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rfc:rfc3263

Network Working Group J. Rosenberg Request for Comments: 3263 dynamicsoft Obsoletes: 2543 H. Schulzrinne Category: Standards Track Columbia U.

                                                             June 2002
      Session Initiation Protocol (SIP): Locating SIP Servers

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) The Internet Society (2002).  All Rights Reserved.

Abstract

 The Session Initiation Protocol (SIP) uses DNS procedures to allow a
 client to resolve a SIP Uniform Resource Identifier (URI) into the IP
 address, port, and transport protocol of the next hop to contact.  It
 also uses DNS to allow a server to send a response to a backup client
 if the primary client has failed.  This document describes those DNS
 procedures in detail.

Table of Contents

 1          Introduction ........................................    2
 2          Problems DNS is Needed to Solve .....................    2
 3          Terminology .........................................    5
 4          Client Usage ........................................    5
 4.1        Selecting a Transport Protocol ......................    6
 4.2        Determining Port and IP Address .....................    8
 4.3        Details of RFC 2782 Process .........................    9
 4.4        Consideration for Stateless Proxies .................   10
 5          Server Usage ........................................   11
 6          Constructing SIP URIs ...............................   12
 7          Security Considerations .............................   12
 8          The Transport Determination Application .............   13
 9          IANA Considerations .................................   14
 10         Acknowledgements ....................................   14
 11         Normative References ................................   15
 12         Informative References ..............................   15

Rosenberg & Schulzrinne Standards Track [Page 1] RFC 3263 SIP: Locating SIP Servers June 2002

 13         Authors' Addresses ..................................   16
 14         Full Copyright Statement ............................   17

1 Introduction

 The Session Initiation Protocol (SIP) (RFC 3261 [1]) is a client-
 server protocol used for the initiation and management of
 communications sessions between users.  SIP end systems are called
 user agents, and intermediate elements are known as proxy servers.  A
 typical SIP configuration, referred to as the SIP "trapezoid", is
 shown in Figure 1.  In this diagram, a caller in domain A (UA1)
 wishes to call Joe in domain B (joe@B).  To do so, it communicates
 with proxy 1 in its domain (domain A).  Proxy 1 forwards the request
 to the proxy for the domain of the called party (domain B), which is
 proxy 2.  Proxy 2 forwards the call to the called party, UA 2.
 As part of this call flow, proxy 1 needs to determine a SIP server
 for domain B.  To do this, proxy 1 makes use of DNS procedures, using
 both SRV [2] and NAPTR [3] records.  This document describes the
 specific problems that SIP uses DNS to help solve, and provides a
 solution.

2 Problems DNS is Needed to Solve

 DNS is needed to help solve two aspects of the general call flow
 described in the Introduction.  The first is for proxy 1 to discover
 the SIP server in domain B, in order to forward the call for joe@B.
 The second is for proxy 2 to identify a backup for proxy 1 in the
 event it fails after forwarding the request.
 For the first aspect, proxy 1 specifically needs to determine the IP
 address, port, and transport protocol for the server in domain B.
 The choice of transport protocol is particularly noteworthy.  Unlike
 many other protocols, SIP can run over a variety of transport
 protocols, including TCP, UDP, and SCTP.  SIP can also use TLS.
 Currently, use of TLS is defined for TCP only.  Thus, clients need to
 be able to automatically determine which transport protocols are
 available.  The proxy sending the request has a particular set of
 transport protocols it supports and a preference for using those
 transport protocols.  Proxy 2 has its own set of transport protocols
 it supports, and relative preferences for those transport protocols.
 All proxies must implement both UDP and TCP, along with TLS over TCP,
 so that there is always an intersection of capabilities.  Some form
 of DNS procedures are needed for proxy 1 to discover the available
 transport protocols for SIP services at domain B, and the relative
 preferences of those transport protocols.  Proxy 1 intersects its
 list of supported transport protocols with those of proxy 2 and then
 chooses the protocol preferred by proxy 2.

Rosenberg & Schulzrinne Standards Track [Page 2] RFC 3263 SIP: Locating SIP Servers June 2002

  ............................          ..............................
  .                          .          .                            .
  .                +-------+ .          . +-------+                  .
  .                |       | .          . |       |                  .
  .                | Proxy |------------- | Proxy |                  .
  .                |   1   | .          . |  2    |                  .
  .                |       | .          . |       |                  .
  .              / +-------+ .          . +-------+ \                .
  .             /            .          .            \               .
  .            /             .          .             \              .
  .           /              .          .              \             .
  .          /               .          .               \            .
  .         /                .          .                \           .
  .        /                 .          .                 \          .
  .       /                  .          .                  \         .
  .   +-------+              .          .                +-------+   .
  .   |       |              .          .                |       |   .
  .   |       |              .          .                |       |   .
  .   | UA 1  |              .          .                | UA 2  |   .
  .   |       |              .          .                |       |   .
  .   +-------+              .          .                +-------+   .
  .              Domain A    .          .   Domain B                 .
  ............................          ..............................
                      Figure 1: The SIP trapezoid
 It is important to note that DNS lookups can be used multiple times
 throughout the processing of a call.  In general, an element that
 wishes to send a request (called a client) may need to perform DNS
 processing to determine the IP address, port, and transport protocol
 of a next hop element, called a server (it can be a proxy or a user
 agent).  Such processing could, in principle, occur at every hop
 between elements.
 Since SIP is used for the establishment of interactive communications
 services, the time it takes to complete a transaction between a
 caller and called party is important.  Typically, the time from when
 the caller initiates a call until the time the called party is
 alerted should be no more than a few seconds.  Given that there can
 be multiple hops, each of which is doing DNS lookups in addition to
 other potentially time-intensive operations, the amount of time
 available for DNS lookups at each hop is limited.
 Scalability and high availability are important in SIP. SIP services
 scale up through clustering techniques.  Typically, in a realistic
 version of the network in Figure 1, proxy 2 would be a cluster of
 homogeneously configured proxies.  DNS needs to provide the ability

Rosenberg & Schulzrinne Standards Track [Page 3] RFC 3263 SIP: Locating SIP Servers June 2002

 for domain B to configure a set of servers, along with prioritization
 and weights, in order to provide a crude level of capacity-based load
 balancing.
 SIP assures high availability by having upstream elements detect
 failures.  For example, assume that proxy 2 is implemented as a
 cluster of two proxies, proxy 2.1 and proxy 2.2.  If proxy 1 sends a
 request to proxy 2.1 and the request fails, it retries the request by
 sending it to proxy 2.2.  In many cases, proxy 1 will not know which
 domains it will ultimately communicate with.  That information would
 be known when a user actually makes a call to another user in that
 domain.  Proxy 1 may never communicate with that domain again after
 the call completes.  Proxy 1 may communicate with thousands of
 different domains within a few minutes, and proxy 2 could receive
 requests from thousands of different domains within a few minutes.
 Because of this "many-to-many" relationship, and the possibly long
 intervals between communications between a pair of domains, it is not
 generally possible for an element to maintain dynamic availability
 state for the proxies it will communicate with.  When a proxy gets
 its first call with a particular domain, it will try the servers in
 that domain in some order until it finds one that is available.  The
 identity of the available server would ideally be cached for some
 amount of time in order to reduce call setup delays of subsequent
 calls.  The client cannot query a failed server continuously to
 determine when it becomes available again, since this does not scale.
 Furthermore, the availability state must eventually be flushed in
 order to redistribute load to recovered elements when they come back
 online.
 It is possible for elements to fail in the middle of a transaction.
 For example, after proxy 2 forwards the request to UA 2, proxy 1
 fails.  UA 2 sends its response to proxy 2, which tries to forward it
 to proxy 1, which is no longer available.  The second aspect of the
 flow in the introduction for which DNS is needed, is for proxy 2 to
 identify a backup for proxy 1 that it can send the response to.  This
 problem is more realistic in SIP than it is in other transactional
 protocols.  The reason is that some SIP responses can take a long
 time to be generated, because a human user frequently needs to be
 consulted in order to generate that response.  As such, it is not
 uncommon for tens of seconds to elapse between a call request and its
 acceptance.

Rosenberg & Schulzrinne Standards Track [Page 4] RFC 3263 SIP: Locating SIP Servers June 2002

3 Terminology

 In this document, the key words "MUST", "MUST NOT", "REQUIRED",
 "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
 and "OPTIONAL" are to be interpreted as described in RFC 2119 [4] and
 indicate requirement levels for compliant SIP implementations.

4 Client Usage

 Usage of DNS differs for clients and for servers.  This section
 discusses client usage.  We assume that the client is stateful
 (either a User Agent Client (UAC) or a stateful proxy).  Stateless
 proxies are discussed in Section 4.4.
 The procedures here are invoked when a client needs to send a request
 to a resource identified by a SIP or SIPS (secure SIP) URI.  This URI
 can identify the desired resource to which the request is targeted
 (in which case, the URI is found in the Request-URI), or it can
 identify an intermediate hop towards that resource (in which case,
 the URI is found in the Route header).  The procedures defined here
 in no way affect this URI (i.e., the URI is not rewritten with the
 result of the DNS lookup), they only result in an IP address, port
 and transport protocol where the request can be sent.  RFC 3261 [1]
 provides guidelines on determining which URI needs to be resolved in
 DNS to determine the host that the request needs to be sent to.  In
 some cases, also documented in [1], the request can be sent to a
 specific intermediate proxy not identified by a SIP URI, but rather,
 by a hostname or numeric IP address.  In that case, a temporary URI,
 used for purposes of this specification, is constructed.  That URI is
 of the form sip:<proxy>, where <proxy> is the FQDN or numeric IP
 address of the next-hop proxy.  As a result, in all cases, the
 problem boils down to resolution of a SIP or SIPS URI in DNS to
 determine the IP address, port, and transport of the host to which
 the request is to be sent.
 The procedures here MUST be done exactly once per transaction, where
 transaction is as defined in [1].  That is, once a SIP server has
 successfully been contacted (success is defined below), all
 retransmissions of the SIP request and the ACK for non-2xx SIP
 responses to INVITE MUST be sent to the same host.  Furthermore, a
 CANCEL for a particular SIP request MUST be sent to the same SIP
 server that the SIP request was delivered to.
 Because the ACK request for 2xx responses to INVITE constitutes a
 different transaction, there is no requirement that it be delivered
 to the same server that received the original request (indeed, if
 that server did not record-route, it will not get the ACK).

Rosenberg & Schulzrinne Standards Track [Page 5] RFC 3263 SIP: Locating SIP Servers June 2002

 We define TARGET as the value of the maddr parameter of the URI, if
 present, otherwise, the host value of the hostport component of the
 URI.  It identifies the domain to be contacted.  A description of the
 SIP and SIPS URIs and a definition of these parameters can be found
 in [1].
 We determine the transport protocol, port and IP address of a
 suitable instance of TARGET in Sections 4.1 and 4.2.

4.1 Selecting a Transport Protocol

 First, the client selects a transport protocol.
 If the URI specifies a transport protocol in the transport parameter,
 that transport protocol SHOULD be used.
 Otherwise, if no transport protocol is specified, but the TARGET is a
 numeric IP address, the client SHOULD use UDP for a SIP URI, and TCP
 for a SIPS URI.  Similarly, if no transport protocol is specified,
 and the TARGET is not numeric, but an explicit port is provided, the
 client SHOULD use UDP for a SIP URI, and TCP for a SIPS URI.  This is
 because UDP is the only mandatory transport in RFC 2543 [6], and thus
 the only one guaranteed to be interoperable for a SIP URI.  It was
 also specified as the default transport in RFC 2543 when no transport
 was present in the SIP URI.  However, another transport, such as TCP,
 MAY be used if the guidelines of SIP mandate it for this particular
 request.  That is the case, for example, for requests that exceed the
 path MTU.
 Otherwise, if no transport protocol or port is specified, and the
 target is not a numeric IP address, the client SHOULD perform a NAPTR
 query for the domain in the URI.  The services relevant for the task
 of transport protocol selection are those with NAPTR service fields
 with values "SIP+D2X" and "SIPS+D2X", where X is a letter that
 corresponds to a transport protocol supported by the domain.  This
 specification defines D2U for UDP, D2T for TCP, and D2S for SCTP.  We
 also establish an IANA registry for NAPTR service name to transport
 protocol mappings.
 These NAPTR records provide a mapping from a domain to the SRV record
 for contacting a server with the specific transport protocol in the
 NAPTR services field.  The resource record will contain an empty
 regular expression and a replacement value, which is the SRV record
 for that particular transport protocol.  If the server supports
 multiple transport protocols, there will be multiple NAPTR records,
 each with a different service value.  As per RFC 2915 [3], the client
 discards any records whose services fields are not applicable.  For
 the purposes of this specification, several rules are defined.

Rosenberg & Schulzrinne Standards Track [Page 6] RFC 3263 SIP: Locating SIP Servers June 2002

 First, a client resolving a SIPS URI MUST discard any services that
 do not contain "SIPS" as the protocol in the service field.  The
 converse is not true, however.  A client resolving a SIP URI SHOULD
 retain records with "SIPS" as the protocol, if the client supports
 TLS.  Second, a client MUST discard any service fields that identify
 a resolution service whose value is not "D2X", for values of X that
 indicate transport protocols supported by the client.  The NAPTR
 processing as described in RFC 2915 will result in the discovery of
 the most preferred transport protocol of the server that is supported
 by the client, as well as an SRV record for the server.  It will also
 allow the client to discover if TLS is available and its preference
 for its usage.
 As an example, consider a client that wishes to resolve
 sip:user@example.com.  The client performs a NAPTR query for that
 domain, and the following NAPTR records are returned:
 ;          order pref flags service      regexp  replacement
    IN NAPTR 50   50  "s"  "SIPS+D2T"     ""  _sips._tcp.example.com.
    IN NAPTR 90   50  "s"  "SIP+D2T"      ""  _sip._tcp.example.com
    IN NAPTR 100  50  "s"  "SIP+D2U"      ""  _sip._udp.example.com.
 This indicates that the server supports TLS over TCP, TCP, and UDP,
 in that order of preference.  Since the client supports TCP and UDP,
 TCP will be used, targeted to a host determined by an SRV lookup of
 _sip._tcp.example.com.  That lookup would return:
 ;;          Priority Weight Port   Target
     IN SRV  0        1      5060   server1.example.com
     IN SRV  0        2      5060   server2.example.com
 If a SIP proxy, redirect server, or registrar is to be contacted
 through the lookup of NAPTR records, there MUST be at least three
 records - one with a "SIP+D2T" service field, one with a "SIP+D2U"
 service field, and one with a "SIPS+D2T" service field.  The records
 with SIPS as the protocol in the service field SHOULD be preferred
 (i.e., have a lower value of the order field) above records with SIP
 as the protocol in the service field.  A record with a "SIPS+D2U"
 service field SHOULD NOT be placed into the DNS, since it is not
 possible to use TLS over UDP.
 It is not necessary for the domain suffixes in the NAPTR replacement
 field to match the domain of the original query (i.e., example.com
 above).  However, for backwards compatibility with RFC 2543, a domain
 MUST maintain SRV records for the domain of the original query, even
 if the NAPTR record is in a different domain.  As an example, even
 though the SRV record for TCP is _sip._tcp.school.edu, there MUST
 also be an SRV record at _sip._tcp.example.com.

Rosenberg & Schulzrinne Standards Track [Page 7] RFC 3263 SIP: Locating SIP Servers June 2002

    RFC 2543 will look up the SRV records for the domain directly.  If
    these do not exist because the NAPTR replacement points to a
    different domain, the client will fail.
 For NAPTR records with SIPS protocol fields, (if the server is using
 a site certificate), the domain name in the query and the domain name
 in the replacement field MUST both be valid based on the site
 certificate handed out by the server in the TLS exchange.  Similarly,
 the domain name in the SRV query and the domain name in the target in
 the SRV record MUST both be valid based on the same site certificate.
 Otherwise, an attacker could modify the DNS records to contain
 replacement values in a different domain, and the client could not
 validate that this was the desired behavior or the result of an
 attack.
 If no NAPTR records are found, the client constructs SRV queries for
 those transport protocols it supports, and does a query for each.
 Queries are done using the service identifier "_sip" for SIP URIs and
 "_sips" for SIPS URIs.  A particular transport is supported if the
 query is successful.  The client MAY use any transport protocol it
 desires which is supported by the server.
    This is a change from RFC 2543.  It specified that a client would
    lookup SRV records for all transports it supported, and merge the
    priority values across those records.  Then, it would choose the
    most preferred record.
 If no SRV records are found, the client SHOULD use TCP for a SIPS
 URI, and UDP for a SIP URI.  However, another transport protocol,
 such as TCP, MAY be used if the guidelines of SIP mandate it for this
 particular request.  That is the case, for example, for requests that
 exceed the path MTU.

4.2 Determining Port and IP Address

 Once the transport protocol has been determined, the next step is to
 determine the IP address and port.
 If TARGET is a numeric IP address, the client uses that address.  If
 the URI also contains a port, it uses that port.  If no port is
 specified, it uses the default port for the particular transport
 protocol.
 If the TARGET was not a numeric IP address, but a port is present in
 the URI, the client performs an A or AAAA record lookup of the domain
 name.  The result will be a list of IP addresses, each of which can
 be contacted at the specific port from the URI and transport protocol

Rosenberg & Schulzrinne Standards Track [Page 8] RFC 3263 SIP: Locating SIP Servers June 2002

 determined previously.  The client SHOULD try the first record.  If
 an attempt should fail, based on the definition of failure in Section
 4.3, the next SHOULD be tried, and if that should fail, the next
 SHOULD be tried, and so on.
    This is a change from RFC 2543.  Previously, if the port was
    explicit, but with a value of 5060, SRV records were used.  Now, A
    or AAAA records will be used.
 If the TARGET was not a numeric IP address, and no port was present
 in the URI, the client performs an SRV query on the record returned
 from the NAPTR processing of Section 4.1, if such processing was
 performed.  If it was not, because a transport was specified
 explicitly, the client performs an SRV query for that specific
 transport, using the service identifier "_sips" for SIPS URIs.  For a
 SIP URI, if the client wishes to use TLS, it also uses the service
 identifier "_sips" for that specific transport, otherwise, it uses
 "_sip".  If the NAPTR processing was not done because no NAPTR
 records were found, but an SRV query for a supported transport
 protocol was successful, those SRV records are selected. Irregardless
 of how the SRV records were determined, the procedures of RFC 2782,
 as described in the section titled "Usage rules" are followed,
 augmented by the additional procedures of Section 4.3 of this
 document.
 If no SRV records were found, the client performs an A or AAAA record
 lookup of the domain name.  The result will be a list of IP
 addresses, each of which can be contacted using the transport
 protocol determined previously, at the default port for that
 transport.  Processing then proceeds as described above for an
 explicit port once the A or AAAA records have been looked up.

4.3 Details of RFC 2782 Process

 RFC 2782 spells out the details of how a set of SRV records are
 sorted and then tried.  However, it only states that the client
 should "try to connect to the (protocol, address, service)" without
 giving any details on what happens in the event of failure.  Those
 details are described here for SIP.
 For SIP requests, failure occurs if the transaction layer reports a
 503 error response or a transport failure of some sort (generally,
 due to fatal ICMP errors in UDP or connection failures in TCP).
 Failure also occurs if the transaction layer times out without ever
 having received any response, provisional or final (i.e., timer B or
 timer F in RFC 3261 [1] fires).  If a failure occurs, the client
 SHOULD create a new request, which is identical to the previous, but

Rosenberg & Schulzrinne Standards Track [Page 9] RFC 3263 SIP: Locating SIP Servers June 2002

 has a different value of the Via branch ID than the previous (and
 therefore constitutes a new SIP transaction).  That request is sent
 to the next element in the list as specified by RFC 2782.

4.4 Consideration for Stateless Proxies

 The process of the previous sections is highly stateful.  When a
 server is contacted successfully, all retransmissions of the request
 for the transaction, as well as ACK for a non-2xx final response, and
 CANCEL requests for that transaction, MUST go to the same server.
 The identity of the successfully contacted server is a form of
 transaction state.  This presents a challenge for stateless proxies,
 which still need to meet the requirement for sending all requests in
 the transaction to the same server.
 The problem is similar, but different, to the problem of HTTP
 transactions within a cookie session getting routed to different
 servers based on DNS randomization.  There, such distribution is not
 a problem.  Farms of servers generally have common back-end data
 stores, where the session data is stored.  Whenever a server in the
 farm receives an HTTP request, it takes the session identifier, if
 present, and extracts the needed state to process the request.  A
 request without a session identifier creates a new one.  The problem
 with stateless proxies is at a lower layer; it is retransmitted
 requests within a transaction that are being potentially spread
 across servers.  Since none of these retransmissions carries a
 "session identifier" (a complete dialog identifier in SIP terms), a
 new dialog would be created identically at each server.  This could,
 for example result in multiple phone calls to be made to the same
 phone.  Therefore, it is critical to prevent such a thing from
 happening in the first place.
 The requirement is not difficult to meet in the simple case where
 there were no failures when attempting to contact a server.  Whenever
 the stateless proxy receives the request, it performs the appropriate
 DNS queries as described above.  However, the procedures of RFC 2782
 are not guaranteed to be deterministic.  This is because records that
 contain the same priority have no specified order.  The stateless
 proxy MUST define a deterministic order to the records in that case,
 using any algorithm at its disposal.  One suggestion is to
 alphabetize them, or, more generally, sort them by ASCII-compatible
 encoding.  To make processing easier for stateless proxies, it is
 RECOMMENDED that domain administrators make the weights of SRV
 records with equal priority different (for example, using weights of
 1000 and 1001 if two servers are equivalent, rather than assigning
 both a weight of 1000), and similarly for NAPTR records.  If the
 first server is contacted successfully, the proxy can remain

Rosenberg & Schulzrinne Standards Track [Page 10] RFC 3263 SIP: Locating SIP Servers June 2002

 stateless.  However, if the first server is not contacted
 successfully, and a subsequent server is, the proxy cannot remain
 stateless for this transaction.  If it were stateless, a
 retransmission could very well go to a different server if the failed
 one recovers between retransmissions.  As such, whenever a proxy does
 not successfully contact the first server, it SHOULD act as a
 stateful proxy.
 Unfortunately, it is still possible for a stateless proxy to deliver
 retransmissions to different servers, even if it follows the
 recommendations above.  This can happen if the DNS TTLs expire in the
 middle of a transaction, and the entries had changed.  This is
 unavoidable.  Network implementors should be aware of this
 limitation, and not use stateless proxies that access DNS if this
 error is deemed critical.

5 Server Usage

 RFC 3261 [1] defines procedures for sending responses from a server
 back to the client.  Typically, for unicast UDP requests, the
 response is sent back to the source IP address where the request came
 from, using the port contained in the Via header.  For reliable
 transport protocols, the response is sent over the connection the
 request arrived on.  However, it is important to provide failover
 support when the client element fails between sending the request and
 receiving the response.
 A server, according to RFC 3261 [1], will send a response on the
 connection it arrived on (in the case of reliable transport
 protocols), and for unreliable transport protocols, to the source
 address of the request, and the port in the Via header field.  The
 procedures here are invoked when a server attempts to send to that
 location and that response fails (the specific conditions are
 detailed in RFC 3261). "Fails" is defined as any closure of the
 transport connection the request came in on before the response can
 be sent, or communication of a fatal error from the transport layer.
 In these cases, the server examines the value of the sent-by
 construction in the topmost Via header.  If it contains a numeric IP
 address, the server attempts to send the response to that address,
 using the transport protocol from the Via header, and the port from
 sent-by, if present, else the default for that transport protocol.
 The transport protocol in the Via header can indicate "TLS", which
 refers to TLS over TCP.  When this value is present, the server MUST
 use TLS over TCP to send the response.

Rosenberg & Schulzrinne Standards Track [Page 11] RFC 3263 SIP: Locating SIP Servers June 2002

 If, however, the sent-by field contained a domain name and a port
 number, the server queries for A or AAAA records with that name.  It
 tries to send the response to each element on the resulting list of
 IP addresses, using the port from the Via, and the transport protocol
 from the Via (again, a value of TLS refers to TLS over TCP).  As in
 the client processing, the next entry in the list is tried if the one
 before it results in a failure.
 If, however, the sent-by field contained a domain name and no port,
 the server queries for SRV records at that domain name using the
 service identifier "_sips" if the Via transport is "TLS", "_sip"
 otherwise, and the transport from the topmost Via header ("TLS"
 implies that the transport protocol in the SRV query is TCP).  The
 resulting list is sorted as described in [2], and the response is
 sent to the topmost element on the new list described there.  If that
 results in a failure, the next entry on the list is tried.

6 Constructing SIP URIs

 In many cases, an element needs to construct a SIP URI for inclusion
 in a Contact header in a REGISTER, or in a Record-Route header in an
 INVITE.  According to RFC 3261 [1], these URIs have to have the
 property that they resolve to the specific element that inserted
 them.  However, if they are constructed with just an IP address, for
 example:
 sip:1.2.3.4
 then should the element fail, there is no way to route the request or
 response through a backup.
 SRV provides a way to fix this.  Instead of using an IP address, a
 domain name that resolves to an SRV record can be used:
 sip:server23.provider.com
 The SRV records for a particular target can be set up so that there
 is a single record with a low value for the priority field
 (indicating the preferred choice), and this record points to the
 specific element that constructed the URI.  However, there are
 additional records with higher values of the priority field that
 point to backup elements that would be used in the event of failure.
 This allows the constraint of RFC 3261 [1] to be met while allowing
 for robust operation.

Rosenberg & Schulzrinne Standards Track [Page 12] RFC 3263 SIP: Locating SIP Servers June 2002

7 Security Considerations

 DNS NAPTR records are used to allow a client to discover that the
 server supports TLS.  An attacker could potentially modify these
 records, resulting in a client using a non-secure transport when TLS
 is in fact available and preferred.
 This is partially mitigated by the presence of the sips URI scheme,
 which is always sent only over TLS.  An attacker cannot force a bid
 down through deletion or modification of DNS records.  In the worst
 case, they can prevent communication from occurring by deleting all
 records.  A sips URI itself is generally exchanged within a secure
 context, frequently on a business card or secure web page, or within
 a SIP message which has already been secured with TLS.  See RFC 3261
 [1] for details.  The sips URI is therefore preferred when security
 is truly needed, but we allow TLS to be used for requests resolved by
 a SIP URI to allow security that is better than no TLS at all.
 The bid down attack can also be mitigated through caching.  A client
 which frequently contacts the same domain SHOULD cache whether or not
 its NAPTR records contain SIPS in the services field.  If such
 records were present, but in later queries cease to appear, it is a
 sign of a potential attack.  In this case, the client SHOULD generate
 some kind of alert or alarm, and MAY reject the request.
 An additional problem is that proxies, which are intermediaries
 between the users of the system, are frequently the clients that
 perform the NAPTR queries.  It is therefore possible for a proxy to
 ignore SIPS entries even though they are present, resulting in
 downgraded security.  There is very little that can be done to
 prevent such attacks.  Clients are simply dependent on proxy servers
 for call completion, and must trust that they implement the protocol
 properly in order for security to be provided.  Falsifying DNS
 records can be done by tampering with wire traffic (in the absence of
 DNSSEC), whereas compromising and commandeering a proxy server
 requires a break-in, and is seen as the considerably less likely
 downgrade threat.

8 The Transport Determination Application

 This section more formally defines the NAPTR usage of this
 specification, using the Dynamic Delegation Discovery System (DDDS)
 framework as a guide [7].  DDDS represents the evolution of the NAPTR
 resource record.  DDDS defines applications, which can make use of
 the NAPTR record for specific resolution services.  This application
 is called the Transport Determination Application, and its goal is to
 map an incoming SIP or SIPS URI to a set of SRV records for the
 various servers that can handle the URI.

Rosenberg & Schulzrinne Standards Track [Page 13] RFC 3263 SIP: Locating SIP Servers June 2002

 The following is the information that DDDS requests an application to
 provide:
    Application Unique String: The Application Unique String (AUS) is
       the input to the resolution service.  For this application, it
       is the URI to resolve.
    First Well Known Rule: The first well known rule extracts a key
       from the AUS.  For this application, the first well known rule
       extracts the host portion of the SIP or SIPS URI.
    Valid Databases: The key resulting from the first well known rule
       is looked up in a single database, the DNS [8].
    Expected Output: The result of the application is an SRV record
       for the server to contact.

9 IANA Considerations

 The usage of NAPTR records described here requires well known values
 for the service fields for each transport supported by SIP.  The
 table of mappings from service field values to transport protocols is
 to be maintained by IANA.  New entries in the table MAY be added
 through the publication of standards track RFCs, as described in RFC
 2434 [5].
 The registration in the RFC MUST include the following information:
    Service Field: The service field being registered.  An example for
       a new fictitious transport protocol called NCTP might be
       "SIP+D2N".
    Protocol: The specific transport protocol associated with that
       service field.  This MUST include the name and acronym for the
       protocol, along with reference to a document that describes the
       transport protocol.  For example - "New Connectionless
       Transport Protocol (NCTP), RFC 5766".
    Name and Contact Information: The name, address, email address and
       telephone number for the person performing the registration.
 The following values have been placed into the registry:
 Services Field               Protocol
 SIP+D2T                       TCP
 SIPS+D2T                      TCP
 SIP+D2U                       UDP
 SIP+D2S                       SCTP (RFC 2960)

Rosenberg & Schulzrinne Standards Track [Page 14] RFC 3263 SIP: Locating SIP Servers June 2002

10 Acknowledgements

 The authors would like to thank Randy Bush, Leslie Daigle, Patrik
 Faltstrom, Jo Hornsby, Rohan Mahy, Allison Mankin, Michael Mealling,
 Thomas Narten, and Jon Peterson for their useful comments.

11 Normative References

 [1]   Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
       Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
       Session Initiation Protocol", RFC 3261, June 2002.
 [2]   Gulbrandsen, A., Vixie, P. and L. Esibov, "A DNS RR for
       Specifying the Location of Services (DNS SRV)", RFC 2782,
       February 2000.
 [3]   Mealling, M. and R. Daniel, "The Naming Authority Pointer
       (NAPTR) DNS Resource Record", RFC 2915, September 2000.
 [4]   Bradner, S., "Key Words for Use in RFCs to Indicate Requirement
       Levels", BCP 14, RFC 2119, March 1997.
 [5]   Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
       Considerations Section in RFCs", BCP 26, RFC 2434, October
       1998.

12 Informative References

 [6]   Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
       "SIP: Session Initiation Protocol", RFC 2543, March 1999.
 [7]   Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part
       One: The Comprehensive DDDS Standard", Work in Progress.
 [8]   Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part
       Three: The DNS Database", Work in Progress.

Rosenberg & Schulzrinne Standards Track [Page 15] RFC 3263 SIP: Locating SIP Servers June 2002

13 Authors' Addresses

 Jonathan Rosenberg
 dynamicsoft
 72 Eagle Rock Avenue
 First Floor
 East Hanover, NJ 07936
 EMail: jdrosen@dynamicsoft.com
 Henning Schulzrinne
 Columbia University
 M/S 0401
 1214 Amsterdam Ave.
 New York, NY 10027-7003
 EMail: schulzrinne@cs.columbia.edu

Rosenberg & Schulzrinne Standards Track [Page 16] RFC 3263 SIP: Locating SIP Servers June 2002

14 Full Copyright Statement

 Copyright (C) The Internet Society (2002).  All Rights Reserved.
 This document and translations of it may be copied and furnished to
 others, and derivative works that comment on or otherwise explain it
 or assist in its implementation may be prepared, copied, published
 and distributed, in whole or in part, without restriction of any
 kind, provided that the above copyright notice and this paragraph are
 included on all such copies and derivative works.  However, this
 document itself may not be modified in any way, such as by removing
 the copyright notice or references to the Internet Society or other
 Internet organizations, except as needed for the purpose of
 developing Internet standards in which case the procedures for
 copyrights defined in the Internet Standards process must be
 followed, or as required to translate it into languages other than
 English.
 The limited permissions granted above are perpetual and will not be
 revoked by the Internet Society or its successors or assigns.
 This document and the information contained herein is provided on an
 "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
 BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgement

 Funding for the RFC Editor function is currently provided by the
 Internet Society.

Rosenberg & Schulzrinne Standards Track [Page 17]

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