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
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all provisions of Section 10 of RFC2026.
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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
J.Rosenberg et. al. [Page 2]
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.
J.Rosenberg et. al. [Page 16]
Internet Draft sip-srv February 21, 2002
[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
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Internet Draft sip-srv February 21, 2002
The limited permissions granted above are perpetual and will not be
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J.Rosenberg et. al. [Page 18]
