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Internet Engineering Task Force                                   SIP WG
Internet Draft                                 J.Rosenberg,H.Schulzrinne
draft-ietf-sip-srv-03.txt                        dynamicsoft,Columbia U.
December 24, 2001
Expires: May 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) makes use of DNS procedures to
   allow a client to resolve a SIP URI into the IP address, port, and
   transport of the next hop to contact. It also uses DNS to allow a
   server to send a response to a backup client in the event of a
   failure of the primary client. This document describes those DNS pro-
   cedures in detail.


1 Introduction

   The Session Initiation Protocol (SIP) [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 intermedi-
   ate elements are known as proxy servers. A typical SIP configuration,
   referred to as the SIP "trapezoid" is shown in Figure 1. In this



J.Rosenberg,H.Schulzrinne                                     [Page 1]


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   diagram, a caller, UA1 wishes to call joe@B. To do so, it communi-
   cates 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 the 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 several aspects of the general call flow
   described in the Introduction.

   First off, proxy 1 needs to discover the SIP server in domain B, in
   order to forward the call for joe@B. Specifically, it needs to deter-
   mine the IP address, port and transport for the server in domain B.
   Transport is particularly noteworthy. Unlike other protocols, SIP can
   run over a variety of transports, including TCP, UDP, TLS/TCP and
   SCTP. Therefore, discovery of transports for a particular domain is
   an important part of the processing. The proxy sending the request
   has a particular set of transports it supports (all proxies must
   implement both TCP and UDP) and a preference for using those tran-
   sports. Proxy 2 has its own set of transports it supports (the
   minimal overlap is UDP and TCP in this case), and relative prefer-
   ences for those transports. Some form of DNS procedures are needed
   for proxy 1 to discover the available transports for SIP services at
   domain B, and the relative preferences of those transports. This
   information can be merged with the supported transports and prefer-
   ences at proxy 1, resulting in a selection of a transport.

   It is important to note that DNS processing can be used multiple
   times throughout processing of a call. In general, an element that
   wishes to send a request (generally called a client) may need to per-
   form DNS processing to determine the IP address, port, and transport
   of a next hop element, generally 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 total delay
   between when a user initiates the call, and when they get an indica-
   tion that the called party is being alerted to the call, needs to be



J.Rosenberg,H.Schulzrinne                                     [Page 2]


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
















   Figure 1: The SIP trapezoid


   less than a few seconds. Given that there can be multiple hops, each
   of which is doing DNS processing in addition to other potentially
   time-intensive operations, the amount of time available for DNS pro-
   cessing at each hop is limited.




J.Rosenberg,H.Schulzrinne                                     [Page 3]


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   Scalability and high availability are important in SIP. SIP services
   scale up through clustering techniques. In a more realistic version
   of the network in Figure 1, proxy 2 would typically 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.

   High availability is accomplished in SIP through detection of
   failures by upstream elements. For example, proxy 1 would send a
   request to proxy 2.1 (one of the proxies in the "cluster" proxy 2).
   This request would fail, and that would be detected by proxy 1. Proxy
   1 would then try another of the proxies, proxy 2.2. In many cases,
   such as the one above, 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 com-
   pletes. Proxy 1 could 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, it is not generally possible for an ele-
   ment to perpetually maintain dynamic availability state for the prox-
   ies 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 thats available. The identity of the avail-
   able server would ideally be cached for some amount of time in order
   to reduce call setup delays of subsequent calls. However, the client
   cannot actively "ping" the failed servers to determine when they come
   back alive, because of scalability concerns. Furthermore, the availa-
   bility 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. Ideally, we would like
   proxy 2 to use DNS procedures to identify a backup server for proxy 1
   that it can use to forward the response. This problem is more realis-
   tic in SIP than it is in other transactional protocols. The reason is
   that a SIP response 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 Client Usage

   Usage of DNS differs for clients and for servers. This section
   discusses client usage. The assumption is that the client is stateful



J.Rosenberg,H.Schulzrinne                                     [Page 4]


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   (either a UAC or a stateful proxy). Considerations for stateless
   proxies are discussed in Section 3.4.

   The procedures here are invoked when a client needs to send a request
   to a server for which it does not already know an explicit IP
   address, port, and transport. This occurs when an element wishes to
   send a request to a server identified by a SIP URI, or when an ele-
   ment wishes to send a request to a specific configured server,
   independent of the SIP URI, but the configured server is identified
   by a domain name instead of a numeric IP address.

   The procedures here MUST only be done once per transaction. That is,
   once a server has successfully been contacted (success is defined
   below), all retransmissions of the request and the ACK for non-2xx
   responses MUST be sent to the same server. Furthermore, a CANCEL for
   a particular request MUST be sent to the same server that the request
   was delivered to.

   Note that, because the ACK request for 2xx responses 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 most definitely not get the
   ACK).

   If the request is being delivered to an outbound proxy, a temporary
   URI, used for purposes of this specification, is constructed. That
   URI is of the form sip:<proxy>, where <proxy> is the domain of the
   outbound proxy.

   The first step is to identify the TARGET.  The TARGET is set to the
   value of the maddr parameter of the URI, if present, otherwise, the
   host value of the hostport construction. It represents the domain to
   be contacted.

3.1 Selecting a Transport

   Next, a transport is selected.

   If the URI specifies a transport, that transport MUST be used.

   Otherwise, if no transport is specified, but the TARGET is a numeric
   IP address, the client SHOULD use UDP.

   Otherwise, if no transport is specified, and the target is not a
   numeric IP address, the client SHOULD perform a NAPTR query. This
   query is for the service "SIP+D2T", which provides a mapping from a
   domain to a transport for contacting that domain. The transport is of
   the form of an SRV record, using the "S" NAPTR flag. The resource



J.Rosenberg,H.Schulzrinne                                     [Page 5]


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   record will contain a replacement value (not a regular expression),
   which is the SRV record for a particular transport. If the server
   supports multiple transports, there will be multiple NAPTR records,
   each with a different order value. The client MUST discard any
   records that contain an SRV value with a transport not supported by
   the client, but otherwise follow the processing rules of [3]. The
   result is that the most preferred transport of the server that is
   supported by the client will get used.

   As an example, consider foo.com. A client wishes to contact a SIP
   server in foo.com. It performs a NAPTR query for that domain, and the
   following records are returned:


   ;;       order pref flags service           regexp  replacement
    IN NAPTR 90   50  "s"  "SIP+D2T"           ""  _sip._tcp.foo.com
    IN NAPTR 100  50  "s"  "SIP+D2T"           ""  _sip._udp.foo.com
    IN NAPTR 110  50  "s"  "SIP+D2T"           ""  _sip._tls.foo.com



   This indicates that the server supports TCP, UDP, and TLS, in that
   order of preference. If the client supports UDP and TLS, UDP will be
   used, based on an SRV lookup of _sip._udp.foo.com.


        Somehow this doesn't seem right, since the client needs to
        look at the replacement values to discard entries. Perhaps
        the query should instead be done for sip.<domain>, and the
        service field is "TCP+D2T" or "UDP+D2T"?

   It is STRONGLY RECOMMENDED that the domain suffixes in the replace-
   ment field (i.e., foo.com above) match the domain of the original
   query. Without that, backwards compatibility between RFC 2543 and
   this specification will not be possible.


        THis is because RFC 2543 clients will go directly to SRV
        records using the domain suffixes. If these are non-
        existent, because the NAPTR replacement used a different
        suffix, communication will not take place.

   In the event that no NAPTR records are found, the client constructs
   SRV records for those transports it supports, and does a query for
   each. Queries are done using the service identifier "_sip". If the
   query is successful, it means that the particular transport is sup-
   ported. The client MAY use any transport it desires which is sup-
   ported by the server.



J.Rosenberg,H.Schulzrinne                                     [Page 6]


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        This is a change from RFC 2543, which used to merge the
        priority values across different SRV records.

3.2 Determining port and IP

   Once the transport has been determined, the next step is to determine
   the IP address and port.

   If TARGET is a numeric IP address, use that address. If the URI also
   contains a port, use that port. If no port is specified, use the
   default port for the particular transport.

   If the TARGET was not a numeric IP address, but a port is present in
   the URI, first check the cache to determine if a server has been pre-
   viously contacted successfully for that TARGET and port. If one has
   been, use that server. Otherwise, perform an A or AAAA record lookup
   of the domain name. The result will be a list of IP address, each of
   which can be contacted at the specific port from the URI and tran-
   sport determined previously. Processing then proceeds as described in
   Section 3.3.


        There is a weird case where, where the URI had a domain
        name and a port. SRV records will potentially be used to
        determine the transport, based on the algorithms above, but
        A records used for the actual lookup. That seems odd.

   If the TARGET was not a numeric IP address, and no port was present
   in the URI, first check the cache to see if a server had been previ-
   ously contacted successfully for that TARGET. If one had been, use
   that. Otherwise, perform an SRV query using the service identifier
   "_sip" and the transport as determined from Section 3.1, as specified
   in RFC 2782 [2]. The procedures of RFC 2782, as described in the Sec-
   tion titled "Usage rules" are followed, augmented by the additional
   procedures of Section 3.3.


        This is a change. Previously, if the port was explicit, but
        with a value of 5060, SRV records were used. Now, A records
        will be used. A result of this is that the URL comparison
        rules need to change to reflect that sip:user@foo and
        sip:user@foo:5060 are NOT equivalent any longer. I think
        this should not cause any serious interoperability issues,
        but further consideration is needed.

3.3 Details of 2782 process

   RFC 2782 spells out the details of how a set of SRV records are



J.Rosenberg,H.Schulzrinne                                     [Page 7]


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   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,
   in the case of SIP, are described here.

   For SIP requests, failure occurs if the transaction layer reports a
   503 error response or a transport failure of some sort (generally,
   due to ICMP errors or TCP connection failures). 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 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 rfc2782.

   A server has been contacted "successfully" if a request sent to that
   server generates any kind of response, provisional or final. A map-
   ping of the tuple (TARGET, input TRANSPORT, input PORT) to a specific
   server (IP address, transport, port) that was contacted successfully
   SHOULD be cached for a duration equal to the TTL of the A record for
   that server itself. Note, in the above tuple, input TRANSPORT and
   input PORT refer to the transport and port values from the URI
   itself, if present.

   If a client attempts to contact the server listed in the cache, but
   the request fails, the server MUST be removed from the cache, and the
   entire DNS processing must restart by following the procedures in
   Section 3.1 again.

3.4 Consideration for Stateless Proxies

   The process of the previous sections is highly stateful. When a
   server is contacted successfully, all requests for the transaction
   (plus a CANCEL for that transaction) MUST go to the same server. The
   identity of the successfully contacted server is a form of transac-
   tion state. This presents a challenge for stateless proxies, which
   still need to meet the requiretment for sending all requests in the
   transaction to the same server.

   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. Unfortunately, the procedures of RFC
   2782 and RFC 2915 are not guaranteed to be deterministic. This is
   because records that contain the same priority and weight (in the
   case of SRV) or order and preference (in the case of NAPTR) have no
   specified order. The stateless proxy MUST define a deterministic



J.Rosenberg,H.Schulzrinne                                     [Page 8]


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   order to the records in that case, using any algorithm at its dispo-
   sal. One suggestion is to alphabetize them, for example. To make life
   easier for stateless proxies, it is RECOMMENDED that domain adminis-
   trators 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 simi-
   larly for NAPTR records. If the first server is contacted success-
   fully, things are fine. However, if the first server is not contacted
   successfully, and a subsequent server is, the proxy cannot remain
   stateless for this transaction. This is because a retransmission
   could very well go to a different server if the failed one recovers
   between retransmissions. As such, whenever a proxy does not success-
   fully contact the first server, it SHOULD act as a stateful proxy.

4 Server Usage

   RFC 2543bis defines procedures for sending responses from a server
   back to the client. Typically, for unicast requests, the response is
   sent back to the source IP address where the request came from, using
   the port contained in the Via header. However, it is important to
   provide failover support when the client element fails between send-
   ing the request and receiving the response.

   The procedures here are invoked when a server sends a response to the
   client and that response fails. "Fails" is defined here as any
   response which causes an ICMP error message to be returned, or when
   the transport connection the request came in on closes before the
   response can be sent.

   In these cases, the server examines the value of the sent-by con-
   struction 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 from the Via header, and the port from sent-by,
   if present, else the default for that transport.

   If, however, the sent-by field contained a domain name and a port
   number, the server queries for A 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 from the
   Via. As in the client processing, the next entry in the list is tred
   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 using the service identifier
   "_sip" and the transport from the topmost Via header. 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.



J.Rosenberg,H.Schulzrinne                                     [Page 9]


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5 Security Considerations

   The authors do not believe that this specification introduces any
   additional security issues beyond those already described in RFC 2782
   and RFC 2915.

6 Registration of NATPR D2T Resolution Service


   Name: Domain Name to Transport
      * Mnemonic: D2T
      * Number of Operands: 1
      * Type of Each Operand: Each operand is a domain
      * Format of Each Operand: Each operand is a domain name in standard
        format
      * Algorithm: Opaque
      * Output: One or more SRV record keys
      * Error Conditions:
         o No overlap in transport between client and server

      * Security Considerations:



7 Author's Addresses


   Jonathan Rosenberg
   dynamicsoft
   72 Eagle Rock Avenue
   First Floor
   East Hanover, NJ 07936
   email: jdrosen@dynamicsoft.com




8 Bibliography

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

   [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)



J.Rosenberg,H.Schulzrinne                                    [Page 10]


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   DNS resource record," Request for Comments 2915, Internet Engineering
   Task Force, Sept. 2000.

















































J.Rosenberg,H.Schulzrinne                                    [Page 11]


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