分享
 
 
 

RFC3263 - Session Initiation Protocol (SIP): Locating SIP Servers

王朝other·作者佚名  2008-05-31
窄屏简体版  字體: |||超大  

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 RFC2782 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

13 Authors' Addresses .................................. 16

14 Full Copyright Statement ............................ 17

1 Introduction

The Session Initiation Protocol (SIP) (RFC3261 [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.

............................ ..............................

. . . .

. +-------+ . . +-------+ .

. . . .

. 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

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.

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 RFC2119 [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. RFC3261 [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).

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 RFC2543 [6], and thus

the only one guaranteed to be interoperable for a SIP URI. It was

also specified as the default transport in RFC2543 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 RFC2915 [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 RFC2915 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 RFC2543, 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.

RFC2543 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 RFC2543. 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

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 RFC2543. 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 RFC2782,

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 RFC2782 Process

RFC2782 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 RFC3261 [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 RFC2782.

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 RFC2782

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

RFC3261 [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 RFC3261 [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 RFC3261). "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.

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 RFC3261 [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 RFC3261 [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 RFC3261

[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.

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 RFCMUST 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 have been placed into the registry:

Services Field Protocol

SIP+D2T TCP

SIPS+D2T TCP

SIP+D2U UDP

SIP+D2S SCTP (RFC2960)

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", RFC3261, June 2002.

[2] Gulbrandsen, A., Vixie, P. and L. Esibov, "A DNS RR for

Specifying the Location of Services (DNS SRV)", RFC2782,

February 2000.

[3] Mealling, M. and R. Daniel, "The Naming Authority Pointer

(NAPTR) DNS Resource Record", RFC2915, September 2000.

[4] Bradner, S., "Key Words for Use in RFCs to Indicate Requirement

Levels", BCP 14, RFC2119, March 1997.

[5] Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA

Considerations Section in RFCs", BCP 26, RFC2434, October

1998.

12 Informative References

[6] Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,

"SIP: Session Initiation Protocol", RFC2543, 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.

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

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

or assist in its implementation may be prepared, copied, published

and distributed, in whole or in part, without restriction of any

kind, provided that the above copyright notice and this paragraph are

included on all such copies and derivative works. However, this

document itself may not be modified in any way, such as by removing

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

copyrights defined in the Internet Standards process must be

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 RFCEditor function is currently provided by the

Internet Society.

 
 
 
免责声明:本文为网络用户发布,其观点仅代表作者个人观点,与本站无关,本站仅提供信息存储服务。文中陈述内容未经本站证实,其真实性、完整性、及时性本站不作任何保证或承诺,请读者仅作参考,并请自行核实相关内容。
2023年上半年GDP全球前十五强
 百态   2023-10-24
美众议院议长启动对拜登的弹劾调查
 百态   2023-09-13
上海、济南、武汉等多地出现不明坠落物
 探索   2023-09-06
印度或要将国名改为“巴拉特”
 百态   2023-09-06
男子为女友送行,买票不登机被捕
 百态   2023-08-20
手机地震预警功能怎么开?
 干货   2023-08-06
女子4年卖2套房花700多万做美容:不但没变美脸,面部还出现变形
 百态   2023-08-04
住户一楼被水淹 还冲来8头猪
 百态   2023-07-31
女子体内爬出大量瓜子状活虫
 百态   2023-07-25
地球连续35年收到神秘规律性信号,网友:不要回答!
 探索   2023-07-21
全球镓价格本周大涨27%
 探索   2023-07-09
钱都流向了那些不缺钱的人,苦都留给了能吃苦的人
 探索   2023-07-02
倩女手游刀客魅者强控制(强混乱强眩晕强睡眠)和对应控制抗性的关系
 百态   2020-08-20
美国5月9日最新疫情:美国确诊人数突破131万
 百态   2020-05-09
荷兰政府宣布将集体辞职
 干货   2020-04-30
倩女幽魂手游师徒任务情义春秋猜成语答案逍遥观:鹏程万里
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案神机营:射石饮羽
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案昆仑山:拔刀相助
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案天工阁:鬼斧神工
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案丝路古道:单枪匹马
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案镇郊荒野:与虎谋皮
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案镇郊荒野:李代桃僵
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案镇郊荒野:指鹿为马
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案金陵:小鸟依人
 干货   2019-11-12
倩女幽魂手游师徒任务情义春秋猜成语答案金陵:千金买邻
 干货   2019-11-12
 
推荐阅读
 
 
 
>>返回首頁<<
 
靜靜地坐在廢墟上,四周的荒凉一望無際,忽然覺得,淒涼也很美
© 2005- 王朝網路 版權所有