Internet Engineering Task Force SIP WG
Internet Draft J.Rosenberg,H.Schulzrinne
draft-ietf-sip-srv-04.txt dynamicsoft,Columbia U.
January 24, 2002
Expires: July 2002
SIP: Locating SIP Servers
STATUS OF THIS MEMO
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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.
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
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
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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.
............................ ..............................
. . . .
. +-------+ . . +-------+ .
. | | . . | | .
. | Proxy |------------- | Proxy | .
. | 1 | . . | 2 | .
. | | . . | | .
. / +-------+ . . +-------+ \ .
. / . . \ .
. / . . \ .
. / . . \ .
. / . . \ .
. / . . \ .
. / . . \ .
. / . . \ .
. +-------+ . . +-------+ .
. | | . . | | .
. | | . . | | .
. | UA 1 | . . | UA 2 | .
. | | . . | | .
. +-------+ . . +-------+ .
. Domain A . . Domain B .
............................ ..............................
Figure 1: The SIP trapezoid
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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.
Transport Protocol is particularly noteworthy. Unlike many other
protocols, SIP can run over a variety of transport protocols,
including TCP, UDP, TLS/TCP and SCTP. 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, 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.
It is important to note that DNS lookups can be used multiple times
throughout 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.
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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. This request would fail, and that would be
detected by proxy 1. Proxy 1 would then try 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 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 Terminology
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In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALLNOT", "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 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 server identified by a SIP URI, or when an element wishes to
send a request to a specific configured server, independent of the
SIP URI (called an outbound proxy), but the outbound proxy is
identified by a domain name instead of a numeric IP address.
Frequently, this is because the URI is contained in the Request-URI
of a request to be sent. The procedures defined here in no way affect
this URI (i.e., the URI is not rewritten with the result of the DNS
looksup), they only result in an IP address, port and transport
protocol where the request can be sent.
The procedures here MUST be done exactly 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 host. Furthermore, a CANCEL for a
particular request MUST be sent to the same host that the request was
delivered to.
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.
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.
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
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First, the client selects a transport protocol.
If the URI specifies a transport protocol in the transport parameter,
that transport protocol MUST be used.
Otherwise, if no transport protocol is specified, but the TARGET is a
numeric IP address, the client SHOULD use UDP.
Otherwise, if no transport protocol is specified, and the target is
not a numeric IP address, the client SHOULD perform a NAPTR query for
the domain in the SIP URI. The services relevant for the task of
transport protocol selection are those with NAPTR service fields with
values "SIP+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, D2S for SCTP and D2L for TLS over TCP. 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 a replacement
value and an empty regular expression, 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 MUST
discard any records whose services fields indicate transport
protocols not supported by the client. The NAPTR processing in RFC
2915 will result in selection of a transport protocol (and an SRV
record along with it) with most preferred transport protocol of the
server that is supported by the client.
As an example, consider example.com. A client wishes to contact a SIP
server in example.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.school.edu
IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.example.com
IN NAPTR 110 50 "s" "SIP+D2S" "" tls-sip.example.com
This indicates that the server supports TCP, UDP, and TLS, in that
order. If the client supports UDP and TLS, UDP will be used, based on
an SRV lookup of _sip._udp.example.com.
It is not necessary for the domain suffixes in the replacement field
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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.
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". 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, which used to merge the
priority values across different SRV records.
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 address, each of which can be
contacted at the specific port from the URI and transport protocol
determined previously. Processing then proceeds as described in
Section 4.3 of this document.
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 protocol, 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
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in the URI, the client performs an SRV query using the service
identifier "_sip" and the transport protocol as determined from
Section 4.1, as specified in RFC 2782 [2]. 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. 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@example.com
and sip:user@example.com:5060 are NOT equivalent any
longer. I think this should not cause any serious
interoperability issues, but further consideration is
needed.
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.
The client client MAY maintain a table indicating the status of a
particular host (that is, whether it was ever successfully contacted,
or whether attempts to contact it resulted in a failure). The table
is indexed with the IP address, port, and transport for a particular
host. If a particular host is listed with a status of "failed", that
entry SHOULD be discarded after one hour, so that the host can be
used once more if it has recovered.
When processing the list of SRV entries (or A records, depending on
how the URI was resolved), the client MAY remove any entries for
hosts which are marked as "failed" in the table. The remaining
entries are then tried according to RFC 2782.
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 RFC 2782.
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4.4 Consideration for Stateless Proxies
The process of the previous sections is highly stateful. When a
server is contacted successfully, all requests for the transaction,
as well as 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 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
order to the records in that case, using any algorithm at its
disposal. One suggestion is to alphabetize them, for example. 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 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
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sending 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
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.
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 protocol
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 protocol 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.
6 Constructing SIP URIs
In many cases, and 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 [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
sip:user@foo.com;maddr=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
sip:user@foo.com;maddr=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, 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 [1] to be met while allowing for robust
operation.
7 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.
8 Registration of NATPR D2X Resolution Service
Name: Domain Name to Transport Protocol
* Mnemonic: D2X, where X is managed by an IANA registration process
* 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
* Input String: The domain name from the SIP URI being used to
generate the NAPTR query.
* Output: One or more SRV record keys
* Constraints: All records MUST only use the S flag. The P flag is
expressly forbidden.
* Error Conditions:
o No overlap in transport protocol between client and server
* Security Considerations: none
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 at any time
when new transport protocols become available. Such additions are
subject to expert review.
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The registration 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), RFC5766".
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
SIP+D2U UDP
SIP+D2L TLS over TCP (RFC 2246)
SIP+D2S SCTP (RFC 2960)
10 Changes Since -03
o Added IANA registration process.
o Included text discussing the problem of DNS TTL expiration for
stateless proxies.
o Clarified that maintenance of the table of availability for
servers is not a cache, and it is totally unrelated to DNS
processing.
o Changed the construction of the services field in NAPTR to
include the transport protocol, so its SIP+D2X, where X
depends on the transport protocol.
o Relaxed the constraint that the domain suffix in the NAPTR
records equal that of the target.
o Added a section on how to construct URIs for insertion into
Contact and Record-Route headers.
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11 Acknowledgements
The authors would like to thank Patrik Faltstrom for his useful
comments.
12 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
13 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)
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.
Full Copyright Statement
Copyright (c) The Internet Society (2002). All Rights Reserved.
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