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Network Working Group                                   C. Jennings, Ed.Request for Comments: 5626                                 Cisco SystemsUpdates:3261,3327                                         R. Mahy, Ed.Category: Standards Track                                   Unaffiliated                                                           F. Audet, Ed.                                                              Skype Labs                                                            October 2009Managing Client-Initiated Connectionsin 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 toBCP 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 inSection 4.e ofJennings, 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 Contents1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .42.  Conventions and Terminology  . . . . . . . . . . . . . . . . .52.1.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .53.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .63.1.  Summary of Mechanism . . . . . . . . . . . . . . . . . . .63.2.  Single Registrar and UA  . . . . . . . . . . . . . . . . .73.3.  Multiple Connections from a User Agent . . . . . . . . . .83.4.  Edge Proxies . . . . . . . . . . . . . . . . . . . . . . .103.5.  Keep-Alive Technique . . . . . . . . . . . . . . . . . . .113.5.1.  CRLF Keep-Alive Technique  . . . . . . . . . . . . . .123.5.2.  STUN Keep-Alive Technique  . . . . . . . . . . . . . .124.  User Agent Procedures  . . . . . . . . . . . . . . . . . . . .134.1.  Instance ID Creation . . . . . . . . . . . . . . . . . . .134.2.  Registrations  . . . . . . . . . . . . . . . . . . . . . .144.2.1.  Initial Registrations  . . . . . . . . . . . . . . . .144.2.2.  Subsequent REGISTER Requests . . . . . . . . . . . . .164.2.3.  Third-Party Registrations  . . . . . . . . . . . . . .174.3.  Sending Non-REGISTER Requests  . . . . . . . . . . . . . .174.4.  Keep-Alives and Detecting Flow Failure . . . . . . . . . .184.4.1.  Keep-Alive with CRLF . . . . . . . . . . . . . . . . .194.4.2.  Keep-Alive with STUN . . . . . . . . . . . . . . . . .214.5.  Flow Recovery  . . . . . . . . . . . . . . . . . . . . . .215.  Edge Proxy Procedures  . . . . . . . . . . . . . . . . . . . .225.1.  Processing Register Requests . . . . . . . . . . . . . . .225.2.  Generating Flow Tokens . . . . . . . . . . . . . . . . . .235.3.  Forwarding Non-REGISTER Requests . . . . . . . . . . . . .235.3.1.  Processing Incoming Requests . . . . . . . . . . . . .245.3.2.  Processing Outgoing Requests . . . . . . . . . . . . .245.4.  Edge Proxy Keep-Alive Handling . . . . . . . . . . . . . .256.  Registrar Procedures . . . . . . . . . . . . . . . . . . . . .257.  Authoritative Proxy Procedures: Forwarding Requests  . . . . .27Jennings, et al.            Standards Track                     [Page 2]

RFC 5626          Client-Initiated Connections in SIP       October 20098.  STUN Keep-Alive Processing . . . . . . . . . . . . . . . . . .288.1.  Use with SigComp . . . . . . . . . . . . . . . . . . . . .299.  Example Message Flow . . . . . . . . . . . . . . . . . . . . .309.1.  Subscription to Configuration Package  . . . . . . . . . .309.2.  Registration . . . . . . . . . . . . . . . . . . . . . . .329.3.  Incoming Call and Proxy Crash  . . . . . . . . . . . . . .349.4.  Re-Registration  . . . . . . . . . . . . . . . . . . . . .379.5.  Outgoing Call  . . . . . . . . . . . . . . . . . . . . . .3810. Grammar  . . . . . . . . . . . . . . . . . . . . . . . . . . .4011. IANA Considerations  . . . . . . . . . . . . . . . . . . . . .4011.1. Flow-Timer Header Field  . . . . . . . . . . . . . . . . .4011.2. "reg-id" Contact Header Field Parameter  . . . . . . . . .4011.3. SIP/SIPS URI Parameters  . . . . . . . . . . . . . . . . .4111.4. SIP Option Tag . . . . . . . . . . . . . . . . . . . . . .4111.5. 430 (Flow Failed) Response Code  . . . . . . . . . . . . .4111.6. 439 (First Hop Lacks Outbound Support) Response Code . . .4211.7. Media Feature Tag  . . . . . . . . . . . . . . . . . . . .4212. Security Considerations  . . . . . . . . . . . . . . . . . . .4313. Operational Notes on Transports  . . . . . . . . . . . . . . .4414. Requirements . . . . . . . . . . . . . . . . . . . . . . . . .4415. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . .4516. References . . . . . . . . . . . . . . . . . . . . . . . . . .4516.1. Normative References . . . . . . . . . . . . . . . . . . .4516.2. Informative References . . . . . . . . . . . . . . . . . .47Appendix A.  Default Flow Registration Backoff Times . . . . . . .49Appendix B.  ABNF  . . . . . . . . . . . . . . . . . . . . . . . .49Jennings, et al.            Standards Track                     [Page 3]

RFC 5626          Client-Initiated Connections in SIP       October 20091.  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 inSection 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 20092.  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-feedJennings, 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 inSection 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 inSection 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 newJennings, 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 inSection 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 inSection 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 inSection 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 usedJennings, 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 Procedures4.1.  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 inSection 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.  Registrations4.2.1.  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 inSection 4.2.3, MUST   include an instance-id media feature tag as specified inSection 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 supportJennings, 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   inSection 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.  SeeSection 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 20094.2.3.  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 inSection 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 inSection 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 bySection 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 Sections4.4.1 or4.4.2.  If   a flow with a registration has failed, the UA follows the procedures   inSection 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 = %x0AJennings, 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 20094.4.2.  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 inSection 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 inSection 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 Procedures5.1.  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, asJennings, 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   inSection 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 inSection 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   inSection 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.  SeeSection 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 inSection 11.4) in a Supported header fieldJennings, 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 toRFC3261, 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 forJennings, 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: 0Jennings, 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: 09.3.  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 200910.  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 inRFC 3261   The value of the reg-id MUST NOT be 0 and MUST be less than 2^31.11.  IANA Considerations11.1.  Flow-Timer Header Field   This specification defines a new SIP header field "Flow-Timer" whose   syntax is defined inSection 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 inSection 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 200911.3.  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 inSection 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 inSection 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 inSection 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 200911.6.  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 inSection 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 ifJennings, 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 seeSection 5.2.  A discussion of   preventing the avalanche restart problem is inSection 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.  References16.1.  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 200916.2.  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 2009Appendix A.  Default Flow Registration Backoff Times   The base-time used for the flow re-registration backoff times   described inSection 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 inRFC 3261Jennings, et al.            Standards Track                    [Page 49]

RFC 5626          Client-Initiated Connections in SIP       October 2009Authors' 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.comJennings, et al.            Standards Track                    [Page 50]