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rfc:rfc5626

Network Working Group C. Jennings, Ed. Request for Comments: 5626 Cisco Systems Updates: 3261, 3327 R. Mahy, Ed. Category: Standards Track Unaffiliated

                                                         F. Audet, Ed.
                                                            Skype Labs
                                                          October 2009
               Managing Client-Initiated Connections
              in the Session Initiation Protocol (SIP)

Abstract

 The Session Initiation Protocol (SIP) allows proxy servers to
 initiate TCP connections or to send asynchronous UDP datagrams to
 User Agents in order to deliver requests.  However, in a large number
 of real deployments, many practical considerations, such as the
 existence of firewalls and Network Address Translators (NATs) or the
 use of TLS with server-provided certificates, prevent servers from
 connecting to User 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.

Status of This Memo

 This document specifies an Internet standards track protocol for the
 Internet community, and requests discussion and suggestions for
 improvements.  Please refer to the current edition of the "Internet
 Official Protocol Standards" (STD 1) for the standardization state
 and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

 Copyright (c) 2009 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (http://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of

Jennings, et al. Standards Track [Page 1] RFC 5626 Client-Initiated Connections in SIP October 2009

 the Trust Legal Provisions and are provided without warranty as
 described in the BSD License.
 This document may contain material from IETF Documents or IETF
 Contributions published or made publicly available before November
 10, 2008.  The person(s) controlling the copyright in some of this
 material may not have granted the IETF Trust the right to allow
 modifications of such material outside the IETF Standards Process.
 Without obtaining an adequate license from the person(s) controlling
 the copyright in such materials, this document may not be modified
 outside the IETF Standards Process, and derivative works of it may
 not be created outside the IETF Standards Process, except to format
 it for publication as an RFC or to translate it into languages other
 than English.

Table of Contents

 1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
 2.  Conventions and Terminology  . . . . . . . . . . . . . . . . .  5
   2.1.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .  5
 3.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .  6
   3.1.  Summary of Mechanism . . . . . . . . . . . . . . . . . . .  6
   3.2.  Single Registrar and UA  . . . . . . . . . . . . . . . . .  7
   3.3.  Multiple Connections from a User Agent . . . . . . . . . .  8
   3.4.  Edge Proxies . . . . . . . . . . . . . . . . . . . . . . . 10
   3.5.  Keep-Alive Technique . . . . . . . . . . . . . . . . . . . 11
     3.5.1.  CRLF Keep-Alive Technique  . . . . . . . . . . . . . . 12
     3.5.2.  STUN Keep-Alive Technique  . . . . . . . . . . . . . . 12
 4.  User Agent Procedures  . . . . . . . . . . . . . . . . . . . . 13
   4.1.  Instance ID Creation . . . . . . . . . . . . . . . . . . . 13
   4.2.  Registrations  . . . . . . . . . . . . . . . . . . . . . . 14
     4.2.1.  Initial Registrations  . . . . . . . . . . . . . . . . 14
     4.2.2.  Subsequent REGISTER Requests . . . . . . . . . . . . . 16
     4.2.3.  Third-Party Registrations  . . . . . . . . . . . . . . 17
   4.3.  Sending Non-REGISTER Requests  . . . . . . . . . . . . . . 17
   4.4.  Keep-Alives and Detecting Flow Failure . . . . . . . . . . 18
     4.4.1.  Keep-Alive with CRLF . . . . . . . . . . . . . . . . . 19
     4.4.2.  Keep-Alive with STUN . . . . . . . . . . . . . . . . . 21
   4.5.  Flow Recovery  . . . . . . . . . . . . . . . . . . . . . . 21
 5.  Edge Proxy Procedures  . . . . . . . . . . . . . . . . . . . . 22
   5.1.  Processing Register Requests . . . . . . . . . . . . . . . 22
   5.2.  Generating Flow Tokens . . . . . . . . . . . . . . . . . . 23
   5.3.  Forwarding Non-REGISTER Requests . . . . . . . . . . . . . 23
     5.3.1.  Processing Incoming Requests . . . . . . . . . . . . . 24
     5.3.2.  Processing Outgoing Requests . . . . . . . . . . . . . 24
   5.4.  Edge Proxy Keep-Alive Handling . . . . . . . . . . . . . . 25
 6.  Registrar Procedures . . . . . . . . . . . . . . . . . . . . . 25
 7.  Authoritative Proxy Procedures: Forwarding Requests  . . . . . 27

Jennings, et al. Standards Track [Page 2] RFC 5626 Client-Initiated Connections in SIP October 2009

 8.  STUN Keep-Alive Processing . . . . . . . . . . . . . . . . . . 28
   8.1.  Use with SigComp . . . . . . . . . . . . . . . . . . . . . 29
 9.  Example Message Flow . . . . . . . . . . . . . . . . . . . . . 30
   9.1.  Subscription to Configuration Package  . . . . . . . . . . 30
   9.2.  Registration . . . . . . . . . . . . . . . . . . . . . . . 32
   9.3.  Incoming Call and Proxy Crash  . . . . . . . . . . . . . . 34
   9.4.  Re-Registration  . . . . . . . . . . . . . . . . . . . . . 37
   9.5.  Outgoing Call  . . . . . . . . . . . . . . . . . . . . . . 38
 10. Grammar  . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
 11. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 40
   11.1. Flow-Timer Header Field  . . . . . . . . . . . . . . . . . 40
   11.2. "reg-id" Contact Header Field Parameter  . . . . . . . . . 40
   11.3. SIP/SIPS URI Parameters  . . . . . . . . . . . . . . . . . 41
   11.4. SIP Option Tag . . . . . . . . . . . . . . . . . . . . . . 41
   11.5. 430 (Flow Failed) Response Code  . . . . . . . . . . . . . 41
   11.6. 439 (First Hop Lacks Outbound Support) Response Code . . . 42
   11.7. Media Feature Tag  . . . . . . . . . . . . . . . . . . . . 42
 12. Security Considerations  . . . . . . . . . . . . . . . . . . . 43
 13. Operational Notes on Transports  . . . . . . . . . . . . . . . 44
 14. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 44
 15. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 45
 16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 45
   16.1. Normative References . . . . . . . . . . . . . . . . . . . 45
   16.2. Informative References . . . . . . . . . . . . . . . . . . 47
 Appendix A.  Default Flow Registration Backoff Times . . . . . . . 49
 Appendix B.  ABNF  . . . . . . . . . . . . . . . . . . . . . . . . 49

Jennings, et al. Standards Track [Page 3] RFC 5626 Client-Initiated Connections in SIP October 2009

1. Introduction

 There are many environments for SIP [RFC3261] 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, but the most likely
 is a NAT or a firewall in between the SIP UA and the proxy.  Many
 such devices will only allow outgoing connections.  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 the Dynamic Host
 Configuration Protocol (DHCP) [RFC2131].  These systems typically do
 not have a useful name in the Domain Name System (DNS) [RFC1035], 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 [RFC3261].  However, these systems can still act as a
 Transport Layer Security (TLS) [RFC5246] client and form outbound
 connections to a proxy or registrar that 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.  This specification allows a SIP User Agent
 who has to initiate the TLS connection to receive inbound traffic
 associated with registrations or dialogs that it initiates.
 The key idea of this specification is that when a UA sends a REGISTER
 request 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 in 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 two keep-alive schemes.  The keep-alive
 mechanism is used to keep NAT bindings fresh, and to allow the UA to
 detect when a flow has failed.

Jennings, et al. Standards Track [Page 4] RFC 5626 Client-Initiated Connections in SIP October 2009

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 [RFC2119].

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 [RFC3261], and forwards those requests
    to specific Contact URIs.  (In [RFC3261], the role that is
    authoritative for REGISTER requests for a specific AOR is a
    Registration Server.)
 Edge Proxy:  An edge proxy is any proxy that is located topologically
    between the registering User Agent and the Authoritative Proxy.
    The "first" edge proxy refers to the first edge proxy encountered
    when a UA sends a request.
 Flow:  A Flow is a transport-layer association between two hosts that
    is represented by the network address and port number of both ends
    and by the transport 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.
 Flow Token:  An identifier that uniquely identifies a flow which can
    be included in a SIP URI (Uniform Resource Identifier [RFC3986]).
 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 for a different flow, 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 which appears
    as a "+sip.instance" Contact header field parameter.  This is a
    Uniform Resource Name (URN) that uniquely identifies this specific
    UA instance.
 "ob" Parameter:  The "ob" parameter is a SIP URI parameter that has a
    different meaning depending on context.  In a Path header field
    value, it is used by the first edge proxy to indicate that a flow
    token was added to the URI.  In a Contact or Route header field
    value, it indicates that the UA would like other requests in the
    same dialog to be routed over the same flow.

Jennings, et al. Standards Track [Page 5] RFC 5626 Client-Initiated Connections in SIP October 2009

 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.
 This entire section is non-normative.

3.1. Summary of Mechanism

 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 flows 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 can use the instance-id to recognize that two
 different registrations both correspond to the same UA.  The
 registrar can use the reg-id label to recognize whether a UA is
 creating 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.
 When sending a dialog-forming request, a UA can also ask its first
 edge proxy to route subsequent requests in that dialog over the same
 flow.  This is necessary whether the UA has registered or not.
 UAs use a simple periodic message as a keep-alive mechanism to keep
 their flow to the proxy or registrar alive.  For connection-oriented
 transports such as TCP this is based on carriage-return and line-feed

Jennings, et al. Standards Track [Page 6] RFC 5626 Client-Initiated Connections in SIP October 2009

 sequences (CRLF), 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) [RFC5389].

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 that form only a single flow continue to register
 normally but include the instance-id as described in Section 4.1.
 The UA also includes a "reg-id" Contact header field parameter that
 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.
 REGISTER sip:example.com SIP/2.0
 Via: SIP/2.0/TCP 192.0.2.2;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@192.0.2.2;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.

Jennings, et al. Standards Track [Page 7] RFC 5626 Client-Initiated Connections in SIP October 2009

 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 over
 an existing flow, instead of resolving the Contact URI using the
 procedures in [RFC3263] 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.

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.

Jennings, et al. Standards Track [Page 8] RFC 5626 Client-Initiated Connections in SIP October 2009

     +-------------------+
     | 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 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" and "sip:
 secondary.example.com".  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 DNS SRV [RFC2782]
 can be used to do load-balancing across a proxy farm.  The approach
 in this document does not prevent future extensions, such as the SIP
 UA configuration framework [CONFIG-FMWK], from adding other ways for
 a User Agent to discover its outbound-proxy-set.
 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 a recommended amount of time before trying to form a new

Jennings, et al. Standards Track [Page 9] RFC 5626 Client-Initiated Connections in SIP October 2009

 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.
 Scalability is achieved by using DNS SRV [RFC2782] 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 to [RFC3263].
 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.  There could be a NAT or firewall between the UA and the
 edge proxy.
              +---------+
              |Registrar|
              |Proxy    |
              +---------+
               /      \
              /        \
             /          \
          +-----+     +-----+
          |Edge1|     |Edge2|
          +-----+     +-----+
             \           /
              \         /
      ----------------------------NAT/FW
                \     /
                 \   /
                +------+
                |User  |
                |Agent |
                +------+

Jennings, et al. Standards Track [Page 10] RFC 5626 Client-Initiated Connections in SIP October 2009

 The edge proxy includes a Path header [RFC3327] so that when the
 proxy/registrar later forwards a request to this UA, the request is
 routed through the edge proxy.
 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 of whether they are in the same administrative domain,
 this specification requires that registrars and edge proxies support
 the Path header mechanism in [RFC3327].

3.5. Keep-Alive Technique

 This document describes two keep-alive mechanisms: a CRLF keep-alive
 and a STUN keep-alive.  Each of these mechanisms uses a client-to-
 server "ping" keep-alive 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 (allowing for
 retransmission for STUN as described in Section 4.4.2), it declares
 the flow dead and opens a new flow in its place.
 This document also suggests timer values for these client keep-alive
 mechanisms.  These timer values were chosen to keep most NAT and
 firewall bindings open, to detect unresponsive servers within 2
 minutes, and to mitigate against 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,

Jennings, et al. Standards Track [Page 11] RFC 5626 Client-Initiated Connections in SIP October 2009

 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.
 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 4.5 to provide congestion
 relief when a large number of agents simultaneously reboot.
 A keep-alive 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 [RFC0793] and SCTP [RFC4960], this
 specification describes a keep-alive approach based on sending CRLFs.
 For connectionless transport, such as UDP [RFC0768], this
 specification describes using STUN [RFC5389] over the same flow as
 the SIP traffic to perform the keep-alive.
 UAs and Proxies are also free to use native transport keep-alives;
 however, the 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
 keep-alives is outside the scope of this document.

3.5.1. CRLF Keep-Alive Technique

 This approach can only be used with connection-oriented transports
 such as TCP or SCTP.  The client periodically sends a double-CRLF
 (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.
    Note: Sending a CRLF over a connection-oriented transport is
    backwards compatible (because of requirements in Section 7.5 of
    [RFC3261]), but only implementations which support this
    specification will respond to a "ping" with a "pong".

3.5.2. STUN Keep-Alive Technique

 This approach 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

Jennings, et al. Standards Track [Page 12] RFC 5626 Client-Initiated Connections in SIP October 2009

 for the SIP traffic.  The proxy or registrar acts as a limited
 Session Traversal Utilities for NAT (STUN) [RFC5389] server on the
 SIP signaling port.
    Note: The STUN mechanism is very robust and allows the detection
    of a changed IP address and port.  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 Procedures

4.1. Instance ID Creation

 Each UA MUST have an Instance Identifier Uniform Resource Name (URN)
 [RFC2141] 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 Universally Unique Identifier (UUID) URN
 [RFC4122] 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.
    Note: A device like a "soft phone", when first installed, can
    generate a UUID [RFC4122] 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 on this device.
 If a URN scheme other than UUID is used, the UA MUST only use URNs
 for which an RFC (from the IETF stream) defines how the specific URN
 needs to be constructed and used in the "+sip.instance" Contact
 header field parameter for outbound behavior.

Jennings, et al. Standards Track [Page 13] RFC 5626 Client-Initiated Connections in SIP October 2009

 To convey its instance-id in both requests and responses, the UA
 includes a "sip.instance" media feature tag as a UA characteristic
 [RFC3840].  This media feature tag is encoded in the Contact header
 field as the "+sip.instance" Contact header field parameter.  One
 case where a UA could prefer to omit the "sip.instance" media feature
 tag is when it is making an anonymous request or some other privacy
 concern requires that the UA not reveal its identity.
    Note: [RFC3840] 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 the callee capabilities
    [RFC3840] and the caller preferences [RFC3841] specifications.
    When the instance ID is used in this specification, it is
    "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 [RFC2141].  Lexical equality could result in two
    URNs being considered unequal when they are actually equal.  In
    this specific usage of URNs, the only element that provides the
    URN is the SIP UA instance identified by that URN.  As a result,
    the UA instance has to 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 to (yet
    lexicographically different from) previous registrations.

4.2. Registrations

4.2.1. 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.

Jennings, et al. Standards Track [Page 14] RFC 5626 Client-Initiated Connections in SIP October 2009

 For each outbound proxy URI in the set, the User Agent Client (UAC)
 SHOULD send a REGISTER request using this URI as the default outbound
 proxy.  (Alternatively, the UA could limit the number of flows formed
 to conserve battery power, for example).  If the set has more than
 one URI, the UAC MUST send a REGISTER request to at least two of the
 default outbound proxies from the set.  UAs that support this
 specification MUST include the outbound option tag in a Supported
 header field in a REGISTER request.  Each of these REGISTER requests
 will use a unique Call-ID.  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.
 REGISTER requests, other than those described in Section 4.2.3, MUST
 include an instance-id media feature tag as specified in Section 4.1.
 A UAC conforming to this specification MUST include in the Contact
 header field, a "reg-id" parameter that is distinct from other
 "reg-id" parameters used in other registrations that use the same
 "+sip.instance" Contact header field parameter and AOR.  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.
    Note: The UAC can situationally decide whether to request outbound
    behavior by including or omitting the "reg-id" Contact header
    field 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 ensure consistency.
    Similarly, if the registration through EP1 did not support
    outbound, the UA might not register with EP2 at all.
 The UAC MUST support the Path header [RFC3327] mechanism, and
 indicate its support 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 examines 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

Jennings, et al. Standards Track [Page 15] RFC 5626 Client-Initiated Connections in SIP October 2009

 outbound, the UA has potentially registered an un-routable contact.
 It is the responsibility of the UA to remove any inappropriate
 Contacts.
 If outbound registration succeeded, as indicated by the presence of
 the outbound option-tag in the Require header field of a successful
 registration response, the UA begins sending keep-alives as described
 in Section 4.4.
    Note: The UA needs to honor 503 (Service Unavailable) responses to
    registrations as described in [RFC3261] and [RFC3263].  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 request
    immediately.  Implementations need to ensure that when retrying
    the REGISTER request, 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.
 If the registering UA receives a 439 (First Hop Lacks Outbound
 Support) response to a REGISTER request, it MAY re-attempt
 registration without using the outbound mechanism (subject to local
 policy at the client).  If the client has one or more alternate
 outbound proxies available, it MAY re-attempt registration through
 such outbound proxies.  See Section 11.6 for more information on the
 439 response code.

4.2.2. Subsequent REGISTER Requests

 Registrations for refreshing a binding and for removing a binding use
 the same instance-id and reg-id values as the corresponding initial
 registration where the binding was added.  Registrations that merely
 refresh an existing binding are sent over the same flow as the
 original registration where the binding was added.
 If a re-registration is rejected with a recoverable error response,
 for example by a 503 (Service Unavailable) containing a Retry-After
 header, the UAC SHOULD NOT tear down the corresponding flow if the
 flow uses a connection-oriented transport such as TCP.  As long as
 "pongs" are received in response to "pings", the flow SHOULD be kept
 active until a non-recoverable error response is received.  This
 prevents unnecessary closing and opening of connections.

Jennings, et al. Standards Track [Page 16] RFC 5626 Client-Initiated Connections in SIP October 2009

4.2.3. Third-Party Registrations

 In an initial registration or re-registration, a UA MUST NOT include
 a "reg-id" header field parameter in the Contact header field 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 feature tag or "reg-id"
 Contact header field parameter in a request to un-register all
 Contacts (a single Contact header field value with the value of "*").

4.3. Sending Non-REGISTER Requests

 When a UAC 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.
 UAs that support this specification SHOULD include the outbound
 option tag in a Supported header field in a request that is not a
 REGISTER request.
 The UAC performs normal DNS resolution on the next hop URI (as
 described in [RFC3263]) to find a protocol, IP address, and port.
 For protocols that don't use TLS, if the UAC has an existing flow to
 this IP address, and port with the correct protocol, then the UAC
 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
 UAC 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.
 Typically, a UAC using the procedures of this document and sending a
 dialog-forming request will want all subsequent requests in the
 dialog to arrive over the same flow.  If the UAC is using a Globally
 Routable UA URI (GRUU) [RFC5627] that was instantiated using a
 Contact header field value that included an "ob" parameter, the UAC
 sends the request over the flow used for registration, and subsequent
 requests will arrive over that same flow.  If the UAC is not using
 such a GRUU, then the UAC adds an "ob" parameter to its Contact
 header field value.  This will cause all subsequent requests in the
 dialog to arrive over the flow instantiated by the dialog-forming
 request.  This case is typical when the request is sent prior to
 registration, such as in the initial subscription dialog for the
 configuration framework [CONFIG-FMWK].

Jennings, et al. Standards Track [Page 17] RFC 5626 Client-Initiated Connections in SIP October 2009

    Note: If the UAC wants a UDP flow to work through NATs or
    firewalls, it still needs to put the 'rport' parameter [RFC3581]
    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 [NAT-SCEN].

4.4. Keep-Alives and Detecting Flow Failure

 Keep-alives are used for refreshing NAT/firewall bindings and
 detecting flow failure.  Flows can fail for many reasons including
 the rebooting of NATs and the crashing of edge proxies.
 As described in Section 4.2, a UA that registers will begin sending
 keep-alives after an appropriate registration response.  A UA that
 does not register (for example, a PSTN gateway behind a firewall) can
 also send keep-alives under certain circumstances.
 Under specific circumstances, a UAC might be allowed to send STUN
 keep-alives even if the procedures in Section 4.2 were not completed,
 provided that there is an explicit indication that the target first-
 hop SIP node supports STUN keep-alives.  For example, this applies to
 a non-registering UA or to a case where the UA registration
 succeeded, but the response did not include the outbound option-tag
 in the Require header field.
    Note: A UA can "always" send a double CRLF (a "ping") over
    connection-oriented transports as this is already allowed by
    Section 7.5 of [RFC3261].  However a UA that did not register
    using outbound registration cannot expect a CRLF in response (a
    "pong") unless the UA has an explicit indication that CRLF keep-
    alives are supported as described in this section.  Likewise, a UA
    that did not successfully register with outbound procedures needs
    explicit indication that the target first-hop SIP node supports
    STUN keep-alives before it can send any STUN messages.
 A configuration option indicating keep-alive support for a specific
 target is considered an explicit indication.  If these conditions are
 satisfied, the UA sends its keep-alives according to the same
 guidelines as those used when UAs register; these guidelines are
 described below.
 The UA needs to detect when a specific flow fails.  The UA actively
 tries to detect failure by periodically sending keep-alive messages
 using one of the techniques described in Sections 4.4.1 or 4.4.2.  If
 a flow with a registration has failed, the UA follows the procedures
 in Section 4.2 to form a new flow to replace the failed one.

Jennings, et al. Standards Track [Page 18] RFC 5626 Client-Initiated Connections in SIP October 2009

 When a successful registration response contains the Flow-Timer
 header field, the value of this header field is the number of seconds
 the server is prepared to wait without seeing keep-alives before it
 could consider the corresponding flow dead.  Note that the server
 would wait for an amount of time larger than the Flow-Timer in order
 to have a grace period to account for transport delay.  The UA MUST
 send keep-alives at least as often as this number of seconds.  If the
 UA uses the server-recommended keep-alive frequency it SHOULD send
 its keep-alives so that the interval between each keep-alive is
 randomly distributed between 80% and 100% of the server-provided
 time.  For example, if the server suggests 120 seconds, the UA would
 send each keep-alive with a different frequency between 95 and 120
 seconds.
 If no Flow-Timer header field was present in a register response for
 this flow, the UA can send keep-alives at its discretion.  The
 sections below provide RECOMMENDED default values for these keep-
 alives.
 The client needs to perform normal [RFC3263] SIP DNS resolution on
 the URI from the outbound-proxy-set to pick a transport.  Once a
 transport is selected, the UA selects the keep-alive approach that is
 recommended for that transport.
 Section 4.4.1 describes a keep-alive mechanism for connection-
 oriented transports such as TCP or SCTP.  Section 4.4.2 describes a
 keep-alive mechanism for connection-less transports such as UDP.
 Support for other transports such as DCCP [RFC4340] is for further
 study.

4.4.1. Keep-Alive with CRLF

 This approach MUST only be used with connection oriented transports
 such as TCP or SCTP; it MUST NOT 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 resolves to a connection-oriented transport
 (e.g., TCP and TLS over TCP).
 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 = %x0D
 LF = %x0A

Jennings, et al. Standards Track [Page 19] RFC 5626 Client-Initiated Connections in SIP October 2009

 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 [RFC3320] compressed data stream, the CRLF keep-alives are
 sent inside the compressed stream.  The double CRLF is considered a
 single SigComp 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 after sending a ping (or
 immediately after processing any incoming message being received when
 that 10 seconds expires), then the client MUST treat the flow as
 failed.  Clients MUST support this CRLF keep-alive.
    Note: This value of 10-second timeout was selected to be long
    enough that it allows plenty of time for a server to send a
    response even if the server is temporarily busy with an
    administrative activity.  At the same time, it was selected to be
    small enough that a UA registered to two redundant servers with
    unremarkable hardware uptime could still easily provide very high
    levels of overall reliability.  Although some Internet protocols
    are designed for round-trip times over 10 seconds, SIP for real-
    time communications is not really usable in these type of
    environments as users often abandon calls before waiting much more
    than a few seconds.
 When a Flow-Timer header field is not provided in the most recent
 success registration response, the proper selection of keep-alive
 frequency is primarily a trade-off between battery usage and
 availability.  The UA MUST select a random number between a fixed or
 configurable upper bound and a lower bound, where the lower bound is
 20% less then the upper bound.  The fixed upper bound or the default
 configurable upper bound SHOULD be 120 seconds (95 seconds for the
 lower bound) where battery power is not a concern and 840 seconds
 (672 seconds for the lower bound) where battery power is a concern.
 The random number will be different for each keep-alive "ping".
    Note on selection of time values: the 120-second 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 will be set up.  The 14-minute upper bound for battery-
    powered devices was selected based on NATs with TCP timeouts as
    low as 15 minutes.  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-second lower bound was chosen so that the jitter introduced
    will result in a relatively even load on the servers after 30
    minutes.

Jennings, et al. Standards Track [Page 20] RFC 5626 Client-Initiated Connections in SIP October 2009

4.4.2. Keep-Alive with STUN

 This approach MUST only be used with connection-less transports, such
 as UDP; it MUST NOT be used for connection-oriented transports such
 as TCP and SCTP.
 A User Agent that forms flows checks if the configured URI to which
 the UA is connecting resolves to use the UDP transport.  The UA can
 periodically perform keep-alive checks by sending STUN [RFC5389]
 Binding Requests over the flow as described in Section 8.  Clients
 MUST support STUN-based keep-alives.
 When a Flow-Timer header field is not included in a successful
 registration response, the time between each keep-alive request
 SHOULD be a random number between 24 and 29 seconds.
    Note on selection of time values: the upper bound of 29 seconds
    was selected, 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 keep-alive
    requests unsynchronized to evenly spread the load on the servers.
    Note that the short NAT timeouts with UDP have a negative impact
    on battery life.
 If a STUN Binding Error Response is received, or if no Binding
 Response is received after 7 retransmissions (16 times the STUN "RTO"
 timer -- where 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 used for registration (through 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
 value 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 keep-alives are in use on this flow, at least one
 subsequent keep-alive response was received.

Jennings, et al. Standards Track [Page 21] RFC 5626 Client-Initiated Connections in SIP October 2009

 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 a lower value (with a default of 30 seconds);
 otherwise, in the case where at least one of the flows has not
 failed, the base-time is set to a higher value (with a default of 90
 seconds).  The upper-bound wait time (W) 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
 configurable maximum time (with a default of 1800 seconds).
 W = 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 upper-bound wait time is min(1800, 30*(2^3)) or
 240 seconds.  The actual amount of time the UA waits before retrying
 registration (the retry delay time) is computed by selecting a
 uniform random time between 50 and 100% of the upper-bound wait time.
 The UA MUST wait for at least the value of the retry delay time
 before trying another registration to form a new flow for that URI (a
 503 response to an earlier failed registration attempt with a Retry-
 After header field value may cause the UA to wait longer).
 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 Procedures

5.1. Processing Register Requests

 When an edge proxy receives a registration request with a "reg-id"
 header field 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 is the first hop, as

Jennings, et al. Standards Track [Page 22] RFC 5626 Client-Initiated Connections in SIP October 2009

 indicated by the Via header field, it MUST insert its URI in a Path
 header field value as described in [RFC3327].  If it is not the first
 hop, it might still decide to add itself to the Path header based on
 local policy.  In addition, if the edge proxy is the first SIP node
 after the UAC, the edge proxy 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 infeasible 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 [RFC2104].  The
    concatenation of the HMAC and S are base64 encoded, as defined in
    [RFC4648], 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 additions.  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 follows the procedures of this section.  Otherwise
 the edge proxy skips the procedures in this section, removes itself
 from the Route header field, and continues processing the request.

Jennings, et al. Standards Track [Page 23] RFC 5626 Client-Initiated Connections in SIP October 2009

 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 identified by the topmost Route header
 field value matches the source IP address and port of the request,
 the request is an "outgoing" request; otherwise, it is an "incoming"
 request.

5.3.1. Processing Incoming Requests

 If the Route header value contains an "ob" URI parameter, the Route
 header was probably copied from the Path header in a registration.
 If the Route header value contains an "ob" URI parameter, and the
 request is a new dialog-forming request, the proxy needs to adjust
 the route set to ensure that subsequent requests in the dialog can be
 delivered over a valid flow to the UA instance identified by the flow
 token.
    Note: A simple approach to satisfy this requirement is for the
    proxy to add a Record-Route header field value that contains the
    flow-token, by copying the URI in the Route header minus the "ob"
    parameter.
 Next, whether the Route header field contained an "ob" URI parameter
 or not, 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.  If
 the flow token has been tampered with, the proxy SHOULD send a 403
 (Forbidden) response.  If the flow no longer exists, the proxy SHOULD
 send a 430 (Flow Failed) response to the request.
 Proxies that used the example algorithm described in Section 5.2 to
 form a flow token follow the procedures below to determine the
 correct flow.  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 request has been tampered with.

5.3.2. Processing Outgoing Requests

 For mid-dialog requests to work with outbound UAs, the requests need
 to be forwarded over some valid flow to the appropriate UA instance.
 If the edge proxy receives an outgoing dialog-forming request, the
 edge proxy can use the presence of the "ob" URI parameter in the
 UAC's Contact URI (or topmost Route header field) to determine if the
 edge proxy needs to assist in mid-dialog request routing.

Jennings, et al. Standards Track [Page 24] RFC 5626 Client-Initiated Connections in SIP October 2009

    Implementation note: 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.

5.4. Edge Proxy Keep-Alive Handling

 All edge proxies compliant with this specification MUST implement
 support for STUN NAT keep-alives on their SIP UDP ports as described
 in Section 8.
 When a server receives a double CRLF sequence between SIP messages on
 a connection-oriented transport such as TCP or SCTP, it MUST
 immediately respond with a single CRLF over the same connection.
 The last proxy to forward a successful registration response to a UA
 MAY include a Flow-Timer header field if the response contains the
 outbound option-tag in a Require header field value in the response.
 The reason a proxy would send a Flow-Timer is if it wishes to detect
 flow failures proactively and take appropriate action (e.g., log
 alarms, provide alternative treatment if incoming requests for the UA
 are received, etc.).  The server MUST wait for an amount of time
 larger than the Flow-Timer in order to have a grace period to account
 for transport delay.

6. Registrar Procedures

 This specification updates the definition of a binding in [RFC3261],
 Section 10 and [RFC3327], Section 5.3.
 Registrars that implement this specification MUST support the Path
 header mechanism [RFC3327].
 When receiving a REGISTER request, the registrar MUST check from its
 Via header field if the registrar is the first hop or not.  If the
 registrar is not the first hop, it MUST examine 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, then outbound
 processing MUST NOT be applied to the registration.  In this case,
 the following processing applies: if the REGISTER request contains
 the reg-id and the outbound option tag in a Supported header field,
 then the registrar MUST respond to the REGISTER request with a 439
 (First Hop Lacks Outbound Support) response; otherwise, the registrar
 MUST ignore the "reg-id" parameter of the Contact header.  See
 Section 11.6 for more information on the 439 response code.

Jennings, et al. Standards Track [Page 25] RFC 5626 Client-Initiated Connections in SIP October 2009

 A Contact header field value with an instance-id media feature tag
 but no "reg-id" header field parameter is valid (this combination
 will result in the creation of a GRUU, as described in the GRUU
 specification [RFC5627]), but one with a reg-id but no instance-id is
 not valid.  If the registrar processes a Contact header field value
 with a reg-id but no instance-id, it simply ignores the reg-id
 parameter.
 A registration containing a "reg-id" header field parameter and a
 non-zero expiration is used to register a single UA instance over a
 single flow, and can also de-register any Contact header fields with
 zero expiration.  Therefore, if the Contact header field contains
 more than one header field value with a non-zero expiration and any
 of these header field values contain a "reg-id" Contact header field
 parameter, the entire registration SHOULD be rejected with a 400 (Bad
 Request) response.  The justification for recommending rejection
 versus making it mandatory is that the receiver is allowed by
 [RFC3261] to squelch (not respond to) excessively malformed or
 malicious messages.
 If the Contact header did not contain a "reg-id" Contact header field
 parameter or if that parameter was 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" URI parameter.  If the Contact header field does not
 contain a "+sip.instance" Contact header field parameter, the
 registrar processes the request using the Contact binding rules in
 [RFC3261].
 When a "+sip.instance" Contact header field parameter and a "reg-id"
 Contact header field parameter are 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" Contact
 header parameter) and the value of "reg-id" Contact header field
 parameter parameter.  The registrar MUST store in the binding the
 Contact URI, all the Contact header field parameters, and any Path
 header field values.  (Even though the Contact URI is not used for
 binding comparisons, it is still needed by the authoritative proxy to
 form the target set.)  Provided that the UAC had included an outbound
 option-tag (defined in Section 11.4) in a Supported header field

Jennings, et al. Standards Track [Page 26] RFC 5626 Client-Initiated Connections in SIP October 2009

 value in the REGISTER request, the registrar MUST include the
 outbound option-tag in a Require header field value in its response
 to that 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 be just 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, and instance-id and reg-id values, the registrar
 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.
 To be compliant with this specification, registrars that can receive
 SIP requests directly from a UAC without intervening edge proxies
 MUST implement the same keep-alive mechanisms as edge proxies
 (Section 5.4).  Registrars with a direct flow with a UA MAY include a
 Flow-Timer header in a 2xx class registration response that includes
 the outbound option-tag in the Require header.

7. Authoritative Proxy Procedures: 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.

Jennings, et al. Standards Track [Page 27] RFC 5626 Client-Initiated Connections in SIP October 2009

 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 is co-located with the registrar and stored information
 about the flow to the UA 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.
    Implementation note: 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.

8. STUN Keep-Alive Processing

 This section describes changes to the SIP transport layer that allow
 SIP and STUN [RFC5389] Binding Requests to be mixed over the same
 flow.  This constitutes a new STUN usage.  The STUN messages are used
 to verify that connectivity is still available over a UDP flow, and
 to provide periodic keep-alives.  These STUN keep-alives 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.
 The corresponding Binding Responses do not require any STUN
 attributes except the XOR-MAPPED-ADDRESS.  The UAS, proxy, or
 registrar responds to a valid Binding Request with a Binding Response
 that 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.

Jennings, et al. Standards Track [Page 28] RFC 5626 Client-Initiated Connections in SIP October 2009

    Note: 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.
 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
 explicit indication that the target next-hop SIP server claims to
 support this specification.  UACs MUST NOT use an ambiguous
 configuration option such as "Work through NATs?" or "Do keep-
 alives?" to imply next-hop STUN support.  A UAC MAY use the presence
 of an "ob" URI parameter in the Path header in a registration
 response as an indication that its first edge proxy supports the
 keep-alives defined in this document.
    Note: Typically, a SIP node first sends a SIP request and waits to
    receive a 2xx 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 keep-alive STUN usage requires no backwards compatibility
 with [RFC3489].

8.1. Use with SigComp

 When STUN is used together with SigComp [RFC3320] compressed SIP
 messages over the same flow, 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.
    Note: 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 [RFC3629] encoded text messages, so for

Jennings, et al. Standards Track [Page 29] RFC 5626 Client-Initiated Connections in SIP October 2009

    applications that 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.

9. Example Message Flow

 Below is an example message flow illustrating most of the concepts
 discussed in this specification.  In many cases, Via, Content-Length,
 and Max-Forwards headers are omitted for brevity and readability.
 In these examples, "EP1" and "EP2" are outbound proxies, and "Proxy"
 is the authoritativeProxy.
 The section is subdivided into independent calls flows; however, they
 are structured in sequential order of a hypothetical sequence of call
 flows.

9.1. Subscription to Configuration Package

 If the outbound proxy set is already configured on Bob's UA, then
 this subsection can be skipped.  Otherwise, if the outbound proxy set
 is learned through the configuration package, Bob's UA sends a
 SUBSCRIBE request for the UA profile configuration package
 [CONFIG-FMWK].  This request is a poll (Expires is zero).  After
 receiving the NOTIFY request, Bob's UA fetches the external
 configuration using HTTPS (not shown) and obtains a configuration
 file that contains the outbound-proxy-set "sip:ep1.example.com;lr"
 and "sip:ep2.example.com;lr".
   [----example.com domain-------------------------]
   Bob         EP1   EP2     Proxy             Config
    |           |     |        |                  |
  1)|SUBSCRIBE->|     |        |                  |
  2)|           |---SUBSCRIBE Event: ua-profile ->|
  3)|           |<--200 OK -----------------------|
  4)|<--200 OK--|     |        |                  |
  5)|           |<--NOTIFY------------------------|
  6)|<--NOTIFY--|     |        |                  |
  7)|---200 OK->|     |        |                  |
  8)|           |---200 OK ---------------------->|
    |           |     |        |                  |
 In this example, the DNS server happens to be configured so that sip:
 example.com resolves to EP1 and EP2.

Jennings, et al. Standards Track [Page 30] RFC 5626 Client-Initiated Connections in SIP October 2009

 Example Message #1:
 SUBSCRIBE sip:00000000-0000-1000-8000-AABBCCDDEEFF@example.com
   SIP/2.0
 Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnlsdkdj2
 Max-Forwards: 70
 From: <anonymous@example.com>;tag=23324
 To: <sip:00000000-0000-1000-8000-AABBCCDDEEFF@example.com>
 Call-ID: nSz1TWN54x7My0GvpEBj
 CSeq: 1 SUBSCRIBE
 Event: ua-profile ;profile-type=device
  ;vendor="example.com";model="uPhone";version="1.1"
 Expires: 0
 Supported: path, outbound
 Accept: message/external-body, application/x-uPhone-config
 Contact: <sip:192.0.2.2;transport=tcp;ob>
  ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
 Content-Length: 0
 In message #2, EP1 adds the following Record-Route header:
 Record-Route:
  <sip:GopIKSsn0oGLPXRdV9BAXpT3coNuiGKV@ep1.example.com;lr>
 In message #5, the configuration server sends a NOTIFY with an
 external URL for Bob to fetch his configuration.  The NOTIFY has a
 Subscription-State header that ends the subscription.
 Message #5
 NOTIFY sip:192.0.2.2;transport=tcp;ob SIP/2.0
 Via: SIP/2.0/TCP 192.0.2.5;branch=z9hG4bKn81dd2
 Max-Forwards: 70
 To: <anonymous@example.com>;tag=23324
 From: <sip:00000000-0000-1000-8000-AABBCCDDEEFF@example.com>;tag=0983
 Call-ID: nSz1TWN54x7My0GvpEBj
 CSeq: 1 NOTIFY
 Route: <sip:GopIKSsn0oGLPXRdV9BAXpT3coNuiGKV@ep1.example.com;lr>
 Subscription-State: terminated;reason=timeout
 Event: ua-profile
 Content-Type: message/external-body; access-type="URL"
  ;expiration="Thu, 01 Jan 2009 09:00:00 UTC"
  ;URL="http://example.com/uPhone.cfg"
  ;size=9999;hash=10AB568E91245681AC1B
 Content-Length: 0

Jennings, et al. Standards Track [Page 31] RFC 5626 Client-Initiated Connections in SIP October 2009

 EP1 receives this NOTIFY request, strips off the Route header,
 extracts the flow-token, calculates the correct flow, and forwards
 the request (message #6) over that flow to Bob.
 Bob's UA fetches the configuration file and learns the outbound proxy
 set.

9.2. Registration

 Now that Bob's UA is configured with the outbound-proxy-set whether
 through configuration or using the configuration framework procedures
 of the previous section, Bob's UA sends REGISTER requests through
 each edge proxy in the set.  Once the registrations succeed, Bob's UA
 begins sending CRLF keep-alives about every 2 minutes.
   Bob         EP1   EP2     Proxy     Alice
    |           |     |        |         |
  9)|-REGISTER->|     |        |         |
 10)|           |---REGISTER-->|         |
 11)|           |<----200 OK---|         |
 12)|<-200 OK---|     |        |         |
 13)|----REGISTER---->|        |         |
 14)|           |     |--REG-->|         |
 15)|           |     |<-200---|         |
 16)|<----200 OK------|        |         |
    |           |     |        |         |
    |  about 120 seconds later...        |
    |           |     |        |         |
 17)|--2CRLF--->|     |        |         |
 18)|<--CRLF----|     |        |         |
 19)|------2CRLF----->|        |         |
 20)|<------CRLF------|        |         |
    |           |     |        |         |
 In message #9, Bob's UA sends its first registration through the
 first edge proxy in the outbound-proxy-set by including a loose
 route.  The UA includes an instance-id and reg-id in its Contact
 header field value.  Note the option-tags in the Supported header.

Jennings, et al. Standards Track [Page 32] RFC 5626 Client-Initiated Connections in SIP October 2009

 Message #9
 REGISTER sip:example.com SIP/2.0
 Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7
 Max-Forwards: 70
 From: Bob <sip:bob@example.com>;tag=7F94778B653B
 To: Bob <sip:bob@example.com>
 Call-ID: 16CB75F21C70
 CSeq: 1 REGISTER
 Supported: path, outbound
 Route: <sip:ep1.example.com;lr>
 Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1
  ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
 Content-Length: 0
 Message #10 is similar.  EP1 removes the Route header field value,
 decrements Max-Forwards, and adds its Via header field value.  Since
 EP1 is the first edge proxy, it adds a Path header with a flow token
 and includes the "ob" parameter.
 Path: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob>
 Since the response to the REGISTER (message #11) contains the
 outbound option-tag in the Require header field, Bob's UA will know
 that the registrar used outbound binding rules.  The response also
 contains the currently active Contacts, and the Path for the current
 registration.
 Message #11
 SIP/2.0 200 OK
 Via: SIP/2.0/TCP 192.0.2.15;branch=z9hG4bKnuiqisi
 Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7
 From: Bob <sip:bob@example.com>;tag=7F94778B653B
 To: Bob <sip:bob@example.com>;tag=6AF99445E44A
 Call-ID: 16CB75F21C70
 CSeq: 1 REGISTER
 Supported: path, outbound
 Require: outbound
 Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1;expires=3600
  ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
 Path: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob>
 Content-Length: 0
 The second registration through EP2 (message #13) is similar except
 that the Call-ID has changed, the reg-id is 2, and the Route header
 goes through EP2.

Jennings, et al. Standards Track [Page 33] RFC 5626 Client-Initiated Connections in SIP October 2009

 Message #13
 REGISTER sip:example.com SIP/2.0
 Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnqr9bym
 Max-Forwards: 70
 From: Bob <sip:bob@example.com>;tag=755285EABDE2
 To: Bob <sip:bob@example.com>
 Call-ID: E05133BD26DD
 CSeq: 1 REGISTER
 Supported: path, outbound
 Route: <sip:ep2.example.com;lr>
 Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=2
  ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
 Content-Length: 0
 Likewise in message #14, EP2 adds a Path header with flow token and
 "ob" parameter.
 Path: <sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr;ob>
 Message #16 tells Bob's UA that outbound registration was successful,
 and shows both Contacts.  Note that only the Path corresponding to
 the current registration is returned.
 Message #16
 SIP/2.0 200 OK
 Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnqr9bym
 From: Bob <sip:bob@example.com>;tag=755285EABDE2
 To: Bob <sip:bob@example.com>;tag=49A9AD0B3F6A
 Call-ID: E05133BD26DD
 Supported: path, outbound
 Require: outbound
 CSeq: 1 REGISTER
 Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1;expires=3600
  ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
 Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=2;expires=3600
  ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
 Path: <sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr;ob>
 Content-Length: 0

9.3. Incoming Call and Proxy Crash

 In this example, after registration, EP1 crashes and reboots.  Before
 Bob's UA notices that its flow to EP1 is no longer responding, Alice
 calls Bob.  Bob's authoritative proxy first tries the flow to EP1,

Jennings, et al. Standards Track [Page 34] RFC 5626 Client-Initiated Connections in SIP October 2009

 but EP1 no longer has a flow to Bob, so it responds with a 430 (Flow
 Failed) response.  The proxy removes the stale registration and tries
 the next binding for the same instance.
   Bob         EP1   EP2     Proxy     Alice
    |           |     |        |         |
    |    CRASH  X     |        |         |
    |        Reboot   |        |         |
    |           |     |        |         |
 21)|           |     |        |<-INVITE-|
 22)|           |<---INVITE----|         |
 23)|           |----430------>|         |
 24)|           |     |<-INVITE|         |
 25)|<---INVITE-------|        |         |
 26)|----200 OK------>|        |         |
 27)|           |     |200 OK->|         |
 28)|           |     |        |-200 OK->|
 29)|           |     |<----------ACK----|
 30)|<---ACK----------|        |         |
    |           |     |        |         |
 31)|           |     |<----------BYE----|
 32)|<---BYE----------|        |         |
 33)|----200 OK------>|        |         |
 34)|           |     |--------200 OK--->|
    |           |     |        |         |
 Message #21
 INVITE sip:bob@example.com SIP/2.0
 To: Bob <sip:bob@example.com>
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 1 INVITE
 Bob's proxy rewrites the Request-URI to the Contact URI used in Bob's
 registration, and places the path for one of the registrations
 towards Bob's UA instance into a Route header field.  This Route goes
 through EP1.
 Message #22
 INVITE sip:bob@192.0.2.2;transport=tcp SIP/2.0
 To: Bob <sip:bob@example.com>
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 1 INVITE
 Route: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob>

Jennings, et al. Standards Track [Page 35] RFC 5626 Client-Initiated Connections in SIP October 2009

 Since EP1 just rebooted, it does not have the flow described in the
 flow token.  It returns a 430 (Flow Failed) response.
 Message #23
 SIP/2.0 430 Flow Failed
 To: Bob <sip:bob@example.com>
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 1 INVITE
 The proxy deletes the binding for this path and tries to forward the
 INVITE again, this time with the path through EP2.
 Message #24
 INVITE sip:bob@192.0.2.2;transport=tcp SIP/2.0
 To: Bob <sip:bob@example.com>
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 1 INVITE
 Route: <sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr;ob>
 In message #25, EP2 needs to add a Record-Route header field value,
 so that any subsequent in-dialog messages from Alice's UA arrive at
 Bob's UA.  EP2 can determine it needs to Record-Route since the
 request is a dialog-forming request and the Route header contained a
 flow token and an "ob" parameter.  This Record-Route information is
 passed back to Alice's UA in the responses (messages #26, 27, and
 28).
 Message #25
 INVITE sip:bob@192.0.2.2;transport=tcp SIP/2.0
 To: Bob <sip:bob@example.com>
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 1 INVITE
 Record-Route:
   <sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr>

Jennings, et al. Standards Track [Page 36] RFC 5626 Client-Initiated Connections in SIP October 2009

 Message #26
 SIP/2.0 200 OK
 To: Bob <sip:bob@example.com>;tag=skduk2
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 1 INVITE
 Record-Route:
   <sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr>
 At this point, both UAs have the correct route-set for the dialog.
 Any subsequent requests in this dialog will route correctly.  For
 example, the ACK request in message #29 is sent from Alice's UA
 directly to EP2.  The BYE request in message #31 uses the same route-
 set.
 Message #29
 ACK sip:bob@192.0.2.2;transport=tcp SIP/2.0
 To: Bob <sip:bob@example.com>;tag=skduk2
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 1 ACK
 Route: <sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr>
 Message #31
 BYE sip:bob@192.0.2.2;transport=tcp SIP/2.0
 To: Bob <sip:bob@example.com>;tag=skduk2
 From: Alice <sip:alice@a.example>;tag=02935
 Call-ID: klmvCxVWGp6MxJp2T2mb
 CSeq: 2 BYE
 Route: <sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr>

9.4. Re-Registration

 Somewhat later, Bob's UA sends keep-alives to both its edge proxies,
 but it discovers that the flow with EP1 failed.  Bob's UA re-
 registers through EP1 using the same reg-id and Call-ID it previously
 used.

Jennings, et al. Standards Track [Page 37] RFC 5626 Client-Initiated Connections in SIP October 2009

   Bob         EP1   EP2     Proxy     Alice
    |           |     |        |         |
 35)|------2CRLF----->|        |         |
 36)|<------CRLF------|        |         |
 37)|--2CRLF->X |     |        |         |
    |           |     |        |         |
 38)|-REGISTER->|     |        |         |
 39)|           |---REGISTER-->|         |
 40)|           |<----200 OK---|         |
 41)|<-200 OK---|     |        |         |
    |           |     |        |         |
 Message #38
 REGISTER sip:example.com SIP/2.0
 From: Bob <sip:bob@example.com>;tag=7F94778B653B
 To: Bob <sip:bob@example.com>
 Call-ID: 16CB75F21C70
 CSeq: 2 REGISTER
 Supported: path, outbound
 Route: <sip:ep1.example.com;lr>
 Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1
  ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
 In message #39, EP1 inserts a Path header with a new flow token:
 Path: <sip:3yJEbr1GYZK9cPYk5Snocez6DzO7w+AX@ep1.example.com;lr;ob>

9.5. Outgoing Call

 Finally, Bob makes an outgoing call to Alice.  Bob's UA includes an
 "ob" parameter in its Contact URI in message #42.  EP1 adds a Record-
 Route with a flow-token in message #43.  The route-set is returned to
 Bob in the response (messages #45, 46, and 47), and either Bob or
 Alice can send in-dialog requests.

Jennings, et al. Standards Track [Page 38] RFC 5626 Client-Initiated Connections in SIP October 2009

   Bob         EP1   EP2     Proxy     Alice
    |           |     |        |         |
 42)|--INVITE-->|     |        |         |
 43)|           |---INVITE---->|         |
 44)|           |     |        |-INVITE->|
 45)|           |     |        |<--200---|
 46)|           |<----200 OK---|         |
 47)|<-200 OK---|     |        |         |
 48)|--ACK----->|     |        |         |
 49)|           |-----ACK--------------->|
    |           |     |        |         |
 50)|-- BYE---->|     |        |         |
 51)|           |-----------BYE--------->|
 52)|           |<----------200 OK-------|
 53)|<--200 OK--|     |        |         |
    |           |     |        |         |
 Message #42
 INVITE sip:alice@a.example SIP/2.0
 From: Bob <sip:bob@example.com>;tag=ldw22z
 To: Alice <sip:alice@a.example>
 Call-ID: 95KGsk2V/Eis9LcpBYy3
 CSeq: 1 INVITE
 Route: <sip:ep1.example.com;lr>
 Contact: <sip:bob@192.0.2.2;transport=tcp;ob>
 In message #43, EP1 adds the following Record-Route header.
 Record-Route:
   <sip:3yJEbr1GYZK9cPYk5Snocez6DzO7w+AX@ep1.example.com;lr>
 When EP1 receives the BYE (message #50) from Bob's UA, it can tell
 that the request is an "outgoing" request (since the source of the
 request matches the flow in the flow token) and simply deletes its
 Route header field value and forwards the request on to Alice's UA.
 Message #50
 BYE sip:alice@a.example SIP/2.0
 From: Bob <sip:bob@example.com>;tag=ldw22z
 To: Alice <sip:alice@a.example>;tag=plqus8
 Call-ID: 95KGsk2V/Eis9LcpBYy3
 CSeq: 2 BYE
 Route: <sip:3yJEbr1GYZK9cPYk5Snocez6DzO7w+AX@ep1.example.com;lr>
 Contact: <sip:bob@192.0.2.2;transport=tcp;ob>

Jennings, et al. Standards Track [Page 39] RFC 5626 Client-Initiated Connections in SIP October 2009

10. Grammar

 This specification defines a new header field "Flow-Timer", and new
 Contact header field parameters, "reg-id" and "+sip.instance".  The
 grammar includes the definitions from [RFC3261].  Flow-Timer is an
 extension-header from the message-header in the [RFC3261] ABNF.
 The ABNF [RFC5234] is:
  Flow-Timer     = "Flow-Timer" HCOLON 1*DIGIT
  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
                    DQUOTE "<" instance-val ">" DQUOTE
  instance-val   = 1*uric ; defined in RFC 3261
 The value of the reg-id MUST NOT be 0 and MUST be less than 2^31.

11. IANA Considerations

11.1. Flow-Timer Header Field

 This specification defines a new SIP header field "Flow-Timer" whose
 syntax is defined in Section 10.
   Header Name        compact    Reference
   -----------------  -------    ---------
   Flow-Timer                    [RFC5626]

11.2. "reg-id" Contact Header Field Parameter

 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 [RFC3968].  The syntax is
 defined in Section 10.  The required information is:
                                                Predefined
 Header Field            Parameter Name         Values      Reference
 ----------------------  ---------------------  ----------  ---------
 Contact                 reg-id                 No          [RFC5626]

Jennings, et al. Standards Track [Page 40] RFC 5626 Client-Initiated Connections in SIP October 2009

11.3. SIP/SIPS URI Parameters

 This specification augments the "SIP/SIPS URI Parameters" sub-
 registry as per the registry created by [RFC3969].  The required
 information is:
 Parameter Name     Predefined Values     Reference
 --------------     -----------------     ---------
 ob                 No                    [RFC5626]

11.4. SIP Option Tag

 This specification registers a new SIP option tag, as per the
 guidelines in Section 27.1 of [RFC3261].
 Name:  outbound
 Description:  This option-tag is used to identify UAs and registrars
    that support extensions for Client-Initiated Connections.  A UA
    places this option in a Supported header to communicate its
    support for this extension.  A registrar places this option-tag in
    a Require header to indicate to the registering User Agent that
    the registrar used registrations using the binding rules defined
    in this extension.

11.5. 430 (Flow Failed) Response Code

 This document registers a new SIP response code (430 Flow Failed), as
 per the guidelines in Section 27.4 of [RFC3261].  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.  Endpoints should never receive a 430 response.  If an endpoint
 receives a 430 response, it should treat it as a 400 (Bad Request)
 per normal procedures, as in Section 8.1.3.2 of [RFC3261].  This
 response code is defined by the following information, which has been
 added to the method and response-code sub-registry under the SIP
 Parameters registry.
   Response Code                               Reference
   ------------------------------------------  ---------
   Request Failure 4xx
     430 Flow Failed                           [RFC5626]

Jennings, et al. Standards Track [Page 41] RFC 5626 Client-Initiated Connections in SIP October 2009

11.6. 439 (First Hop Lacks Outbound Support) Response Code

 This document registers a new SIP response code (439 First Hop Lacks
 Outbound Support), as per the guidelines in Section 27.4 of
 [RFC3261].  This response code is used by a registrar to indicate
 that it supports the 'outbound' feature described in this
 specification, but that the first outbound proxy that the user is
 attempting to register through does not.  Note that this response
 code is only appropriate in the case that the registering User Agent
 advertises support for outbound processing by including the outbound
 option tag in a Supported header field.  Proxies MUST NOT send a 439
 response to any requests that do not contain a "reg-id" parameter and
 an outbound option tag in a Supported header field.  This response
 code is defined by the following information, which has been added to
 the method and response-code sub-registry under the SIP Parameters
 registry.
   Response Code                               Reference
   ------------------------------------------  ---------
   Request Failure 4xx
     439 First Hop Lacks Outbound Support      [RFC&rfc.number;]

11.7. Media Feature Tag

 This section registers a new media feature tag, per the procedures
 defined in [RFC2506].  The tag is placed into the sip tree, which is
 defined in [RFC3840].
 Media feature tag name:  sip.instance
 ASN.1 Identifier:  23
 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 (equality
    relationship).
 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.

Jennings, et al. Standards Track [Page 42] RFC 5626 Client-Initiated Connections in SIP October 2009

 Related standards or documents:  RFC 5626
 Security Considerations:  This media feature tag can be used in ways
    which affect application behaviors.  For example, the SIP caller
    preferences extension [RFC3841] 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
    that 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.

12. 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 ensure 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.
 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

Jennings, et al. Standards Track [Page 43] RFC 5626 Client-Initiated Connections in SIP October 2009

 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 [RFC3261] and [RFC3327] 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.

13. Operational Notes on Transports

 This entire section is non-normative.
 [RFC3261] 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.  [RFC3261] states that
 when a request is too large for UDP, the device sending the request
 attempts to switch over to TCP.  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 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.

14. 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.

Jennings, et al. Standards Track [Page 44] RFC 5626 Client-Initiated Connections in SIP October 2009

 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.

15. Acknowledgments

 Francois Audet acted as document shepherd for this document, tracking
 hundreds of comments and incorporating many grammatical fixes as well
 as prodding the editors to "get on with it".  Jonathan Rosenberg,
 Erkki Koivusalo, and Byron Campen 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
 document 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, Flemming Andreasen, Mike Hammer, Dan Wing,
 Srivatsa Srinivasan, Dale Worely, Juha Heinanen, Eric Rescorla,
 Lyndsay Campbell, Christer Holmberg, Kevin Johns, Jeroen van Bemmel,
 Derek MacDonald, Dean Willis, and Robert Sparks.

16. References

16.1. Normative References

 [RFC2119]      Bradner, S., "Key words for use in RFCs to Indicate
                Requirement Levels", BCP 14, RFC 2119, March 1997.
 [RFC2141]      Moats, R., "URN Syntax", RFC 2141, May 1997.
 [RFC2506]      Holtman, K., Mutz, A., and T. Hardie, "Media Feature
                Tag Registration Procedure", BCP 31, RFC 2506,
                March 1999.

Jennings, et al. Standards Track [Page 45] RFC 5626 Client-Initiated Connections in SIP October 2009

 [RFC3261]      Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                Johnston, A., Peterson, J., Sparks, R., Handley, M.,
                and E. Schooler, "SIP: Session Initiation Protocol",
                RFC 3261, June 2002.
 [RFC3263]      Rosenberg, J. and H. Schulzrinne, "Session Initiation
                Protocol (SIP): Locating SIP Servers", RFC 3263,
                June 2002.
 [RFC3327]      Willis, D. and B. Hoeneisen, "Session Initiation
                Protocol (SIP) Extension Header Field for Registering
                Non-Adjacent Contacts", RFC 3327, December 2002.
 [RFC3581]      Rosenberg, J. and H. Schulzrinne, "An Extension to the
                Session Initiation Protocol (SIP) for Symmetric
                Response Routing", RFC 3581, August 2003.
 [RFC3629]      Yergeau, F., "UTF-8, a transformation format of ISO
                10646", STD 63, RFC 3629, November 2003.
 [RFC3840]      Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
                "Indicating User Agent Capabilities in the Session
                Initiation Protocol (SIP)", RFC 3840, August 2004.
 [RFC3841]      Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
                "Caller Preferences for the Session Initiation
                Protocol (SIP)", RFC 3841, August 2004.
 [RFC3968]      Camarillo, G., "The Internet Assigned Number Authority
                (IANA) Header Field Parameter Registry for the Session
                Initiation Protocol (SIP)", BCP 98, RFC 3968,
                December 2004.
 [RFC3969]      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.
 [RFC4122]      Leach, P., Mealling, M., and R. Salz, "A Universally
                Unique IDentifier (UUID) URN Namespace", RFC 4122,
                July 2005.
 [RFC5234]      Crocker, D. and P. Overell, "Augmented BNF for Syntax
                Specifications: ABNF", STD 68, RFC 5234, January 2008.
 [RFC5389]      Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
                "Session Traversal Utilities for NAT (STUN)",
                RFC 5389, October 2008.

Jennings, et al. Standards Track [Page 46] RFC 5626 Client-Initiated Connections in SIP October 2009

16.2. Informative References

 [CONFIG-FMWK]  Petrie, D. and S. Channabasappa, Ed., "A Framework for
                Session Initiation Protocol User Agent Profile
                Delivery", Work in Progress, February 2008.
 [NAT-SCEN]     Boulton, C., Rosenberg, J., Camarillo, G., and F.
                Audet, "Best Current Practices for NAT Traversal for
                Client-Server SIP", Work in Progress, September 2008.
 [RFC0768]      Postel, J., "User Datagram Protocol", STD 6, RFC 768,
                August 1980.
 [RFC0793]      Postel, J., "Transmission Control Protocol", STD 7,
                RFC 793, September 1981.
 [RFC1035]      Mockapetris, P., "Domain names - implementation and
                specification", STD 13, RFC 1035, November 1987.
 [RFC2104]      Krawczyk, H., Bellare, M., and R. Canetti, "HMAC:
                Keyed-Hashing for Message Authentication", RFC 2104,
                February 1997.
 [RFC2131]      Droms, R., "Dynamic Host Configuration Protocol",
                RFC 2131, March 1997.
 [RFC2782]      Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR
                for specifying the location of services (DNS SRV)",
                RFC 2782, February 2000.
 [RFC3320]      Price, R., Bormann, C., Christoffersson, J., Hannu,
                H., Liu, Z., and J. Rosenberg, "Signaling Compression
                (SigComp)", RFC 3320, January 2003.
 [RFC3489]      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.
 [RFC3986]      Berners-Lee, T., Fielding, R., and L. Masinter,
                "Uniform Resource Identifier (URI): Generic Syntax",
                STD 66, RFC 3986, January 2005.
 [RFC4340]      Kohler, E., Handley, M., and S. Floyd, "Datagram
                Congestion Control Protocol (DCCP)", RFC 4340,
                March 2006.

Jennings, et al. Standards Track [Page 47] RFC 5626 Client-Initiated Connections in SIP October 2009

 [RFC4648]      Josefsson, S., "The Base16, Base32, and Base64 Data
                Encodings", RFC 4648, October 2006.
 [RFC4960]      Stewart, R., "Stream Control Transmission Protocol",
                RFC 4960, September 2007.
 [RFC5246]      Dierks, T. and E. Rescorla, "The Transport Layer
                Security (TLS) Protocol Version 1.2", RFC 5246,
                August 2008.
 [RFC5627]      Rosenberg, J., "Obtaining and Using Globally Routable
                User Agent URIs (GRUUs) in the Session Initiation
                Protocol (SIP)", RFC 5627, October 2009.

Jennings, et al. Standards Track [Page 48] RFC 5626 Client-Initiated Connections in SIP October 2009

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 amount of time the UA will wait to retry
 registration.
   +-------------------+--------------------+---------------------+
   | # of reg failures | all flows unusable | > 1 non-failed flow |
   +-------------------+--------------------+---------------------+
   | 0                 | 0 s                | 0 s                 |
   | 1                 | 30-60 s            | 90-180 s            |
   | 2                 | 1-2 min            | 3-6 min             |
   | 3                 | 2-4 min            | 6-12 min            |
   | 4                 | 4-8 min            | 12-24 min           |
   | 5                 | 8-16 min           | 15-30 min           |
   | 6 or more         | 15-30 min          | 15-30 min           |
   +-------------------+--------------------+---------------------+

Appendix B. ABNF

 This appendix contains the ABNF defined earlier in this document.
    CRLF = CR LF
    double-CRLF = CR LF CR LF
    CR = %x0D
    LF = %x0A
    Flow-Timer     = "Flow-Timer" HCOLON 1*DIGIT
    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
                      DQUOTE "<" instance-val ">" DQUOTE
    instance-val   = 1*uric ; defined in RFC 3261

Jennings, et al. Standards Track [Page 49] RFC 5626 Client-Initiated Connections in SIP October 2009

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
 Rohan Mahy (editor)
 Unaffiliated
 EMail: rohan@ekabal.com
 Francois Audet (editor)
 Skype Labs
 EMail: francois.audet@skypelabs.com

Jennings, et al. Standards Track [Page 50]

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