One document matched: draft-ietf-sip-outbound-11.txt
Differences from draft-ietf-sip-outbound-10.txt
Network Working Group C. Jennings, Ed.
Internet-Draft Cisco Systems
Updates: 3261,3327 R. Mahy, Ed.
(if approved) Plantronics
Intended status: Standards Track November 18, 2007
Expires: May 21, 2008
Managing Client Initiated Connections in the Session Initiation Protocol
(SIP)
draft-ietf-sip-outbound-11
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 May 21, 2008.
Copyright Notice
Copyright (C) The IETF Trust (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
Jennings & Mahy Expires May 21, 2008 [Page 1]
Internet-Draft Client Initiated Connections in SIP November 2007
Agents in this way. This specification defines behaviors for User
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 from the User Agent
to its Registrar.
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 . . . . . . . . . 8
3.4. Edge Proxies . . . . . . . . . . . . . . . . . . . . . . 9
3.5. Keepalive Technique . . . . . . . . . . . . . . . . . . . 11
3.5.1. CRLF Keepalive Technique . . . . . . . . . . . . . . . 11
3.5.2. STUN Keepalive Technique . . . . . . . . . . . . . . . 12
4. User Agent Mechanisms . . . . . . . . . . . . . . . . . . . . 12
4.1. Instance ID Creation . . . . . . . . . . . . . . . . . . 12
4.2. Registrations . . . . . . . . . . . . . . . . . . . . . . 13
4.2.1. Non Outbound Registrations . . . . . . . . . . . . . . 15
4.3. Sending Non-REGISTER Requests . . . . . . . . . . . . . . 15
4.4. Detecting Flow Failure . . . . . . . . . . . . . . . . . 16
4.4.1. Keepalive with CRLF . . . . . . . . . . . . . . . . . 17
4.4.2. Keepalive with STUN . . . . . . . . . . . . . . . . . 18
4.5. Flow Recovery . . . . . . . . . . . . . . . . . . . . . . 18
5. Edge Proxy Mechanisms . . . . . . . . . . . . . . . . . . . . 19
5.1. Processing Register Requests . . . . . . . . . . . . . . 19
5.2. Generating Flow Tokens . . . . . . . . . . . . . . . . . 20
5.3. Forwarding Non-REGISTER Requests . . . . . . . . . . . . 20
5.4. Edge Proxy Keepalive Handling . . . . . . . . . . . . . . 21
6. Registrar Mechanisms: Processing REGISTER Requests . . . . . . 21
7. Authoritative Proxy Mechanisms: Forwarding Requests . . . . . 23
8. STUN Keepalive Processing . . . . . . . . . . . . . . . . . . 24
8.1. 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
Jennings & Mahy Expires May 21, 2008 [Page 2]
Internet-Draft Client Initiated Connections in SIP November 2007
13. Security Considerations . . . . . . . . . . . . . . . . . . . 32
14. Operational Notes on Transports . . . . . . . . . . . . . . . 33
15. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 34
16. Changes . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
16.1. Changes from 09 Version . . . . . . . . . . . . . . . . . 34
16.2. Changes from 08 Version . . . . . . . . . . . . . . . . . 34
16.3. Changes from 07 Version . . . . . . . . . . . . . . . . . 35
16.4. Changes from 06 Version . . . . . . . . . . . . . . . . . 35
16.5. Changes from 05 Version . . . . . . . . . . . . . . . . . 35
16.6. Changes from 04 Version . . . . . . . . . . . . . . . . . 36
16.7. Changes from 03 Version . . . . . . . . . . . . . . . . . 37
16.8. Changes from 02 Version . . . . . . . . . . . . . . . . . 38
16.9. Changes from 01 Version . . . . . . . . . . . . . . . . . 38
16.10. Changes from 00 Version . . . . . . . . . . . . . . . . . 38
17. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 38
Appendix A. Default Flow Registration Backoff Times . . . . . . . 39
18. References . . . . . . . . . . . . . . . . . . . . . . . . . . 39
18.1. Normative References . . . . . . . . . . . . . . . . . . 39
18.2. Informative References . . . . . . . . . . . . . . . . . 40
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 41
Intellectual Property and Copyright Statements . . . . . . . . . . 43
Jennings & Mahy Expires May 21, 2008 [Page 3]
Internet-Draft Client Initiated Connections in SIP November 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 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 a SIP User Agent behind
such a firewall or NAT to receive inbound traffic associated with
registrations or dialogs that it initiates.
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 Transport Layer Security (TLS) [18] 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 (as defined in Section 22 of RFC
3261) over that TLS connection.
The key idea of this specification is that when a UA sends a REGISTER
or a dialog-forming 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 incoming requests that need to go
to this UA in the context of the registration or dialog.
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 register over multiple flows at the same time. This
specification also defines multiple keepalive schemes. 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
Jennings & Mahy Expires May 21, 2008 [Page 4]
Internet-Draft Client Initiated Connections in SIP November 2007
document are to be interpreted as described in RFC 2119 [2].
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 concurrent 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
The mechanisms defined in this document are useful in several
scenarios 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
Jennings & Mahy Expires May 21, 2008 [Page 5]
Internet-Draft Client Initiated Connections in SIP November 2007
reliability. Each registration includes the instance-id for the UA
and a reg-id label that is different for each flow. The registrar
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 whether a UA is creatin a new flow or refreshing or
replacing an old one, possibly 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 to deliver a request 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 can use a simple periodic message as a keepalive mechanism to
keep their flow to the proxy or registrar alive. For connection
oriented transports such as TCP this is based on CRLF or a transport
specific keepalive while for transports that are not connection
oriented this is accomplished by using a SIP-specific usage profile
of STUN (Session Traversal Utilities for NAT) [3].
The UA can also ask its first hop proxy to use an specific flow for
subsequent messages when sending a dialog-forming request. This
allows the UA to setup a subscription dialog for the SIP
configuration package [19] before the UA registers.
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
Jennings & Mahy Expires May 21, 2008 [Page 6]
Internet-Draft Client Initiated Connections in SIP November 2007
reason.
For clarity, here is an example. Bob's UA creates a new TCP flow to
the registrar and sends the following REGISTER request.
REGISTER sip:example.com;keep SIP/2.0
Via: SIP/2.0/TCP 192.168.0.2;rport;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, outbound
Contact: <sip:line1@192.168.0.2;transport=tcp>; reg-id=1;
;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000A95A0E128>"
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:00000000-0000-1000-8000-000A95A0E128") 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 existing flow, instead of resolving the Contact URI using the
procedures in RFC 3263 [4] and trying to form a new flow to that
contact.
As described in the next section, if the proxy has multiple flows
that all go to this UA, the proxy can choose any one of the
registration bindings for this AOR that has the same instance-id as
the selected UA.
Jennings & Mahy Expires May 21, 2008 [Page 7]
Internet-Draft Client Initiated Connections in SIP November 2007
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
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 authoritative 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 [20] 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
additional 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 [19], from adding other ways
for a User Agent to discover its outbound-proxy-set.
+-------------------+
| Domain |
| Logical Proxy/Reg |
| |
|+-----+ +-----+|
||Host1| |Host2||
|+-----+ +-----+|
+---\------------/--+
\ /
\ /
\ /
\ /
+------+
| User |
| Agent|
+------+
Jennings & Mahy Expires May 21, 2008 [Page 8]
Internet-Draft Client Initiated Connections in SIP November 2007
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 address and AOR in the UA. If
the AOR is alice@example.com, the outbound-proxy-set might look
something like "sip:primary.example.com;keep" and "sip:
secondary.example.com;keep". The "keep" tag indicates that a SIP
server will respond correctly to the mandatory to implement keepalive
mechanisms 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 4.5. 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 [5] 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
Jennings & Mahy Expires May 21, 2008 [Page 9]
Internet-Draft Client Initiated Connections in SIP November 2007
between the UA and the Edge Proxy.
+---------+
|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 authenticates with the
registrar successfully, 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 [5].
Jennings & Mahy Expires May 21, 2008 [Page 10]
Internet-Draft Client Initiated Connections in SIP November 2007
3.5. Keepalive Technique
This document describes three keepalive mechanisms. Each of these
mechanisms uses a client-to-server "ping" keepalive and a
corresponding server-to-client "pong" message. This ping-pong
sequence allows the client, and optionally the server, to tell if its
flow is still active and useful for SIP traffic. The server responds
to pings by sending pongs. If the client does not receive a pong in
response to its ping, it declares the flow dead and opens a new flow
in its place.
This document also suggests timer values for two of these client
keepalive mechanisms. These timer values were chosen to keep most
NAT and firewall bindings open, to detect unresponsive servers within
2 minutes, and to prevent the avalanche restart problem. However,
the client may choose different timer values to suit its needs, for
example to optimize battery life. In some environments, the server
can also keep track of the time since a ping was received over a flow
to guess the likelihood that the flow is still useful for delivering
SIP messages. In this case, the server provides an indicator (the
'timed-keepalives' parameter) that the server requires the client to
use the suggested timer values.
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
back-off mechanism described in Section 4 to provide congestion
relief when a large number of agents simultaneously reboot.
A keepalive mechanism needs to keep NAT bindings refreshed; for
connections, it also needs to detect failure of a connection; and for
connectionless transports, it needs to detect flow failures including
changes to the NAT public mapping. For connection oriented
transports such as TCP and SCTP, this specification describes a
keepalive approach based on sending CRLFs, and for TCP, a usage of
TCP transport-layer keepalives. For connectionless transport, such
as UDP, this specification describes using STUN [3] over the same
flow as the SIP traffic to perform the keepalive.
UAs are also free to use native transport keepalives, however the UA
application may not be able to set these timers on a per-connection
basis, and the server certainly cannot make any assumption about what
values are used. Use of native transport keepalives is therefore
outside the scope of this document.
3.5.1. CRLF Keepalive Technique
This approach can only be used with connection-oriented transports
such as TCP or SCTP. The client periodically sends a double-CRLF
Jennings & Mahy Expires May 21, 2008 [Page 11]
Internet-Draft Client Initiated Connections in SIP November 2007
(the "ping") then waits to receive a single CRLF (the "pong"). If
the client does not receive a "pong" within an appropriate amount of
time, it considers the flow failed.
3.5.2. STUN Keepalive Technique
This technique can only be used for connection-less transports, such
as UDP.
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, STUN requests are sent over the same flow that is being used
for the SIP traffic. The proxy or registrar acts as a Session
Traversal Utilities for NAT (STUN) [3] 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.
Approaches using SIP requests were abandoned because many believed
that good performance and full backwards compatibility using this
method were mutually exclusive.
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 create a UUID URN [6] 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 [6] 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
Jennings & Mahy Expires May 21, 2008 [Page 12]
Internet-Draft Client Initiated Connections in SIP November 2007
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 on this device.
If a URN scheme other than UUID is used, the UA MUST only use URNs
for which an IETF consensus RFC defines how the specific URN needs to
be constructed and used in the sip.instance Contact parameter for
outbound behavior.
To convey its instance-id in both requests and responses, the UA
includes a "sip.instance" media feature tag as a UA characteristic
[7] . As described in [7], 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 [8]. One case
where a UA may not want to include the sip.instance media feature tag
at all is when it is making an anonymous request or some other
privacy concern requires that the UA not reveal its identity.
RFC 3840 [7] 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 [7] and in the caller
preferences specification [9]. 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 [8]. 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
MUST 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 lexicographically
different from previous registrations.
4.2. 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
Jennings & Mahy Expires May 21, 2008 [Page 13]
Internet-Draft Client Initiated Connections in SIP November 2007
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. (The
UA could limit the number of flows formed to conserve battery power,
for example). All of these REGISTER requests will use the same
Call-ID. [OPEN ISSUE: This is for consistency with GRUU, Section
5.1 paragraph 5. Is this a bad idea? Alternatively GRUU could check
all reg-ids and preserve temporary GRUU if a registration used the
same Call-ID as used by any of the current bindings for the same
instance.] 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 [21].
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.
These ordinary registration requests include a distinct reg-id
parameter in 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 registrations.
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 Require header field when registering with EP2,
in order to insure consistency. Similarly, if the registration
through EP1 did not support outbound, the UA might not register
with EP2 at all.
The UAC MUST indicate that it supports the Path header [5] 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.
Jennings & Mahy Expires May 21, 2008 [Page 14]
Internet-Draft Client Initiated Connections in SIP November 2007
The UAC MAY examine successful registration responses for the
presence of an 'outbound' option-tag in a Require header field value.
Presence of this option-tag indicates that the registrar is compliant
with this specification, and that any edge proxies which needed to
participate are also compliant. If the registrar did not support
outbound, the UA may have unintentionally registered an unroutable
contact. It is the responsiblity of the UA to remove any
inappropriate Contacts.
Note that the UA needs to honor 503 (Service Unavailable) responses
to registrations as described in RFC 3261 and RFC 3263 [4]. 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.
4.2.1. Non Outbound Registrations
A User Agent MUST NOT include a reg-id header parameter in the
Contact header field of a registration with a non-zero expiration, 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.)
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 Non-REGISTER 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 [22] (Service
Route) and [21].
The UA performs normal DNS resolution on the next hop URI (as
described in RFC 3263 [4]) to find a protocol, IP address, and port.
Jennings & Mahy Expires May 21, 2008 [Page 15]
Internet-Draft Client Initiated Connections in SIP November 2007
For protocols that don't use TLS, 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.
If the UA is sending a dialog-forming request, and wants all
subsequent requests in the dialog to arrive over the same flow, the
UA adds an 'ob' parameter to its Contact header. Typically this is
desirable, but it is not necessary for example if the Contact is a
GRUU [23]. The flow used for the request is typically the same flow
the UA registered over, but it could be a new flow, for example the
initial subcription dialog for the configuration framework [19] needs
to exist before registration.
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 [24].
******
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.
When the outbound-proxy-set contains the "timed-keepalives"
parameter, the UA MUST send its keepalives according to the time
periods described in this section. The server can specify this so
the server can detect liveness of the client within a predictable
time scale. If the parameter is not present, the UA can send
keepalives at its discretion.
The time between each keepalive request when using non connection
based transports such as UDP SHOULD be a random number between 24 and
29 seconds while for connection based transports such as TCP it
SHOULD be a random number between 95 and 120 seconds. These times
MAY be configurable. To clarify, the random number will be different
for each request. Issues such as battery consumption might motivate
longer keepalive intervals. If the 'timed-keepalives' parameter is
Jennings & Mahy Expires May 21, 2008 [Page 16]
Internet-Draft Client Initiated Connections in SIP November 2007
set on the outbound-proxy-set, the UA MUST use these recommended
timer values.
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, as 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.
The client needs to perform normal RFC 3263 [4] SIP DNS resolution on
the URI from the outbound-proxy-set to pick a transport. Once a
transport is selected, if the 'keep' parameter is present in the URI,
the UA selects the keepalive approach that is recommended for that
transport.
4.4.1. Keepalive with CRLF
This approach MUST only be used with connection oriented transports
such as TCP or SCTP.
A User Agent that forms flows, checks if the configured URI to which
the UA is connecting resolves to a stream-based transport (ex: TCP
and TLS over TCP) and has a 'keep' URI parameter (defined in
Section 12). If the parameter is present, the UA can send keep
alives as described in this section.
For this mechanism, the client "ping" is a double-CRLF sequence, and
the server "pong" is a single CRLF, as defined in the ABNF below:
CRLF = CR LF
double-CRLF = CR LF CR LF
CR = 0x0d
LF = 0x0a
The ping and pong need to be sent between SIP messages and cannot be
sent in the middle of a SIP message. If sending over TLS, the CRLFs
are sent inside the TLS protected channel. If sending over a SigComp
[25] compressed data stream, the CRLF keepalives are sent inside the
compressed stream. The double CRLF is considered a single SigComp
Jennings & Mahy Expires May 21, 2008 [Page 17]
Internet-Draft Client Initiated Connections in SIP November 2007
message. The specific mechanism for representing these characters is
an implementation specific matter to be handled by the SigComp
compressor at the sending end.
If a pong is not received within 10 seconds then the client MUST
treat the flow as failed. Clients MUST support this CRLF keepalive.
4.4.2. Keepalive with STUN
This approach MUST only be used with connection-less transports, such
as UDP.
A User Agent that forms flows, checks if the configured URI to which
the UA is connecting resolve to use the UDP transport, and has a
'keep' URI parameter (defined in Section 12). If the parameter is
present, the UA can periodically perform keepalive checks by sending
STUN [3] Binding Requests over the flow as described in Section 8.
Clients MUST support STUN based keepalives.
If a STUN Binding Error Response is received, or if no Binding
Response is received after 7 retransmissions (16 times the STUN "RTO"
timer--RTO is an estimate of round-trip time), the UA considers the
flow failed. If the XOR-MAPPED-ADDRESS in the STUN Binding Response
changes, the UA MUST treat this event as a failure on the flow.
4.5. 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
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 if keepalives are in use on this flow, at least one
subsequent keepalive response was received.
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
Jennings & Mahy Expires May 21, 2008 [Page 18]
Internet-Draft Client Initiated Connections in SIP November 2007
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 three times are:
o max-time with a default of 1800 seconds
o base-time (if all failed) with a default of 30 seconds
o base-time (if all have not failed) with a default of 90 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.
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 [5]. 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.
Jennings & Mahy Expires May 21, 2008 [Page 19]
Internet-Draft Client Initiated Connections in SIP November 2007
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. A stateless
example is 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 cannot be modified by attackers.
Example Algorithm: 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 [26]. The concatenation of the
HMAC and S are base64 encoded, as defined in [27], and used as the
flow identifier. When using IPv4 addresses, this will result in a
32-octet identifier.
5.3. Forwarding Non-REGISTER 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 field value contains a flow token,
the proxy decodes the flow token and compares the flow in the flow
token with the source of the request to determine if this is an
"incoming" or "outgoing" request.
If the flow in the flow token in the topmost Route header field value
matches the source of the request, the request in an "outgoing"
request. For an "outgoing" request, the edge proxy just removes the
Route header and continues processing the request. Otherwise, this
is an "incoming" request. For an incoming request, the proxy removes
the Route header field value and forwards 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.
Proxies which used the example algorithm described in this document
to form a flow token follow the procedures below to determine the
correct flow.
Jennings & Mahy Expires May 21, 2008 [Page 20]
Internet-Draft Client Initiated Connections in SIP November 2007
Example Algorithm: 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 that 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 specific procedures at the edge proxy 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 add a Record-Route header with a flow token is one way
to ensure that mid-dialog requests are routed over the correct flow.
The Edge Proxy can use the presence of the "ob" parameter in dialog-
forming requests in the UAC's Contact URI to determine if it should
add a flow token.
5.4. Edge Proxy Keepalive Handling
All edge proxies compliant with this specification MUST implement
support for STUN NAT Keepalives on its SIP UDP ports as described in
Section 8.
When a server receives a double CRLF sequence on a connection
oriented transport such as TCP or SCTP, it MUST immediately respond
with a single CRLF over the same connection.
6. Registrar Mechanisms: Processing REGISTER Requests
This specification updates the definition of a binding in RFC 3261
[1] Section 10 and RFC 3327 [5] Section 5.3.
Registrars which implement this specification MUST support the Path
header mechanism RFC 3327 [5].
When receiving a REGISTER request, the registrar first checks from
its Via header field if the registrar is the first hop or not. If
the registrar is not the first hop, it examines the Path header of
the request. If the Path header field is missing or it exists but
the first URI does not have an 'ob' URI parameter, the registrar MUST
ignore the reg-id parameter of the Contact header.
A Contact header field value with an instance-id but no reg-id is
Jennings & Mahy Expires May 21, 2008 [Page 21]
Internet-Draft Client Initiated Connections in SIP November 2007
valid (this combination can be used in the GRUU [23] specification),
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. If the Contact
header contains more than one header field value with a non-zero
expiration and a 'reg-id' parameter, the entire registration SHOULD
be rejected with a 400 Bad Request response. If the Contact header
did not contain a 'reg-id' parameter or if that parameter became
ignored (as described above) the registrar MUST NOT include the
'outbound' option-tag in the Require header field of its response.
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. The Registrar MAY be configured with
local policy to reject any registrations that do not include the
instance-id and reg-id, or with Path header field values that do not
contain the 'ob' parameter. If the Contact header field does not
contain a '+sip.instance' media feature parameter, the registrar
processes the request using the Contact binding rules in RFC 3261
[1].
When a '+sip.instance' media feature parameter is present in a
Contact header field of a REGISTER request (after the Contact header
validation as described above), 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. The registrar MUST store in the binding the Contact URI,
all the Contact head field parameters, and any Path header field
values and SHOULD also store the time at which the binding was last
updated. (Even though the Contact URI is not used for binding
comparisons, it is still needed by the authoritative proxy to form
the target set.) The Registrar MUST include the 'outbound' option-
tag (defined in Section 12.1) in a Require header field value in its
response to the REGISTER request.
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 the 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.
If the registrar receives a re-registration for a specific
combination of AOR, instance-id and reg-id values, the registrar MUST
Jennings & Mahy Expires May 21, 2008 [Page 22]
Internet-Draft Client Initiated Connections in SIP November 2007
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.
To be compliant with this specification, registrars which can receive
SIP requests directly from a UAC without intervening edge proxies
MUST implement the same keepalive mechanisms as Edge Proxies
(Section 5.4).
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.
o 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 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 and populate the Route header in the request.
If the proxy doubles as a registrar and stored information about the
flow that created the binding, then the proxy MUST send the request
over the same 'logical flow' saved with the binding, since that flow
is known to deliver data to the specific target UA instance's network
flow that was saved with the binding.
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
Jennings & Mahy Expires May 21, 2008 [Page 23]
Internet-Draft Client Initiated Connections in SIP November 2007
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.
8. STUN Keepalive Processing
This section describes changes to the SIP transport layer that allow
SIP and the STUN [3] Binding Requests to be mixed over the same flow.
This constitues a new STUN usage. The STUN messages are used to
verify that connectivity is still available over a UDP 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 Binding Error Responses. The UAC sends
Binding Requests over the same UDP flow that is used for sending SIP
messages. These Binding Requests do not require any STUN attributes
except the XOR-MAPPED-ADDRESS and never use any form of
authentication. The UAS, proxy, or registrar responds to a valid
Binding Request with a Binding Response which MUST include the XOR-
MAPPED-ADDRESS attribute.
If a server compliant to this section receives SIP requests on a
given interface and UDP port, it MUST also provide a limited version
of a STUN server on the same interface and UDP port.
It is easy to distinguish STUN and SIP packets sent over UDP,
because the first octet of a STUN Binding method has a value of 0
or 1 while the first octet of a SIP message is never a 0 or 1.
When a URI is created that refers to a SIP node that supports STUN as
described in this section, the 'keep' URI parameter, as defined in
Section 12 SHOULD be added to the URI. 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.
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 incorrect configuration) despite these warnings, the
node could be blacklisted for UDP traffic.
A SIP node MUST NOT send STUN requests over a flow unless it has an
Jennings & Mahy Expires May 21, 2008 [Page 24]
Internet-Draft Client Initiated Connections in SIP November 2007
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 'keep' parameter, or receiving
the parameter in the Path header of the edge proxy, 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.
Typically, a SIP node first sends a SIP request and waits to
receive a 200-class 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 request to the
target.
Once a flow is established, 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.
The SIP keepalive STUN usage requires no backwards compatibility with
RFC 3489 [11].
8.1. Use with Sigcomp
When STUN is used together with SigComp [25] compressed SIP messages
over the same flow. 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.
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 [12] encoded text messages, 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 Binding method 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.
Jennings & Mahy Expires May 21, 2008 [Page 25]
Internet-Draft Client Initiated Connections in SIP November 2007
9. Example Message Flow
[----example.com domain------]
Bob EP1 EP2 Proxy Alice
| | | | |
1)|-REGISTER->| | | |
2)| |---REGISTER-->| |
3)| |<----200 OK---| |
4)|<-200 OK---| | | |
5)|----REGISTER---->| | |
6)| | |--REG-->| |
7)| | |<-200---| |
8)|<----200 OK------| | |
| | | | |
| CRASH X | | |
| Reboot | | |
9)| | | |<-INVITE-|
10)| |<---INVITE----| |
11)| |----430------>| |
12)| | |<-INVITE| |
13)|<---INVITE-------| | |
14)|----200 OK------>| | |
15)| | |200 OK->| |
16)| | | |-200 OK->|
17)| | | |<-ACK----|
18)| | |<-ACK---| |
19)|<---ACK----------| | |
| | | | |
20)|--2CRLF->X | | | |
| | | | |
21)|-REGISTER->| | | |
22)|<-200 OK---| | | |
| | | | |
[TODO FIX example] 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 May 21, 2008 [Page 26]
Internet-Draft Client Initiated Connections in SIP November 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;keep-stun" and
"sip:sec.example.com;lr;keep-stun". The Callee REGISTER in message
Jennings & Mahy Expires May 21, 2008 [Page 27]
Internet-Draft Client Initiated Connections in SIP November 2007
(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;keep-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 May 21, 2008 [Page 28]
Internet-Draft Client Initiated Connections in SIP November 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;keep-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 May 21, 2008 [Page 29]
Internet-Draft Client Initiated Connections in SIP November 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 3986 [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 - 1)
c-p-instance = "+sip.instance" EQUAL
LDQUOT "<" instance-val ">" RDQUOT
instance-val = *uric ; defined in RFC 3986
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 No [RFC AAAA]
[NOTE TO RFC Editor: Please replace AAAA with
Jennings & Mahy Expires May 21, 2008 [Page 30]
Internet-Draft Client Initiated Connections in SIP November 2007
the RFC number of this specification.]
12.2. SIP/SIPS URI Parameters
This specification augments the "SIP/SIPS URI Parameters" sub-
registry as per the registry created by [16]. The required
information is:
Parameter Name Predefined Values Reference
____________________________________________
keep No [RFC AAAA]
timed-keepalives No [RFC AAAA]
ob No [RFC AAAA]
[NOTE TO RFC Editor: Please replace AAAA with
the RFC number of this specification.]
12.3. SIP Option Tag
This specification registers a new SIP option tag, as per the
guidelines in Section 27.1 of RFC 3261.
Name: outbound
Description: This option-tag is used to identify UAs and 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, and vice versa.
12.4. Response Code
This section registers a new SIP Response Code, as per the guidelines
in Section 27.4 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 [7].
Media feature tag name: sip.instance
ASN.1 Identifier: New assignment by IANA.
Jennings & Mahy Expires May 21, 2008 [Page 31]
Internet-Draft Client Initiated Connections in SIP November 2007
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 [9] 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. Note that
the intent is not to prevent existing active attacks on SIP UDP and
TCP traffic, but to insure that no new attacks are added by
introducing the outbound mechanism.
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 May 21, 2008 [Page 32]
Internet-Draft Client Initiated Connections in SIP November 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.
The Security Considerations discussed in [1] and [5] are also
relevant to this document. For the security considerations of
generating flow tokens, please also see Section 5.2. A discussion of
preventing the avalanche restart problem is in Section 4.5.
This document does not change the mandatory to implement security
mechanisms in SIP. User Agents are already required to implement
Digest authentication while support of TLS is recommended; proxy
servers are already required to implement Digest and TLS.
14. Operational Notes on Transports
This entire section is non-normative.
RFC 3261 requires proxies, registrars, and User Agents to implement
both TCP and UDP but deployments can chose which transport 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 notification bodies, result in SIP requests that can be too
large to be reasonably transported over UDP. RFC 3261 states that
when a request is too large for UDP, the device sending the request
attempts to switch over to TCP. No known deployments currently use
this feature but it is important to note that when using outbound,
this will only work if the UA has formed both UDP and TCP outbound
flows. This specification allows the UA to do so but in most cases
Jennings & Mahy Expires May 21, 2008 [Page 33]
Internet-Draft Client Initiated Connections in SIP November 2007
it will probably make more sense for the UA to form a TCP outbound
connection only, rather than forming both UDP and TCP flows. 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 messages 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.
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 09 Version
Make outbound consistent with the latest version of STUN 3489bis
draft. The STUN keepalive section of outbound is now a STUN usage
(much less formal).
Fixed references.
16.2. Changes from 08 Version
UAs now include the 'ob' parameter in their Contact header for non-
REGISTER requests, as a hint to the Edge Proxy (so the EP can Record-
Route with a flow-token for example).
Switched to CRLF for keepalives of connection-oriented transports
after brutal consensus at IETF 68.
Added timed-keepalive parameter and removed the unnecessary keep-tcp
param, per consensus at IETF68.
Jennings & Mahy Expires May 21, 2008 [Page 34]
Internet-Draft Client Initiated Connections in SIP November 2007
Removed example "Algorithm 1" which only worked over SIPS, per
consensus at IETF68.
Deleted text about probing and validating with options, per consensus
at IETF68.
Deleted provision for waiting 120 secs before declaring flow stable,
per consensus at IETF68.
fixed example UUIDs
16.3. Changes from 07 Version
Add language to show the working group what adding CRLF keepalives
would look like.
Changed syntax of keep-alive=stun to keep-stun so that it was easier
to support multiple tags in the same URI.
16.4. 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.
16.5. 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 keep-stun could be added to Path.
Jennings & Mahy Expires May 21, 2008 [Page 35]
Internet-Draft Client Initiated Connections in SIP November 2007
Added comment that battery concerns could motivate longer TCP
keepalive intervals than the defaults.
Scrubbed document for avoidable lowercase may, should, and must.
Added text about how Edge Proxies could determine they are the first
hop.
16.6. 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 re-registration 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 May 21, 2008 [Page 36]
Internet-Draft Client Initiated Connections in SIP November 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.
16.7. 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 fail-over 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 May 21, 2008 [Page 37]
Internet-Draft Client Initiated Connections in SIP November 2007
16.8. Changes from 02 Version
Removed Double CRLF Keepalive
Changed ;sip-stun syntax to ;keepalive=stun
Fixed incorrect text about TCP keepalives.
16.9. Changes from 01 Version
Moved definition of instance-id from GRUU[23] 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
16.10. 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, Erkki Koivusalo, and Byron Campben 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
Jennings & Mahy Expires May 21, 2008 [Page 38]
Internet-Draft Client Initiated Connections in SIP November 2007
folks for useful comments: Francois Audet, Flemming Andreasen, Mike
Hammer, Dan Wing, Srivatsa Srinivasan, Dale Worely, Juha Heinanen,
Eric Rescorla, Lyndsay Campbell, Christer Holmberg, Kevin Johns,
Jeroen van Bemmel, and Derek MacDonald.
Appendix A. Default Flow Registration Backoff Times
The base-time used for the flow re-registration backoff times
described in Section 4.5 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] Rosenberg, J., "Simple Traversal Underneath Network Address
Translators (NAT) (STUN)", draft-ietf-behave-rfc3489bis-12
(work in progress), November 2007.
[4] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
(SIP): Locating SIP Servers", RFC 3263, June 2002.
[5] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Registering Non-Adjacent Contacts",
RFC 3327, December 2002.
Jennings & Mahy Expires May 21, 2008 [Page 39]
Internet-Draft Client Initiated Connections in SIP November 2007
[6] Leach, P., Mealling, M., and R. Salz, "A Universally Unique
IDentifier (UUID) URN Namespace", RFC 4122, July 2005.
[7] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating
User Agent Capabilities in the Session Initiation Protocol
(SIP)", RFC 3840, August 2004.
[8] Moats, R., "URN Syntax", RFC 2141, May 1997.
[9] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
Preferences for the Session Initiation Protocol (SIP)",
RFC 3841, August 2004.
[10] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session
Initiation Protocol (SIP) for Symmetric Response Routing",
RFC 3581, August 2003.
[11] Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy, "STUN
- Simple Traversal of User Datagram Protocol (UDP) Through
Network Address Translators (NATs)", RFC 3489, March 2003.
[12] Yergeau, F., "UTF-8, a transformation format of ISO 10646",
STD 63, RFC 3629, November 2003.
[13] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Sy ntax", STD 66, RFC 3986,
January 2005.
[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] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS)
Protocol Version 1.1", RFC 4346, April 2006.
Jennings & Mahy Expires May 21, 2008 [Page 40]
Internet-Draft Client Initiated Connections in SIP November 2007
[19] Petrie, D., "A Framework for Session Initiation Protocol User
Agent Profile Delivery", draft-ietf-sipping-config-framework-13
(work in progress), October 2007.
[20] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
specifying the location of services (DNS SRV)", RFC 2782,
February 2000.
[21] Rosenberg, J., "Construction of the Route Header Field in the
Session Initiation Protocol (SIP)",
draft-rosenberg-sip-route-construct-02 (work in progress).
[22] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Service Route Discovery During
Registration", RFC 3608, October 2003.
[23] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", draft-ietf-sip-gruu-15 (work in progress),
October 2007.
[24] Boulton, C., "Best Current Practices for NAT Traversal for
SIP", draft-ietf-sipping-nat-scenarios-07 (work in progress),
July 2007.
[25] Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu,
Z., and J. Rosenberg, "Signaling Compression (SigComp)",
RFC 3320, January 2003.
[26] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-Hashing
for Message Authentication", RFC 2104, February 1997.
[27] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings",
RFC 4648, 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 May 21, 2008 [Page 41]
Internet-Draft Client Initiated Connections in SIP November 2007
Rohan Mahy (editor)
Plantronics
345 Encincal St
Santa Cruz, CA 95060
USA
Email: rohan@ekabal.com
Jennings & Mahy Expires May 21, 2008 [Page 42]
Internet-Draft Client Initiated Connections in SIP November 2007
Full Copyright Statement
Copyright (C) The IETF Trust (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, THE IETF TRUST 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 May 21, 2008 [Page 43]
| PAFTECH AB 2003-2026 | 2026-04-21 21:25:50 |