (UA) can form a connection to a Registrar or Proxy but in which
connections in the reverse direction to the UA are not possible.
This can happen for several reasons. Connections to the UA can be
blocked by a firewall device between the UA and the proxy or
registrar, which will only allow new connections in the direction of
the UA to the Proxy. Similarly there a NAT could be present, which
is only capable of allowing new connections from the private address
side to the public side. This specification allows SIP registration
when the UA is behind such a firewall or NAT.
Most IP phones and personal computers get their network
configurations dynamically via a protocol such as DHCP (Dynamic Host
Configuration Protocol). These systems typically do not have a
useful name in the Domain Name System (DNS), and they almost never
have a long-term, stable DNS name that is appropriate for use in the
subjectAltName of a certificate, as required by [1]. However, these
systems can still act as a TLS client and form connections to a proxy
or registrar which authenticates with a server certificate. The
server can authenticate the UA using a shared secret in a digest
challenge over that TLS connection.
The key idea of this specification is that when a UA sends a REGISTER
request, the proxy can later use this same network "flow"--whether
this is a bidirectional stream of UDP datagrams, a TCP connection, or
an analogous concept of another transport protocol--to forward any
requests that need to go to this UA. For a UA to receive incoming
requests, the UA has to connect to a server. Since the server can't
connect to the UA, the UA has to make sure that a flow is always
active. This requires the UA to detect when a flow fails. Since
such detection takes time and leaves a window of opportunity for
missed incoming requests, this mechanism allows the UA to use
multiple flows to the proxy or registrar. This specification also
defines how SIP implements the STUN keepalive usage. The keepalive
mechanism is used to keep NAT bindings fresh, and to allow the UA to
detect when a flow has failed.
2. Conventions and 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 RFC 2119 [2].
Jennings & Mahy Expires July 11, 2007 [Page 4]
Internet-Draft Client Initiated Connections in SIP January 2007
2.1. Definitions
Authoritative Proxy: A proxy that handles non-REGISTER requests for
a specific Address-of-Record (AOR), performs the logical Location
Server lookup described in RFC 3261, and forwards those requests
to specific Contact URIs.
Edge Proxy: An Edge Proxy is any proxy that is located topologically
between the registering User Agent and the Authoritative Proxy.
Flow: A Flow is a network protocol layer (layer 4) association
between two hosts that is represented by the network address and
port number of both ends and by the protocol. For TCP, a flow is
equivalent to a TCP connection. For UDP a flow is a bidirectional
stream of datagrams between a single pair of IP addresses and
ports of both peers. With TCP, a flow often has a one to one
correspondence with a single file descriptor in the operating
system.
reg-id: This refers to the value of a new header field parameter
value for the Contact header field. When a UA registers multiple
times, each simultaneous registration gets a unique reg-id value.
instance-id: This specification uses the word instance-id to refer
to the value of the "sip.instance" media feature tag in the
Contact header field. This is a Uniform Resource Name (URN) that
uniquely identifies this specific UA instance.
outbound-proxy-set A set of SIP URIs (Uniform Resource Identifiers)
that represents each of the outbound proxies (often Edge Proxies)
with which the UA will attempt to maintain a direct flow. The
first URI in the set is often referred to as the primary outbound
proxy and the second as the secondary outbound proxy. There is no
difference between any of the URIs in this set, nor does the
primary/secondary terminology imply that one is preferred over the
other.
3. Overview
Several scenarios in which this technique is useful are discussed
below, including the simple co-located registrar and proxy, a User
Agent desiring multiple connections to a resource (for redundancy,
for example), and a system that uses Edge Proxies.
3.1. Summary of Mechanism
The overall approach is fairly simple. Each UA has a unique
instance-id that stays the same for this UA even if the UA reboots or
is power cycled. Each UA can register multiple times over different
connections for the same SIP Address of Record (AOR) to achieve high
reliability. Each registration includes the instance-id for the UA
and a reg-id label that is different for each flow. The registrar
Jennings & Mahy Expires July 11, 2007 [Page 5]
Internet-Draft Client Initiated Connections in SIP January 2007
can use the instance-id to recognize that two different registrations
both reach the same UA. The registrar can use the reg-id label to
recognize that a UA is registering after a reboot or a network
failure.
When a proxy goes to route a message to a UA for which it has a
binding, it can use any one of the flows on which a successful
registration has been completed. A failure on a particular flow can
be tried again on an alternate flow. Proxies can determine which
flows go to the same UA by comparing the instance-id. Proxies can
tell that a flow replaces a previously abandoned flow by looking at
the reg-id.
UAs use the STUN (Simple Traversal of UDP through NATs) protocol as
the keepalive mechanism to keep their flow to the proxy or registrar
alive.
3.2. Single Registrar and UA
In the topology shown below, a single server is acting as both a
registrar and proxy.
+-----------+
| Registrar |
| Proxy |
+-----+-----+
|
|
+----+--+
| User |
| Agent |
+-------+
User Agents which form only a single flow continue to register
normally but include the instance-id as described in Section 4.1.
The UA can also include a reg-id parameter which is used to allow the
registrar to detect and avoid keeping invalid contacts when a UA
reboots or reconnects after its old connection has failed for some
reason.
For clarity, here is an example. Bob's UA creates a new TCP flow to
the registrar and sends the following REGISTER request.
Jennings & Mahy Expires July 11, 2007 [Page 6]
Internet-Draft Client Initiated Connections in SIP January 2007
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/TCP 192.0.2.1;branch=z9hG4bK-bad0ce-11-1036
Max-Forwards: 70
From: Bob <sip:bob@example.com>;tag=d879h76
To: Bob <sip:bob@example.com>
Call-ID: 8921348ju72je840.204
CSeq: 1 REGISTER
Supported: path
Contact: <sip:line1@192.168.0.2>; reg-id=1;
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000A95A0E128>"
Content-Length: 0
The registrar challenges this registration to authenticate Bob. When
the registrar adds an entry for this contact under the AOR for Bob,
the registrar also keeps track of the connection over which it
received this registration.
The registrar saves the instance-id
("urn:uuid:00000000-0000-0000-0000-000A95A0E128") and reg-id ("1")
along with the rest of the Contact header field. If the instance-id
and reg-id are the same as a previous registration for the same AOR,
the registrar replaces the old Contact URI and flow information.
This allows a UA that has rebooted to replace its previous
registration for each flow with minimal impact on overall system
load.
When Alice sends a request to Bob, his authoritative proxy selects
the target set. The proxy forwards the request to elements in the
target set based on the proxy's policy. The proxy looks at the
target set and uses the instance-id to understand if two targets both
end up routing to the same UA. When the proxy goes to forward a
request to a given target, it looks and finds the flows over which it
received the registration. The proxy then forwards the request on
that flow instead of trying to form a new flow to that contact. This
allows the proxy to forward a request to a particular contact over
the same flow that the UA used to register this AOR. If the proxy
has multiple flows that all go to this UA, it can choose any one of
registration bindings for this AOR that has the same instance-id as
the selected UA.
3.3. Multiple Connections from a User Agent
There are various ways to deploy SIP to build a reliable and scalable
system. This section discusses one such design that is possible with
the mechanisms in this specification. Other designs are also
possible.
In the example system below, the logical outbound proxy/registrar for
Jennings & Mahy Expires July 11, 2007 [Page 7]
Internet-Draft Client Initiated Connections in SIP January 2007
the domain is running on two hosts that share the appropriate state
and can both provide registrar and outbound proxy functionality for
the domain. The UA will form connections to two of the physical
hosts that can perform the outbound proxy/registrar function for the
domain. Reliability is achieved by having the UA form two TCP
connections to the domain.
Scalability is achieved by using DNS SRV to load balance the primary
connection across a set of machines that can service the primary
connection, and also using DNS SRV to load balance across a separate
set of machines that can service the secondary connection. The
deployment here requires that DNS is configured with one entry that
resolves to all the primary hosts and another entry that resolves to
all the secondary hosts. While this introduces additional DNS
configuration, the approach works and requires no addition SIP
extensions.
Note: Approaches which select multiple connections from a single
DNS SRV set were also considered, but cannot prevent two
connections from accidentally resolving to the same host. The
approach in this document does not prevent future extensions, such
as the SIP UA configuration framework [18], from adding other ways
for a User Agent to discover its outbound-proxy-set.
+-------------------+
| Domain |
| Logical Proxy/Reg |
| |
|+-----+ +-----+|
||Host1| |Host2||
|+-----+ +-----+|
+---\------------/--+
\ /
\ /
\ /
\ /
+------+
| User |
| Agent|
+------+
The UA is configured with multiple outbound proxy registration URIs.
These URIs are configured into the UA through whatever the normal
mechanism is to configure the proxy or registrar address in the UA.
If the AOR is Alice@example.com, the outbound-proxy-set might look
something like "sip:primary.example.com;keepalive=stun" and "sip:
secondary.example.com;keepalive=stun". The "keepalive=stun" tag
indicates that a SIP server supports STUN and SIP multiplexed over
Jennings & Mahy Expires July 11, 2007 [Page 8]
Internet-Draft Client Initiated Connections in SIP January 2007
the same flow, as described later in this specification. Note that
each URI in the outbound-proxy-set could resolve to several different
physical hosts. The administrative domain that created these URIs
should ensure that the two URIs resolve to separate hosts. These
URIs are handled according to normal SIP processing rules, so
mechanisms like SRV can be used to do load balancing across a proxy
farm.
The domain also needs to ensure that a request for the UA sent to
host1 or host2 is then sent across the appropriate flow to the UA.
The domain might choose to use the Path header approach (as described
in the next section) to store this internal routing information on
host1 or host2.
When a single server fails, all the UAs that have a flow through it
will detect a flow failure and try to reconnect. This can cause
large loads on the server. When large numbers of hosts reconnect
nearly simultaneously, this is referred to as the avalanche restart
problem, and is further discussed in Section 4.4.3. The multiple
flows to many servers help reduce the load caused by the avalanche
restart. If a UA has multiple flows, and one of the servers fails,
the UA delays the specified time before trying to form a new
connection to replace the flow to the server that failed. By
spreading out the time used for all the UAs to reconnect to a server,
the load on the server farm is reduced.
When used in this fashion to achieve high reliability, the operator
will need to configure DNS such that the various URIs in the outbound
proxy set do not resolve to the same host.
Another motivation for maintaining multiple flows between the UA and
its registrar is related to multihomed UAs. Such UAs can benefit
from multiple connections from different interfaces to protect
against the failure of an individual access link.
3.4. Edge Proxies
Some SIP deployments use edge proxies such that the UA sends the
REGISTER to an Edge Proxy that then forwards the REGISTER to the
Registrar. The Edge Proxy includes a Path header [3] so that when
the registrar later forwards a request to this UA, the request is
routed through the Edge Proxy. There could be a NAT or firewall
between the UA and the Edge Proxy.
Jennings & Mahy Expires July 11, 2007 [Page 9]
Internet-Draft Client Initiated Connections in SIP January 2007
+---------+
|Registrar|
|Proxy |
+---------+
/ \
/ \
/ \
+-----+ +-----+
|Edge1| |Edge2|
+-----+ +-----+
\ /
\ /
----------------------------NAT/FW
\ /
\ /
+------+
|User |
|Agent |
+------+
These systems can use effectively the same mechanism as described in
the previous sections but need to use the Path header. When the Edge
Proxy receives a registration, it needs to create an identifier value
that is unique to this flow (and not a subsequent flow with the same
addresses) and put this identifier in the Path header URI. This
identifier has two purposes. First, it allows the Edge Proxy to map
future requests back to the correct flow. Second, because the
identifier will only be returned if the user authentication with the
registrar succeeds, it allows the Edge Proxy to indirectly check the
user's authentication information via the registrar. The identifier
is placed in the user portion of a loose route in the Path header.
If the registration succeeds, the Edge Proxy needs to map future
requests that are routed to the identifier value from the Path
header, to the associated flow.
The term Edge Proxy is often used to refer to deployments where the
Edge Proxy is in the same administrative domain as the Registrar.
However, in this specification we use the term to refer to any proxy
between the UA and the Registrar. For example the Edge Proxy may be
inside an enterprise that requires its use and the registrar could be
from a service provider with no relationship to the enterprise.
Regardless if they are in the same administrative domain, this
specification requires that Registrars and Edge proxies support the
Path header mechanism in RFC 3327 [3].
Jennings & Mahy Expires July 11, 2007 [Page 10]
Internet-Draft Client Initiated Connections in SIP January 2007
3.5. Keepalive Technique
A keepalive mechanism needs to detect failure of a connection and
changes to the NAT public mapping, as well as keeping any NAT
bindings refreshed. This specification describes using STUN [4] over
the same flow as the SIP traffic to perform the keepalive. For
connection-oriented transports (e.g. TCP and TLS over TCP), the UAC
MAY use TCP keepalives to detect flow failure if the UAC can send
these keepalives and detect a keepalive failure according to the time
frames described in Section 4.4.
Note: when TCP is being used, it's natural to think of using TCP
KEEPALIVE. Unfortunately, many operating systems and programming
environments do not allow the keepalive time to be set on a per-
connection basis. Thus, applications may not be able to set an
appropriate time.
For connection-less transports, a flow definition could change
because a NAT device in the network path reboots and the resulting
public IP address or port mapping for the UA changes. To detect
this, requests are sent over the same flow that is being used for the
SIP traffic. The proxy or registrar acts as a STUN server on the SIP
signaling port.
Note: The STUN mechanism is very robust and allows the detection
of a changed IP address. Many other options were considered, but
the SIP Working Group selected the STUN-based approach, since it
works over any transport. Approaches using SIP requests were
abandoned because to achieve the required performance, the server
needs to deviate from the SIP specification in significant ways.
This would result in many undesirable and non-deterministic
behaviors in some environments.
Another approach considered to detect a changed flow was using
OPTIONS messages and the rport parameter. Although the OPTIONS
approach has the advantage of being backwards compatible, it also
significantly increases the load on the proxy or registrar server.
Related to this idea was an idea of creating a new SIP PING method
that was like OPTIONS but faster. It would be critical that this
PING method did not violate the processing requirements of a
proxies and UAS so it was never clear how it would be
significantly faster than OPTIONS given it would still have to
obey things like checking the Proxy-Require header. After
considerable consideration the working group came to some
consensus that the STUN approach was a better solution that these
alternative designs.
When the UA detects that a flow has failed or that the flow
definition has changed, the UA needs to re-register and will use the
Jennings & Mahy Expires July 11, 2007 [Page 11]
Internet-Draft Client Initiated Connections in SIP January 2007
back-off mechanism described in Section 4 to provide congestion
relief when a large number of agents simultaneously reboot.
4. User Agent Mechanisms
4.1. Instance ID Creation
Each UA MUST have an Instance Identifier URN that uniquely identifies
the device. Usage of a URN provides a persistent and unique name for
the UA instance. It also provides an easy way to guarantee
uniqueness within the AOR. This URN MUST be persistent across power
cycles of the device. The Instance ID MUST NOT change as the device
moves from one network to another.
A UA SHOULD use a UUID URN [5] as its instance-id. The UUID URN
allows for non-centralized computation of a URN based on time, unique
names (such as a MAC address), or a random number generator.
A device like a soft-phone, when first installed, can generate a
UUID [5] and then save this in persistent storage for all future
use. For a device such as a hard phone, which will only ever have
a single SIP UA present, the UUID can include the MAC address and
be generated at any time because it is guaranteed that no other
UUID is being generated at the same time on that physical device.
This means the value of the time component of the UUID can be
arbitrarily selected to be any time less than the time when the
device was manufactured. A time of 0 (as shown in the example in
Section 3.2) is perfectly legal as long as the device knows no
other UUIDs were generated at this time.
If a URN scheme other than UUID is used, the URN MUST be selected
such that the instance can be certain that no other instance
registering against the same AOR would choose the same URN value. An
example of a URN that would not meet the requirements of this
specification is the national bibliographic number [19]. Since there
is no clear relationship between a SIP UA instance and a URN in this
namespace, there is no way a selection of a value can be performed
that guarantees that another UA instance doesn't choose the same
value.
The UA SHOULD include a "sip.instance" media feature tag as a UA
characteristic [6] in requests and responses. As described in [6],
this media feature tag will be encoded in the Contact header field as
the "+sip.instance" Contact header field parameter. The value of
this parameter MUST be a URN [7]. One case where a UA may not want
to include the URN in the sip.instance media feature tag is when it
is making an anonymous request or some other privacy concern requires
Jennings & Mahy Expires July 11, 2007 [Page 12]
Internet-Draft Client Initiated Connections in SIP January 2007
that the UA not reveal its identity.
RFC 3840 [6] defines equality rules for callee capabilities
parameters, and according to that specification, the
"sip.instance" media feature tag will be compared by case-
sensitive string comparison. This means that the URN will be
encapsulated by angle brackets ("<" and ">") when it is placed
within the quoted string value of the +sip.instance Contact header
field parameter. The case-sensitive matching rules apply only to
the generic usages defined in RFC 3840 [6] and in the caller
preferences specification [8]. When the instance ID is used in
this specification, it is effectively "extracted" from the value
in the "sip.instance" media feature tag. Thus, equality
comparisons are performed using the rules for URN equality that
are specific to the scheme in the URN. If the element performing
the comparisons does not understand the URN scheme, it performs
the comparisons using the lexical equality rules defined in RFC
2141 [7]. Lexical equality could result in two URNs being
considered unequal when they are actually equal. In this specific
usage of URNs, the only element which provides the URN is the SIP
UA instance identified by that URN. As a result, the UA instance
SHOULD provide lexically equivalent URNs in each registration it
generates. This is likely to be normal behavior in any case;
clients are not likely to modify the value of the instance ID so
that it remains functionally equivalent yet lexigraphically
different from previous registrations.
4.2. Initial Registrations
At configuration time UAs obtain one or more SIP URIs representing
the default outbound-proxy-set. This specification assumes the set
is determined via any of a number of configuration mechanisms, and
future specifications can define additional mechanisms such as using
DNS to discover this set. How the UA is configured is outside the
scope of this specification. However, a UA MUST support sets with at
least two outbound proxy URIs and SHOULD support sets with up to four
URIs. For each outbound proxy URI in the set, the UA SHOULD send a
REGISTER in the normal way using this URI as the default outbound
proxy. Forming the route set for the request is outside the scope of
this document, but typically results in sending the REGISTER such
that the topmost Route header field contains a loose route to the
outbound proxy URI. Other issues related to outbound route
construction are discussed in [20].
Registration requests, other than those described in Section 4.2.1,
MUST include an instance-id media feature tag as specified in
Section 4.1.
Jennings & Mahy Expires July 11, 2007 [Page 13]
Internet-Draft Client Initiated Connections in SIP January 2007
These ordinary registration requests include a distinct reg-id
parameter to the Contact header field. Each one of these
registrations will form a new flow from the UA to the proxy. The
sequence of reg-id values does not have to be sequential but MUST be
exactly the same sequence of reg-id values each time the UA instance
power cycles or reboots so that the reg-id values will collide with
the previously used reg-id values. This is so the registrar can
replace the older registration.
The UAC can situationally decide whether to request outbound
behavior by including or omitting the 'reg-id' parameter. For
example, imagine the outbound-proxy-set contains two proxies in
different domains, EP1 and EP2. If an outbound-style registration
succeeded for a flow through EP1, the UA might decide to include
'outbound' in its option-tag when registering with EP2, in order
to insure consistency. Similarly, if the registration through EP1
did not support outbound, the UA might decide to omit the 'reg-id'
parameter when registering with EP2.
The UAC MUST indicate that it supports the Path header [3] mechanism,
by including the 'path' option-tag in a Supported header field value
in its REGISTER requests. Other than optionally examining the Path
vector in the response, this is all that is required of the UAC to
support Path.
The UAC MAY examine successful registrations for the presence of an
'outbound' option-tag in a Supported header field value. Presence of
this option-tag indicates that the registrar is compliant with this
specification, and that any edge proxies which need to participate
are also compliant.
Note that the UA needs to honor 503 (Service Unavailable) responses
to registrations as described in RFC 3261 and RFC 3263 [9]. In
particular, implementors should note that when receiving a 503
(Service Unavailable) response with a Retry-After header field, the
UA is expected to wait the indicated amount of time and retry the
registration. A Retry-After header field value of 0 is valid and
indicates the UA is expected to retry the REGISTER immediately.
Implementations need to ensure that when retrying the REGISTER, they
revisit the DNS resolution results such that the UA can select an
alternate host from the one chosen the previous time the URI was
resolved.
Finally, re-registrations which merely refresh an existing valid
registration SHOULD be sent over the same flow as the original
registration.
Jennings & Mahy Expires July 11, 2007 [Page 14]
Internet-Draft Client Initiated Connections in SIP January 2007
4.2.1. Registration by Other Instances
A User Agent MUST NOT include a reg-id header parameter in the
Contact header field of a registration if the registering UA is not
the same instance as the UA referred to by the target Contact header
field. (This practice is occasionally used to install forwarding
policy into registrars.)
Note that a UAC also MUST NOT include an instance-id or reg-id
parameter in a request to unregister all Contacts (a single Contact
header field value with the value of "*").
4.3. Sending Requests
When a UA is about to send a request, it first performs normal
processing to select the next hop URI. The UA can use a variety of
techniques to compute the route set and accordingly the next hop URI.
Discussion of these techniques is outside the scope of this document
but could include mechanisms specified in RFC 3608 [21] (Service
Route) and [20].
The UA performs normal DNS resolution on the next hop URI (as
described in RFC 3263 [9]) to find a protocol, IP address, and port.
For non-TLS protocols, if the UA has an existing flow to this IP
address, and port with the correct protocol, then the UA MUST use the
existing connection. For TLS protocols, there MUST also be a match
between the host production in the next hop and one of the URIs
contained in the subjectAltName in the peer certificate. If the UA
cannot use one of the existing flows, then it SHOULD form a new flow
by sending a datagram or opening a new connection to the next hop, as
appropriate for the transport protocol.
Note that if the UA wants its flow to work through NATs or
firewalls it still needs to put the 'rport' parameter [10] in its
Via header field value, and send from the port it is prepared to
receive on. More general information about NAT traversal in SIP
is described in [22].
4.4. Detecting Flow Failure
The UA needs to detect when a specific flow fails. The UA actively
tries to detect failure by periodically sending keepalive messages
using one of the techniques described in Section 4.4.1 or
Section 4.4.2. If a flow has failed, the UA follows the procedures
in Section 4.2 to form a new flow to replace the failed one.
The time between keepalive requests when using UDP-based transports
SHOULD be a random number between 24 and 29 seconds while for TCP-
Jennings & Mahy Expires July 11, 2007 [Page 15]
Internet-Draft Client Initiated Connections in SIP January 2007
based transports it SHOULD be a random number between 95 and 120
seconds. These times MAY be configurable. Issues such as battery
consumption might motivate longer keepalive intervals.
Note on selection of time values: For UDP, the upper bound of 29
seconds was selected so that multiple STUN packets could be sent
before 30 seconds based on information that many NATs have UDP
timeouts as low as 30 seconds. The 24 second lower bound was
selected so that after 10 minutes the jitter introduced by
different timers will make the keepalive requests unsynchronized
to evenly spread the load on the servers. For TCP, the 120
seconds upper bound was chosen based on the idea that for a good
user experience, failures normally will be detected in this amount
of time and a new connection set up. Operators that wish to
change the relationship between load on servers and the expected
time that a user might not receive inbound communications will
probably adjust this time. The 95 seconds lower bound was chosen
so that the jitter introduced will result in a relatively even
load on the servers after 30 minutes.
4.4.1. Keepalive with TCP KEEPALIVE
User Agents that are capable of generating per-connection TCP
keepalives with timer values consistent with those in this section
MAY use TCP keepalives instead of using STUN keepalives for TCP-based
flows.
4.4.2. Keepalive with STUN
User Agents that form flows, check if the configured URI they are
connecting to has a 'keepalive' URI parameter (defined in Section 12)
with the value of 'stun'. If the parameter is present and the UA is
not already performing keepalives using another supported mechanism,
the UA needs to periodically perform keepalive checks by sending STUN
[4] Binding Requests over the flow as described in Section 8.
If the XOR-MAPPED-ADDRESS in the STUN Binding Response changes, the
UA MUST treat this event as a failure on the flow.
4.4.3. Flow Recovery
When a flow to a particular URI in the outbound-proxy-set fails, the
UA needs to form a new flow to replace the old flow and replace any
registrations that were previously sent over this flow. Each new
registration MUST have the same reg-id as the registration it
replaces. This is done in much the same way as forming a brand new
flow as described in Section 4.2; however, if there is a failure in
forming this flow, the UA needs to wait a certain amount of time
Jennings & Mahy Expires July 11, 2007 [Page 16]
Internet-Draft Client Initiated Connections in SIP January 2007
before retrying to form a flow to this particular next hop.
The amount of time to wait depends if the previous attempt at
establishing a flow was successful. For the purposes of this
section, a flow is considered successful if outbound registration
succeeded and keepalives have not timed out for min-regtime seconds
(default of 120 seconds) after a registration. For STUN-based
keepalives, this means three successful STUN transactions over UDP or
one successful STUN transaction over TCP. If a flow is established
and is alive after this amount of time, the number of consecutive
registration failures is set to zero. Each time a flow fails before
two minutes, the number of consecutive registration failures is
incremented by one. Note that a failure during the initial STUN
validation does not count against the number of consecutive
registration failures.
The number of seconds to wait is computed in the following way. If
all of the flows to every URI in the outbound proxy set have failed,
the base time is set to 30 seconds; otherwise, in the case where at
least one of the flows has not failed, the base time is set to 90
seconds. The wait time is computed by taking two raised to the power
of the number of consecutive registration failures for that URI, and
multiplying this by the base time, up to a maximum of 1800 seconds.
wait-time = min( max-time, (base-time * (2 ^ consecutive-failures)))
These times MAY be configurable in the UA. The four times are:
o max-time with a default of 1800 seconds
o base-time-all-fail with a default of 30 seconds
o base-time-not-failed with a default of 90 seconds
o min-regtime with a default of 120 seconds
For example, if the base time is 30 seconds, and there were three
failures, then the wait time is min(1800,30*(2^3)) or 240 seconds.
The delay time is computed by selecting a uniform random time between
50 and 100 percent of the wait time. The UA MUST wait for the value
of the delay time before trying another registration to form a new
flow for that URI.
To be explicitly clear on the boundary conditions: when the UA boots
it immediately tries to register. If this fails and no registration
on other flows succeed, the first retry happens somewhere between 30
and 60 seconds after the failure of the first registration request.
If the number of consecutive-failures is large enough that the
maximum of 1800 seconds is reached, the UA will keep trying
indefinitely with a random time of 15 to 30 minutes between each
attempt.
Jennings & Mahy Expires July 11, 2007 [Page 17]
Internet-Draft Client Initiated Connections in SIP January 2007
5. Edge Proxy Mechanisms
5.1. Processing Register Requests
When an Edge Proxy receives a registration request with a reg-id
header parameter in the Contact header field, it needs to determine
if it (the edge proxy) will have to be visited for any subsequent
requests sent to the user agent identified in the Contact header
field, or not. If the Edge Proxy determines that this is the case,
it inserts its URI in a Path header field value as described in RFC
3327 [3]. If the Edge Proxy is the first SIP node after the UAC, it
either MUST store a "flow token"--containing information about the
flow from the previous hop--in its Path URI, or reject the request.
The flow token MUST be an identifier that is unique to this network
flow. The flow token MAY be placed in the userpart of the URI. In
addition, the first node MUST include an 'ob' URI parameter in its
Path header field value. If the Edge Proxy is not the first SIP node
after the UAC it MUST NOT place an 'ob' URI parameter in a Path
header field value. The Edge Proxy can determine if it is the first
hop by examining the Via header field.
5.2. Generating Flow Tokens
A trivial but impractical way to satisfy the flow token requirement
in Section 5.1 involves storing a mapping between an incrementing
counter and the connection information; however this would require
the Edge Proxy to keep an impractical amount of state. It is unclear
when this state could be removed and the approach would have problems
if the proxy crashed and lost the value of the counter. Two
stateless examples are provided below. A proxy can use any algorithm
it wants as long as the flow token is unique to a flow, the flow can
be recovered from the token, and the token can not be modified by
attackers.
Algorithm 1: The proxy generates a flow token for connection-
oriented transports by concatenating the file descriptor (or
equivalent) with the NTP time the connection was created, and
base64 encoding the result. This results in an identifier
approximately 16 octets long. The proxy generates a flow token
for UDP by concatenating the file descriptor and the remote IP
address and port, then base64 encoding the result. (No NTP time
is needed for UDP.) This algorithm MUST NOT be used unless all
messages between the Edge proxy and Registrar use a SIPS protected
transport. If the SIPS level of integrity protection is not
available, an attacker can hijack another user's calls.
Jennings & Mahy Expires July 11, 2007 [Page 18]
Internet-Draft Client Initiated Connections in SIP January 2007
Algorithm 2: When the proxy boots it selects a 20-octet crypto
random key called K that only the Edge Proxy knows. A byte array,
called S, is formed that contains the following information about
the flow the request was received on: an enumeration indicating
the protocol, the local IP address and port, the remote IP address
and port. The HMAC of S is computed using the key K and the HMAC-
SHA1-80 algorithm, as defined in [11]. The concatenation of the
HMAC and S are base64 encoded, as defined in [12], and used as the
flow identifier. When using IPv4 addresses, this will result in a
32-octet identifier.
5.3. Forwarding Requests
When an Edge Proxy receives a request, it applies normal routing
procedures with the following addition. If the Edge Proxy receives a
request where the edge proxy is the host in the topmost Route header
field value, and the Route header contains a flow token, the proxy
compares the flow in the flow token with the source of the request.
If these refer to the same flow, the Edge Proxy removes the Route
header and continues processing the request. Otherwise, if the top-
most Route header refers to the Edge Proxy and contains a valid flow
identifier token created by this proxy, the proxy MUST remove the
Route header and forward the request over the 'logical flow'
identified by the flow token, that is known to deliver data to the
specific target UA instance. For connection-oriented transports, if
the flow no longer exists the proxy SHOULD send a 430 (Flow Failed)
response to the request.
The advantage to a stateless approach to managing the flow
information is that there is no state on the Edge Proxy that
requires clean up or that has to be synchronized with the
registrar.
Proxies which used one of the two algorithms described in this
document to form a flow token follow the procedures below to
determine the correct flow.
Algorithm 1: The proxy base64 decodes the user part of the Route
header. For a TCP-based transport, if a connection specified by
the file descriptor is present and the creation time of the file
descriptor matches the creation time encoded in the Route header,
the proxy forwards the request over that connection. For a UDP-
based transport, the proxy forwards the request from the encoded
file descriptor to the source IP address and port.
Jennings & Mahy Expires July 11, 2007 [Page 19]
Internet-Draft Client Initiated Connections in SIP January 2007
Algorithm 2: To decode the flow token, take the flow identifier in
the user portion of the URI and base64 decode it, then verify the
HMAC is correct by recomputing the HMAC and checking it matches.
If the HMAC is not correct, the proxy SHOULD send a 403
(Forbidden) response. If the HMAC is correct then the proxy
SHOULD forward the request on the flow that was specified by the
information in the flow identifier. If this flow no longer
exists, the proxy SHOULD send a 430 (Flow Failed) response to the
request.
Note that this specification needs mid-dialog requests to be routed
over the same flows as those stored in the Path vector from the
initial registration, but techniques to ensure that mid-dialog
requests are routed over an existing flow are not part of this
specification. However, an approach such as having the Edge Proxy
Record-Route with a flow token is one way to ensure that mid-dialog
requests are routed over the correct flow.
5.4. Edge Proxy Keepalive Handling
All edge proxies compliant with this specification MUST implement
support for the STUN NAT Keepalive usage on its SIP ports as
described in Section 8.
6. Registrar Mechanisms: Processing REGISTER Requests
This specification updates the definition of a binding in RFC 3261
[1] Section 10 and RFC 3327 [3] Section 5.3.
When no +sip.instance media feature parameter is present in a Contact
header field value in a REGISTER request, the corresponding binding
is still between an AOR and the URI from that Contact header field
value. When a +sip.instance media feature parameter is present in a
Contact header field value in a REGISTER request, the corresponding
binding is between an AOR and the combination of the instance-id
(from the +sip.instance media feature parameter) and the value of
reg-id parameter if it is present. For a binding with an
instance-id, the registrar still stores the Contact header field
value URI with the binding, but does not consider the Contact URI for
comparison purposes. A Contact header field value with an
instance-id but no reg-id is valid, but one with a reg-id but no
instance-id is not. If the registrar processes a Contact header
field value with a reg-id but no instance-id, it simply ignores the
reg-id parameter. The registrar MUST be prepared to receive,
simultaneously for the same AOR, some registrations that use
instance-id and reg-id and some registrations that do not.
Jennings & Mahy Expires July 11, 2007 [Page 20]
Internet-Draft Client Initiated Connections in SIP January 2007
Registrars which implement this specification MUST support the Path
header mechanism [3].
In addition to the normal information stored in the binding record,
some additional information needs to be stored for any registration
that contains an instance-id and a reg-id header parameter in the
Contact header field value. First the registrar examines the first
Path header field value, if any. If the Path header field exists and
the first URI does not have an 'ob' URI parameter, the registrar MUST
ignore the reg-id parameter and continue processing the request as if
it did not support this specification. Likewise if the REGISTER
request visited an edge proxy, but no Path header field values are
present, the registrar MUST ignore the reg-id parameter.
Specifically, the registrar MUST use RFC 3261 Contact binding rules,
and MUST NOT include the 'outbound' option-tag in its Supported
header field. The registrar can determine if it is the first hop by
examining the Via header field.
If the UAC has a direct flow with the registrar, the registrar MUST
store enough information to uniquely identify the network flow over
which the request arrived. For common operating systems with TCP,
this would typically just be the handle to file descriptor where the
handle would become invalid if the TCP session was closed. For
common operating systems with UDP this would typically be the file
descriptor for the local socket that received the request, the local
interface, and the IP address and port number of the remote side that
sent the request. The registrar MAY store this information by adding
itself to the Path header field with an appropriate flow token.
The registrar MUST also store all the Contact header field
information including the reg-id and instance-id parameters and
SHOULD also store the time at which the binding was last updated. If
a Path header field is present, RFC 3327 [3] requires the registrar
to store this information as well. If the registrar receives a re-
registration, it MUST update any information that uniquely identifies
the network flow over which the request arrived if that information
has changed, and SHOULD update the time the binding was last updated.
The Registrar MUST include the 'outbound' option-tag (defined in
Section (Section 12.1)) in a Supported header field value in its
responses to REGISTER requests for which it has performed outbound
processing. The Registrar MAY be configured with local policy to
reject any registrations that do not include the instance-id and
reg-id. Note that the requirements in this section applies to both
REGISTER requests received from an Edge Proxy as well as requests
received directly from the UAC.
To be compliant with this specification, registrars which can receive
Jennings & Mahy Expires July 11, 2007 [Page 21]
Internet-Draft Client Initiated Connections in SIP January 2007
SIP requests directly from a UAC without intervening edge proxies
MUST implement the STUN NAT Keepalive usage on its SIP ports as
described in Section 8.
7. Authoritative Proxy Mechanisms: Forwarding Requests
When a proxy uses the location service to look up a registration
binding and then proxies a request to a particular contact, it
selects a contact to use normally, with a few additional rules:
o The proxy MUST NOT populate the target set with more than one
contact with the same AOR and instance-id at a time. If a request
for a particular AOR and instance-id fails with a 430 (Flow
Failed) response, the proxy SHOULD replace the failed branch with
another target (if one is available) with the same AOR and
instance-id, but a different reg-id.
o If the proxy receives a final response from a branch other than a
408 (Request Timeout) or a 430 (Flow Failed) response, the proxy
MUST NOT forward the same request to another target representing
the same AOR and instance-id. The targeted instance has already
provided its response.
The proxy uses normal forwarding rules looking at the next-hop target
of the message and the value of any stored Path header field vector
in the registration binding to decide how to forward the request and
populate the Route header in the request. If the proxy stored
information about the flow over which it received the REGISTER for
the binding, then the proxy MUST send the request over the same
'logical flow' saved with the binding that is known to deliver data
to the specific target UA instance.
Typically this means that for TCP, the request is sent on the same
TCP socket that received the REGISTER request. For UDP, the
request is sent from the same local IP address and port over which
the registration was received, to the same IP address and port
from which the REGISTER was received.
If a proxy or registrar receives information from the network that
indicates that no future messages will be delivered on a specific
flow, then the proxy MUST invalidate all the bindings in the target
set that use that flow (regardless of AOR). Examples of this are a
TCP socket closing or receiving a destination unreachable ICMP error
on a UDP flow. Similarly, if a proxy closes a file descriptor, it
MUST invalidate all the bindings in the target set with flows that
use that file descriptor.
Jennings & Mahy Expires July 11, 2007 [Page 22]
Internet-Draft Client Initiated Connections in SIP January 2007
8. STUN Keepalive Processing
This section describes changes to the SIP transport layer that allow
SIP and the STUN [4] NAT Keepalive usage to be mixed over the same
flow. The STUN messages are used to verify connectivity is still
available over a flow and to provide periodic keepalives. Note that
these STUN keepalives are always sent to the next SIP hop. STUN
messages are not delivered end-to-end.
The only STUN messages required by this usage are Binding Requests,
Binding Responses, and Error Responses. The UAC sends Binding
Requests over the same UDP flow, TCP connection, or TLS channel used
for sending SIP messages. These Binding Requests do not require any
STUN attributes. The UAS responds to a valid Binding Request with a
Binding Response which MUST include the XOR-MAPPED-ADDRESS attribute.
After a successful STUN response is received over TCP or TLS over
TCP, the underlying TCP connection is left in the active state.
If a server compliant to this section receives SIP requests on a
given interface and port, it MUST also provide a limited version of a
STUN server on the same interface and port as described in Section
12.3 of [4]. When STUN messages are sent with a SIP over TLS over
TCP flow, the STUN messages are sent inside the TLS-protected
channel.
It is easy to distinguish STUN and SIP packets sent over UDP,
because the first octet of a STUN packet has a value of 0 or 1
while the first octet of a SIP message is never a 0 or 1. For TCP
or TLS over TCP flows, determining if the first octet of the next
message in a stream is SIP or STUN is still straightforward. As
with any stream-based protocol, implementations need to be
prepared to receive STUN messages which cross a stream buffer
boundary, and SIP and STUN messages which share the same stream
buffer.
Because sending and receiving binary STUN data on the same ports used
for SIP is a significant and non-backwards compatible change to RFC
3261, this section requires a number of checks before sending STUN
messages to a SIP node. If a SIP node sends STUN requests (for
example due to misconfiguration) despite these warnings, the node
could be blacklisted for UDP traffic, or cause its TCP server to
loose framing over its connection. For each target node (as
determined by IP address, address family, and port number), the
sender needs to determine if that destination is validated to support
STUN, that it does not support STUN, or that it needs to be
validated.
When a URI is created that refers to a SIP node that supports STUN as
Jennings & Mahy Expires July 11, 2007 [Page 23]
Internet-Draft Client Initiated Connections in SIP January 2007
described in this section, the 'keepalive' URI parameter, as defined
in Section 12 SHOULD be added to the URI, with a value of 'stun'.
This allows a UA to inspect the URI to decide if it should attempt to
send STUN requests to this location. For example, an edge proxy
could insert this parameter into its Path URI so that the registering
UA can discover the edge proxy supports STUN keepalives.
A SIP node MUST NOT send STUN requests over a flow unless it has an
explicit indication that the target next hop SIP server claims to
support STUN. For example, automatic or manual configuration of an
outbound-proxy-set which contains the keepalive=stun parameter is
considered sufficient explicit indication. Note that UACs MUST NOT
use an ambiguous configuration option such as "Work through NATs?" or
"Do Keepalives?" to imply next hop STUN support. A SIP node MAY also
probe the next hop using a SIP OPTIONS request to check for support
of the 'sip-stun' option tag in a Supported header field.
Furthermore, even with explicit indication of next hop STUN support,
a SIP node needs to validate support for STUN the first time it sends
traffic to a specific invalidated target destination. A SIP node MAY
send one STUN request and its retransmissions to an invalidated
destination. If a STUN request ever succeeds to a destination, that
destination is thereafter validated for STUN support. If this
initial STUN request does not result in a STUN response, the SIP node
MUST NOT send additional STUN requests over this flow, unless a next-
hop probe later validates the destination. In addition, the SIP node
SHOULD remember invalidated destination nodes that have been used
within one hour and SHOULD NOT send additional STUN messages to any
of these destinations.
If this initial STUN request does not result in a STUN response, the
UA MAY send and explicit next-hop probe as described in Section 8.1.
If an explicit probe indicates support for the 'sip-stun' option-tag,
that destination is validated for STUN support. If an explicit probe
does not indicate support for the 'sip-stun' option-tag, the target
destination does not support STUN request, and the UAC MUST NOT send
further STUN requests to this destination.
Note that until STUN support has been verified, an initial STUN
failure over UDP is not considered a flow failure. For UDP flows, an
invalidated flow can still be reused for SIP traffic, however for
invalidated TCP or TLS over TCP flows, the connection over which STUN
requests were sent MUST be closed.
Typically, a SIP node first sends a SIP request and waits to
receive a final response (other than a 408 response) over a flow
to a new target destination, before sending any STUN messages.
When scheduled for the next NAT refresh, the SIP node sends a STUN
Jennings & Mahy Expires July 11, 2007 [Page 24]
Internet-Draft Client Initiated Connections in SIP January 2007
request to the target. If none of the STUN requests succeed
(result in a STUN success response), and the UAC has not already
done so, the UAC sends an OPTIONS request to the next hop to
verify support for the 'sip-stun' option-tag.
Once a destination is validated to support STUN messages, failure of
a STUN request (including its retransmissions) is considered a
failure of the underlying flow. For SIP over UDP flows, if the XOR-
MAPPED-ADDRESS returned over the flow changes, this indicates that
the underlying connectivity has changed, and is considered a flow
failure. A 408 response to a next-hop OPTIONS probe is also
considered a flow failure.
Note that failure of a flow causes a new flow to be formed and that
the STUN validation needs to be done for this new flow even if it is
to a destination that had previously been validated for STUN.
8.1. Explicit Probes
This section defines a new SIP option-tag called 'sip-stun'.
Advertising this option-tag indicates that the server can receive SIP
messages and STUN messages as part of the NAT Keepalive usage on the
same port. Clients that want to probe a SIP server to determine
support for STUN, can send an OPTIONS request to the next hop by
setting the Max-Forwards header field to zero or addressing the
request to that server. The OPTIONS response will contain a
Supported header field with a list of the server's supported option-
tags.
A UAC SHOULD NOT include the 'sip-stun' option-tag in a Proxy-
Require header. This is because a request with this header will
fail in some topologies where the first proxy support sip-stun,
but a subsequent proxy does not. Note that RFC 3261 does not
allow proxies to remove option-tags from a Proxy-Require header
field.
8.2. Use with Sigcomp
When STUN is used together with SigComp [23] compressed SIP messages
over the same flow, how the STUN messages are sent depends on the
transport protocol. For UDP flows, the STUN messages are simply sent
uncompressed, "outside" of SigComp. This is supported by
multiplexing STUN messages with SigComp messages by checking the two
topmost bits of the message. These bits are always one for SigComp,
or zero for STUN.
Jennings & Mahy Expires July 11, 2007 [Page 25]
Internet-Draft Client Initiated Connections in SIP January 2007
All SigComp messages contain a prefix (the five most-significant
bits of the first byte are set to one) that does not occur in
UTF-8 encoded text messages [RFC-2279], so for applications which
use this encoding (or ASCII encoding) it is possible to multiplex
uncompressed application messages and SigComp messages on the same
UDP port.
The most significant two bits of every STUN message are both
zeroes. This, combined with the magic cookie, aids in
differentiating STUN packets from other protocols when STUN is
multiplexed with other protocols on the same port.
For TCP-based flows, SigComp requires that all messages are processed
by the SigComp compressor to facilitate framing. For these
transports, STUN messages are sent encapsulated in the SigComp "well-
known shim header" as described in Section 11 of [24].
Because the bytecodes expressed in the well-known shim header do
not store any state, correlation of such SigComp requests to a
compartment is not necessary. To avoid ambiguity, we add the
following requirement: any SigComp state that might result from a
message that, once decompressed, turns out to be a STUN message,
MUST be discarded.
9. Example Message Flow
The following call flow shows a basic registration and an incoming
call. At some point, the flow to the Primary proxy is lost. An
incoming INVITE tries to reach the Callee through the Primary flow,
but receives an ICMP Unreachable message. The Caller retries using
the Secondary Edge Proxy, which uses a separate flow. Later, after
the Primary reboots, The Callee discovers the flow failure and
reestablishes a new flow to the Primary.
Jennings & Mahy Expires July 11, 2007 [Page 26]
Internet-Draft Client Initiated Connections in SIP January 2007
[-----example.com domain -------------------]
Caller Secondary Primary Callee
| | | (1) REGISTER |
| | |<-----------------|
| | |(2) 200 OK |
| | |----------------->|
| | | (3) REGISTER |
| |<------------------------------------|
| |(4) 200 OK | |
| |------------------------------------>|
| | | |
| | CRASH X |
|(5) INVITE | | |
|----------------------------------->| |
|(6) ICMP Unreachable | |
|<-----------------------------------| |
|(7) INVITE | | |
|---------------->| | |
| |(8) INVITE | |
| |------------------------------------>|
| |(9) 200 OK | |
| |<------------------------------------|
|(10) 200 OK | | |
|<----------------| | |
|(11) ACK | | |
|---------------->| | |
| |(12) ACK | |
| |------------------------------------>|
| | | |
| | REBOOT | |
| | |(13) REGISTER |
| | |<-----------------|
| | |(14) 200 OK |
| | |----------------->|
| | | |
|(15) BYE | | |
|---------------->| | |
| | (16) BYE | |
| |------------------------------------>|
| | | (17) 200 OK |
| |<------------------------------------|
| (18) 200 OK | | |
|<----------------| | |
| | | |
This call flow assumes that the Callee has been configured with a
proxy set that consists of "sip:pri.example.com;lr;keepalive=stun"
and "sip:sec.example.com;lr;keepalive=stun". The Callee REGISTER in
Jennings & Mahy Expires July 11, 2007 [Page 27]
Internet-Draft Client Initiated Connections in SIP January 2007
message (1) looks like:
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
Max-Forwards: 70
From: Callee <sip:callee@example.com>;tag=7F94778B653B
To: Callee <sip:callee@example.com>
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: path
Route: <sip:pri.example.com;lr;keepalive=stun>
Contact: <sip:callee@192.0.2.1>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=1
Content-Length: 0
In the message, note that the Route is set and the Contact header
field value contains the instance-id and reg-id. The response to the
REGISTER in message (2) would look like:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnashds7
From: Callee <sip:callee@example.com>;tag=7F94778B653B
To: Callee <sip:callee@example.com>;tag=6AF99445E44A
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: outbound
Contact: <sip:callee@192.0.2.1>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=1
;expires=3600
Content-Length: 0
The second registration in message 3 and 4 are similar other than the
Call-ID has changed, the reg-id is 2, and the route is set to the
secondary instead of the primary. They look like:
Jennings & Mahy Expires July 11, 2007 [Page 28]
Internet-Draft Client Initiated Connections in SIP January 2007
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnqr9bym
Max-Forwards: 70
From: Callee <sip:callee@example.com>;tag=755285EABDE2
To: Callee <sip:callee@example.com>
Call-ID: E05133BD26DD
CSeq: 1 REGISTER
Supported: path
Route: <sip:sec.example.com;lr;keepalive=stun>
Contact: <sip:callee@192.0.2.1>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=2
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.0.2.1;branch=z9hG4bKnqr9bym
From: Callee <sip:callee@example.com>;tag=755285EABDE2
To: Callee <sip:callee@example.com>;tag=49A9AD0B3F6A
Call-ID: E05133BD26DD
Supported: outbound
CSeq: 1 REGISTER
Contact: <sip:callee@192.0.2.1>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=1
;expires=3600
Contact: <sip:callee@192.0.2.1>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=2
;expires=3600
Content-Length: 0
The messages in the call flow are very normal. The only interesting
thing to note is that the INVITE in message 8 contains a Record-Route
header for the Secondary proxy, with its flow token.
Record-Route:
<sip:PQPbqQE+Ynf+tzRPD27lU6uxkjQ8LLUG@sec.example.com;lr>
The registrations in message 13 and 14 are the same as message 1 and
2 other than the Call-ID and tags have changed. Because these
messages will contain the same instance-id and reg-id as those in 1
and 2, this flow will partially supersede that for messages 1 and 2
and will be tried first by Primary.
Jennings & Mahy Expires July 11, 2007 [Page 29]
Internet-Draft Client Initiated Connections in SIP January 2007
10. Grammar
This specification defines new Contact header field parameters,
reg-id and +sip.instance. The grammar includes the definitions from
RFC 3261 [1] and includes the definition of uric from RFC 2396 [13].
Note: The "=/" syntax used in this ABNF indicates an extension of
the production on the left hand side.
The ABNF[14] is:
contact-params =/ c-p-reg / c-p-instance
c-p-reg = "reg-id" EQUAL 1*DIGIT ; 1 to 2**31
c-p-instance = "+sip.instance" EQUAL
LDQUOT "<" instance-val ">" RDQUOT
instance-val = *uric ; defined in RFC 2396
The value of the reg-id MUST NOT be 0 and MUST be less than 2**31.
11. Definition of 430 Flow Failed response code
This specification defines a new SIP response code '430 Flow Failed'.
This response code is used by an Edge Proxy to indicate to the
Authoritative Proxy that a specific flow to a UA instance has failed.
Other flows to the same instance could still succeed. The
Authoritative Proxy SHOULD attempt to forward to another target
(flow) with the same instance-id and AOR.
12. IANA Considerations
12.1. Contact Header Field
This specification defines a new Contact header field parameter
called reg-id in the "Header Field Parameters and Parameter Values"
sub-registry as per the registry created by [15]. The required
information is:
Header Field Parameter Name Predefined Reference
Values
____________________________________________________________________
Contact reg-id Yes [RFC AAAA]
[NOTE TO RFC Editor: Please replace AAAA with
Jennings & Mahy Expires July 11, 2007 [Page 30]
Internet-Draft Client Initiated Connections in SIP January 2007
the RFC number of this specification.]
12.2. SIP/SIPS URI Parameters
This specification arguments the "SIP/SIPS URI Parameters" sub-
registry as per the registry created by [16]. The required
information is:
Parameter Name Predefined Values Reference
____________________________________________
keepalive stun [RFC AAAA]
ob [RFC AAAA]
[NOTE TO RFC Editor: Please replace AAAA with
the RFC number of this specification.]
12.3. SIP Option Tag
This specification registers two new SIP option tags, as per the
guidelines in Section 27.1 of RFC 3261.
Name: outbound
Description: This option-tag is used to identify Registrars which
support extensions for Client Initiated Connections. A Registrar
places this option-tag in a Supported header to communicate the
Registrar's support for this extension to the registering User
Agent.
Name: sip-stun
Description: This option-tag is used to identify SIP servers which
can receive STUN requests described in the STUN NAT Keepalive
usage on the same ports they use to receive SIP messages.
12.4. Response Code
This section registers a new SIP Response Code, as per the guidelines
in Section 27.1 of RFC 3261.
Code: 430
Default Reason Phrase: Flow Failed
Reference: This document
12.5. Media Feature Tag
This section registers a new media feature tag, per the procedures
defined in RFC 2506 [17]. The tag is placed into the sip tree, which
is defined in RFC 3840 [6].
Media feature tag name: sip.instance
Jennings & Mahy Expires July 11, 2007 [Page 31]
Internet-Draft Client Initiated Connections in SIP January 2007
ASN.1 Identifier: New assignment by IANA.
Summary of the media feature indicated by this tag: This feature tag
contains a string containing a URN that indicates a unique identifier
associated with the UA instance registering the Contact.
Values appropriate for use with this feature tag: String.
The feature tag is intended primarily for use in the following
applications, protocols, services, or negotiation mechanisms: This
feature tag is most useful in a communications application, for
describing the capabilities of a device, such as a phone or PDA.
Examples of typical use: Routing a call to a specific device.
Related standards or documents: RFC XXXX
[[Note to IANA: Please replace XXXX with the RFC number of this
specification.]]
Security Considerations: This media feature tag can be used in ways
which affect application behaviors. For example, the SIP caller
preferences extension [8] allows for call routing decisions to be
based on the values of these parameters. Therefore, if an attacker
can modify the values of this tag, they might be able to affect the
behavior of applications. As a result, applications which utilize
this media feature tag SHOULD provide a means for ensuring its
integrity. Similarly, this feature tag should only be trusted as
valid when it comes from the user or user agent described by the tag.
As a result, protocols for conveying this feature tag SHOULD provide
a mechanism for guaranteeing authenticity.
13. Security Considerations
One of the key security concerns in this work is making sure that an
attacker cannot hijack the sessions of a valid user and cause all
calls destined to that user to be sent to the attacker.
The simple case is when there are no edge proxies. In this case, the
only time an entry can be added to the routing for a given AOR is
when the registration succeeds. SIP already protects against
attackers being able to successfully register, and this scheme relies
on that security. Some implementers have considered the idea of just
saving the instance-id without relating it to the AOR with which it
registered. This idea will not work because an attacker's UA can
impersonate a valid user's instance-id and hijack that user's calls.
Jennings & Mahy Expires July 11, 2007 [Page 32]
Internet-Draft Client Initiated Connections in SIP January 2007
The more complex case involves one or more edge proxies. When a UA
sends a REGISTER request through an Edge Proxy on to the registrar,
the Edge Proxy inserts a Path header field value. If the
registration is successfully authenticated, the registrar stores the
value of the Path header field. Later when the registrar forwards a
request destined for the UA, it copies the stored value of the Path
header field into the Route header field of the request and forwards
the request to the Edge Proxy.
The only time an Edge Proxy will route over a particular flow is when
it has received a Route header that has the flow identifier
information that it has created. An incoming request would have
gotten this information from the registrar. The registrar will only
save this information for a given AOR if the registration for the AOR
has been successful; and the registration will only be successful if
the UA can correctly authenticate. Even if an attacker has spoofed
some bad information in the Path header sent to the registrar, the
attacker will not be able to get the registrar to accept this
information for an AOR that does not belong to the attacker. The
registrar will not hand out this bad information to others, and
others will not be misled into contacting the attacker.
14. Operational Notes on Transports
RFC 3261 requires proxies, registrars, and UA to implement both TCP
and UDP but deployments can chose which protocols they want to use.
Deployments need to be careful in choosing what transports to use.
Many SIP features and extensions, such as large presence
subscriptions packages, result in SIP requests that can be too large
to be reasonably transported over UDP. RFC 3261 has an option of
when a request is too large for UDP, the device sending the request
can attempt to switch over to TCP. No known deployments currently
use this but it is important to note that when using outbound, this
will only work if the UA has formed both a UDP and TCP outbound
connection. The specification allows the UA to do this but in most
cases it will probably make more sense to only form TCP outbound
connection than forming both UDP and TCP. One of the key reasons
that many deployments choose not to use TCP has to do with the
difficulty of building proxies that can maintain a very large number
of active TCP connections. Many deployments today use SIP in such a
way that the message are small enough that they work over UDP but
they can not take advantage of all the functionality SIP offers.
Deployments that use only UDP outbound connections are going to fail
with sufficiently large SIP messages.
Jennings & Mahy Expires July 11, 2007 [Page 33]
Internet-Draft Client Initiated Connections in SIP January 2007
15. Requirements
This specification was developed to meet the following requirements:
1. Must be able to detect that a UA supports these mechanisms.
2. Support UAs behind NATs.
3. Support TLS to a UA without a stable DNS name or IP address.
4. Detect failure of a connection and be able to correct for this.
5. Support many UAs simultaneously rebooting.
6. Support a NAT rebooting or resetting.
7. Minimize initial startup load on a proxy.
8. Support architectures with edge proxies.
16. Changes
Note to RFC Editor: Please remove this whole section.
16.1. Changes from 06 Version
Added the section on operational selection of transports.
Fixed various editorial typos.
Put back in requirement flow token needs to be unique to flow as it
had accidentally been dropped in earlier version. This did not
change any of the flow token algorithms.
Reordered some of the text on STUN keepalive validation to make it
clearer to implementors. Did not change the actual algorithm or
requirements. Added note to explain how if the proxy changes, the
revalidation will happen.
16.2. Changes from 05 Version
Mention the relevance of the 'rport' parameter.
Change registrar verification so that only first-hop proxy and the
registrar need to support outbound. Other intermediaries in between
do not any more.
Relaxed flow-token language slightly. Instead of flow-token saving
specific UDP address/port tuples over which the request arrived, make
language fuzzy to save token which points to a 'logical flow' that is
known to deliver data to that specific UA instance.
Added comment that keepalive=stun could be added to Path.
Jennings & Mahy Expires July 11, 2007 [Page 34]
Internet-Draft Client Initiated Connections in SIP January 2007
Added comment that battery concerns could motivate longer TCP
keepalive intervals than the defaults.
Scrubbed document for avoidable lowercase mays, shoulds, and musts.
Added text about how Edge Proxies could determine they are the first
hop.
16.3. Changes from 04 Version
Moved STUN to a separate section. Reference this section from within
the relevant sections in the rest of the document.
Add language clarifying that UA MUST NOT send STUN without an
explicit indication the server supports STUN.
Add language describing that UA MUST stop sending STUN if it appears
the server does not support it.
Defined a 'sip-stun' option tag. UAs can optionally probe servers
for it with OPTIONS. Clarified that UAs SHOULD NOT put this in a
Proxy-Require. Explain that the first-hop MUST support this option-
tag.
Clarify that SIP/STUN in TLS is on the "inside". STUN used with
Sigcomp-compressed SIP is "outside" the compression layer for UDP,
but wrapped inside the well-known shim header for TCP-based
transports.
Clarify how to decide what a consecutive registration timer is. Flow
must be up for some time (default 120 seconds) otherwise previous
registration is not considered successful.
Change UAC MUST-->SHOULD register a flow for each member of outbound-
proxy-set.
Reworded registrar and proxy in some places (introduce the term
"Authoritative Proxy").
Loosened restrictions on always storing a complete Path vector back
to the registrar/authoritative proxy if a previous hop in the path
vector is reachable.
Added comment about reregistration typically happening over same flow
as original registration.
Changed 410 Gone to new response code 430 Flow Failed. Was going to
change this to 480 Temporarily Unavailable. Unfortunately this would
Jennings & Mahy Expires July 11, 2007 [Page 35]
Internet-Draft Client Initiated Connections in SIP January 2007
mean that the authoritative proxy deletes all flows of phones who use
480 for Do Not Disturb. Oops!
Restored sanity by restoring text which explains that registrations
with the same reg-id replace the old registration.
Added text about the 'ob' parameter which is used in Path header
field URIs to make sure that the previous proxy that added a Path
understood outbound processing. The registrar doesn't include
Supported: outbound unless it could actually do outbound processing
(ex: any Path headers have to have the 'ob' parameter).
Added some text describing what a registration means when there is an
instance-id, but no reg-id.
16.4. Changes from 03 Version
Added non-normative text motivating STUN vs. SIP PING, OPTIONS, and
Double CRLF. Added discussion about why TCP Keepalives are not
always available.
Explained more clearly that outbound-proxy-set can be "configured"
using any current or future, manual or automatic configuration/
discovery mechanism.
Added a sentence which prevents an Edge Proxy from forwarding back
over the flow over which the request is received if the request
happens to contain a flow token for that flow. This was an
oversight.
Updated example message flow to show a failover example using a new
dialog-creating request instead of a mid-dialog request. The old
scenario was leftover from before the outbound/gruu reorganization.
Fixed tags, Call-IDs, and branch parameters in the example messages.
Made the ABNF use the "=/" production extension mechanism recommended
by Bill Fenner.
Added a table in an appendix expanding the default flow recovery
timers.
Incorporated numerous clarifications and rewordings for better
comprehension.
Fixed many typos and spelling steaks.
Jennings & Mahy Expires July 11, 2007 [Page 36]
Internet-Draft Client Initiated Connections in SIP January 2007
16.5. Changes from 02 Version
Removed Double CRLF Keepalive
Changed ;sip-stun syntax to ;keepalive=stun
Fixed incorrect text about TCP keepalives.
16.6. Changes from 01 Version
Moved definition of instance-id from GRUU[25] draft to this draft.
Added tentative text about Double CRLF Keepalive
Removed pin-route stuff
Changed the name of "flow-id" to "reg-id"
Reorganized document flow
Described the use of STUN as a proper STUN usage
Added 'outbound' option-tag to detect if registrar supports outbound
16.7. Changes from 00 Version
Moved TCP keepalive to be STUN.
Allowed SUBSCRIBE to create flow mappings. Added pin-route option
tags to support this.
Added text about updating dialog state on each usage after a
connection failure.
17. Acknowledgments
Jonathan Rosenberg provided many comments and useful text. Dave Oran
came up with the idea of using the most recent registration first in
the proxy. Alan Hawrylyshen co-authored the draft that formed the
initial text of this specification. Additionally, many of the
concepts here originated at a connection reuse meeting at IETF 60
that included the authors, Jon Peterson, Jonathan Rosenberg, Alan
Hawrylyshen, and Paul Kyzivat. The TCP design team consisting of
Chris Boulton, Scott Lawrence, Rajnish Jain, Vijay K. Gurbani, and
Ganesh Jayadevan provided input and text. Nils Ohlmeier provided
many fixes and initial implementation experience. In addition,
thanks to the following folks for useful comments: Francois Audet,
Jennings & Mahy Expires July 11, 2007 [Page 37]
Internet-Draft Client Initiated Connections in SIP January 2007
Flemming Andreasen, Mike Hammer, Dan Wing, Srivatsa Srinivasan, Dale
Worely, Juha Heinanen, Eric Rescorla, Lyndsay Campbell, and Erkki
Koivusalo.
Appendix A. Default Flow Registration Backoff Times
The base-time used for the flow re-registration backoff times
described in Section 4.4.3 are configurable. If the base-time-all-
fail value is set to the default of 30 seconds and the base-time-not-
failed value is set to the default of 90 seconds, the following table
shows the resulting delay values.
+-------------------+--------------------+--------------------+
| # of reg failures | all flows unusable | >1 non-failed flow |
+-------------------+--------------------+--------------------+
| 0 | 0 secs | 0 secs |
| 1 | 30-60 secs | 90-180 secs |
| 2 | 1-2 mins | 3-6 mins |
| 3 | 2-4 mins | 6-12 mins |
| 4 | 4-8 mins | 12-24 mins |
| 5 | 8-16 mins | 15-30 mins |
| 6 or more | 15-30 mins | 15-30 mins |
+-------------------+--------------------+--------------------+
18. References
18.1. Normative References
[1] 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.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Registering Non-Adjacent Contacts",
RFC 3327, December 2002.
[4] Rosenberg, J., "Simple Traversal Underneath Network Address
Translators (NAT) (STUN)", draft-ietf-behave-rfc3489bis-05
(work in progress), October 2006.
[5] Leach, P., Mealling, M., and R. Salz, "A Universally Unique
IDentifier (UUID) URN Namespace", RFC 4122, July 2005.
Jennings & Mahy Expires July 11, 2007 [Page 38]
Internet-Draft Client Initiated Connections in SIP January 2007
[6] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating
User Agent Capabilities in the Session Initiation Protocol
(SIP)", RFC 3840, August 2004.
[7] Moats, R., "URN Syntax", RFC 2141, May 1997.
[8] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
Preferences for the Session Initiation Protocol (SIP)",
RFC 3841, August 2004.
[9] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
(SIP): Locating SIP Servers", RFC 3263, June 2002.
[10] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session
Initiation Protocol (SIP) for Symmetric Response Routing",
RFC 3581, August 2003.
[11] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-Hashing
for Message Authentication", RFC 2104, February 1997.
[12] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings",
RFC 3548, July 2003.
[13] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifiers (URI): Generic Syntax", RFC 2396,
August 1998.
[14] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 4234, October 2005.
[15] Camarillo, G., "The Internet Assigned Number Authority (IANA)
Header Field Parameter Registry for the Session Initiation
Protocol (SIP)", BCP 98, RFC 3968, December 2004.
[16] 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.
[17] Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag
Registration Procedure", BCP 31, RFC 2506, March 1999.
18.2. Informative References
[18] Petrie, D., "A Framework for Session Initiation Protocol User
Agent Profile Delivery", draft-ietf-sipping-config-framework-09
(work in progress), October 2006.
Jennings & Mahy Expires July 11, 2007 [Page 39]
Internet-Draft Client Initiated Connections in SIP January 2007
[19] Hakala, J., "Using National Bibliography Numbers as Uniform
Resource Names", RFC 3188, October 2001.
[20] Rosenberg, J., "Construction of the Route Header Field in the
Session Initiation Protocol (SIP)",
draft-rosenberg-sip-route-construct-02 (work in progress),
October 2006.
[21] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Service Route Discovery During
Registration", RFC 3608, October 2003.
[22] Boulton, C., "Best Current Practices for NAT Traversal for
SIP", draft-ietf-sipping-nat-scenarios-05 (work in progress),
June 2006.
[23] Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu,
Z., and J. Rosenberg, "Signaling Compression (SigComp)",
RFC 3320, January 2003.
[24] Surtees, A., "Implementer's Guide for SigComp",
draft-ietf-rohc-sigcomp-impl-guide-08 (work in progress),
October 2006.
[25] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", draft-ietf-sip-gruu-11 (work in progress),
October 2006.
Authors' Addresses
Cullen Jennings (editor)
Cisco Systems
170 West Tasman Drive
Mailstop SJC-21/2
San Jose, CA 95134
USA
Phone: +1 408 902-3341
Email: fluffy@cisco.com
Jennings & Mahy Expires July 11, 2007 [Page 40]
Internet-Draft Client Initiated Connections in SIP January 2007
Rohan Mahy (editor)
Plantronics
345 Encincal St
Santa Cruz, CA 95060
USA
Email: rohan@ekabal.com
Jennings & Mahy Expires July 11, 2007 [Page 41]
Internet-Draft Client Initiated Connections in SIP January 2007
Full Copyright Statement
Copyright (C) The Internet Society (2007).
This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Intellectual Property
The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; nor does it represent that it has
made any independent effort to identify any such rights. Information
on the procedures with respect to rights in RFC documents can be
found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use of
such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository at
http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.
Acknowledgment
Funding for the RFC Editor function is provided by the IETF
Administrative Support Activity (IASA).
Jennings & Mahy Expires July 11, 2007 [Page 42]
Html markup produced by rfcmarkup 1.129c, available from
https://tools.ietf.org/tools/rfcmarkup/