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Versions: 00 01 02 03 04 05 RFC 3263

Internet Engineering Task Force                                   SIP WG
Internet Draft                                               J.Rosenberg
                                                             dynamicsoft
                                                           H.Schulzrinne
                                                             Columbia U.
draft-ietf-sip-srv-05.txt
February 21, 2002
Expires: August 2002


                       SIP: Locating SIP Servers

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.


Abstract

   The Session Initiation Protocol (SIP) uses DNS procedures to allow a
   client to resolve a SIP 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.










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                           Table of Contents



   1          Introduction ........................................    3
   2          Problems DNS is Needed to Solve .....................    3
   3          Terminology .........................................    6
   4          Client Usage ........................................    6
   4.1        Selecting a Transport Protocol ......................    7
   4.2        Determining Port and IP .............................   10
   4.3        Details of RFC 2782 Process .........................   11
   4.4        Consideration for Stateless Proxies .................   11
   5          Server Usage ........................................   12
   6          Constructing SIP URIs ...............................   13
   7          Security Considerations .............................   14
   8          The Transport Determination Application .............   15
   9          IANA Considerations .................................   15
   10         Acknowledgements ....................................   16
   11         Author's Addresses ..................................   16
   12         Normative References ................................   16
   13         Non-Normative References ............................   17



























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1 Introduction

   (NOTE TO RFC EDITOR: Please replace all instances of RFC BBBB with
   the RFC number for draft-ietf-sip-rfc2543bis.)

   The Session Initiation Protocol (SIP) (RFC BBBB [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, on top of any
   reliable transport, which is just TCP at the moment. 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



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  ............................          ..............................
  .                          .          .                            .
  .                +-------+ .          . +-------+                  .
  .                |       | .          . |       |                  .
  .                | Proxy |------------- | Proxy |                  .
  .                |   1   | .          . |  2    |                  .
  .                |       | .          . |       |                  .
  .              / +-------+ .          . +-------+ \                .
  .             /            .          .            \               .
  .            /             .          .             \              .
  .           /              .          .              \             .
  .          /               .          .               \            .
  .         /                .          .                \           .
  .        /                 .          .                 \          .
  .       /                  .          .                  \         .
  .   +-------+              .          .                +-------+   .
  .   |       |              .          .                |       |   .
  .   |       |              .          .                |       |   .
  .   | UA 1  |              .          .                | UA 2  |   .
  .   |       |              .          .                |       |   .
  .   +-------+              .          .                +-------+   .
  .              Domain A    .          .   Domain B                 .
  ............................          ..............................
















   Figure 1: The SIP trapezoid


   chooses the protocol preferred by proxy 2.

   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



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   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
   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.



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   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.

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 BBBB [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



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   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).

   We defined 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 TARGET 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



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   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. 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 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. 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



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   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.

   The domain suffixes in the NAPTR replacement field SHOULD match the
   domain of the original query. The reason s 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.


        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.



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   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

   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
   determined previously. The client SHOULD try the one of the records.
   If an attempt should fail, based on the definition of failure in
   Section 4.3, another SHOULD be tried.

   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 records 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". 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.


        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 records will be used.

   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 at the specific port from the URI and
   transport protocol determined previously. Processing then proceeds as
   described above for an explicit port once the A or AAAA records have
   been looked up.




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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 use 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 BBBB [1] fires). If a failure occurs, the client
   SHOULD create a new request, which is identical to the previous, but
   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.




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   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 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 BBBB [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 transports,
   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 BBBB [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 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 BBBB). "Fails" is defined as any closure of the
   transport connection the request came in on before the response can



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   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.

   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 BBBB [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:




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   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 (indicated
   the preferred choice), and this record points to the specific element
   that constructed the URI. However, there are additional records with
   higher priority that point to backup elements that would be used in
   the event of failure. This allows the constraint of RFC BBBB [1] to
   be met while allowing for robust operation.

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 BBBB
   [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



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   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.

   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



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             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 are to be placed into the registry:


   Services Field               Protocol
   SIP+D2T                       TCP
   SIPS+D2T                      TCP
   SIP+D2U                       UDP
   SIP+D2S                       SCTP (RFC 2960)



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 Author's 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



12 Normative References

   [1] J. Rosenberg, H. Schulzrinne, et al.  , "SIP: Session initiation
   protocol," Internet Draft, Internet Engineering Task Force, Oct.
   2001.  Work in progress.




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   [2] A. Gulbrandsen, P. Vixie, and L. Esibov, "A DNS RR for specifying
   the location of services (DNS SRV)," Request for Comments 2782,
   Internet Engineering Task Force, Feb. 2000.

   [3] M. Mealling and R. Daniel, "The naming authority pointer (NAPTR)
   DNS resource record," Request for Comments 2915, Internet Engineering
   Task Force, Sept. 2000.

   [4] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," Request for Comments 2119, Internet Engineering Task Force,
   Mar. 1997.

   [5] T. Narten and H. Alvestrand, "Guidelines for writing an IANA
   considerations section in RFCs," Request for Comments 2434, Internet
   Engineering Task Force, Oct. 1998.

13 Non-Normative References

   [6] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
   session initiation protocol," Request for Comments 2543, Internet
   Engineering Task Force, Mar. 1999.

   [7] M. Mealling, "Dynamic delegation discovery system (DDDS) part
   one: The comprehensive DDDS standard," Internet Draft, Internet
   Engineering Task Force, Oct. 2001.  Work in progress.

   [8] M. Mealling, "Dynamic delegation discovery system (DDDS) part
   three: The DNS database," Internet Draft, Internet Engineering Task
   Force, Oct.  2001.  Work in progress.


   Full Copyright Statement

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

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   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
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   followed, or as required to translate it into languages other than
   English.



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   The limited permissions granted above are perpetual and will not be
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