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Updated by: 7984 PROPOSED STANDARD
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
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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.
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............................ ..............................
. . . .
. +-------+ . . +-------+ .
. | | . . | | .
. | 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
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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.
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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).
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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.
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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.
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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
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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
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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
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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.
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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.
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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.
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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)
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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.
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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
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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
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included on all such copies and derivative works. However, this
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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
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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.
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