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Internet Engineering Task Force MMUSIC WG
Internet Draft Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-03.txt ISI/Columbia U./Caltech
July 31, 1997
Expires: January 20, 1998
SIP: Session Initiation Protocol
STATUS OF THIS MEMO
This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as ``work in progress''.
To learn the current status of any Internet-Draft, please check the
``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or
ftp.isi.edu (US West Coast).
Distribution of this document is unlimited.
ABSTRACT
Many styles of multimedia conferencing are likely to co-
exist on the Internet, and many of them share the need to
invite users to participate. The Session Initiation
Protocol (SIP) is a simple protocol designed to enable
the invitation of users to participate in such multimedia
sessions. It is not tied to any specific conference
control scheme. In particular, it aims to enable user
mobility by relaying and redirecting invitations to a
user's current location.
This document is a product of the Multi-party Multimedia
Session Control (MMUSIC) working group of the Internet
Engineering Task Force. Comments are solicited and
should be addressed to the working group's mailing list
at confctrl@isi.edu and/or the authors.
Handley/Schulzrinne/Schooler [Page 1]
Internet Draft SIP July 31, 1997
1 Introduction
1.1 Overview of SIP Functionality
The Session Initiation Protocol (SIP) is an application-layer
protocol that can establish and control multimedia sessions or calls.
These multimedia sessions include multimedia conferences, distance
learning, Internet telephony and similar applications. SIP can
initiate both unicast and multicast sessions; the initiator does not
necessarily have to be a member of the session. Media and
participants can be added to an existing session. SIP can be used to
"call" both persons and "robots", for example, to invite a media
storage device to record an ongoing conference or to invite a video-
on-demand server to play a video into a conference. (SIP does not
directly control these services, however; see RTSP [4].)
SIP transparently supports name mapping and redirection services,
allowing the implementation of telephony services such as selective
call forwarding, selective call rejection, conditional and
unconditional call forwarding, call forwarding busy, call forwarding
no response. SIP may use multicast to try several possible callee
locations at the same time.
SIP supports personal mobility telecommunications intelligent network
services, this is defined as: "Personal mobility is the ability of
end users to originate and receive calls and access subscribed
telecommunication services on any terminal in any location, and the
ability of the network to identify end users as they move. Personal
mobility is based on the use of a unique personal identity (i.e.,
'personal number')." [1]. Personal mobility complements terminal
mobility, i.e., the ability to maintain communications when moving a
single end system from one network to another.
SIP supports some or all of four facets of establishing multimedia
communications:
1. user location: determination of the end system to be used
for communication;
2. user capabilities: determination of the media and media
parameters to be used;
3. user availability: determination of the willingness of the
called party to engage in communications;
4. call setup ("ringing", establishment of call parameters at
both called and calling party)
In particular, SIP can be used to locate a user and determine
Handley/Schulzrinne/Schooler [Page 2]
Internet Draft SIP July 31, 1997
the appropriate end system, leaving the actual call
establishment to other protocols such as H.323.
SIP may also be used to terminate and transfer a call. SIP can also
initiate multi-party calls using a multipoint control unit (MCU) or
fully-meshed interconnection instead of multicast.
These features are for further study.
SIP is not a conference control protocol, but can be used to
introduce conference control protocols to a session.
SIP is designed as part of the overall IETF multimedia data and
control architecture currently incorporating protocols such as RSVP
[2] for reserving network resources, RTP [3] for transporting real-
time data and providing QOS feedback, RTSP [4] for controlling
delivery of streaming media, SAP [5] for advertising multimedia
sessions via multicast and SDP [6] for describing multimedia
sessions, but SIP does not depend for its operation on any of these
protocols.
1.2 Finding Multimedia Sessions
There are two basic ways to locate and to participate in a multimedia
session:
Advertisement: The session is advertised, potential participants see
the advertisement, then join the session address to participate.
Invitation: Users are invited by others to participate in a session,
which may or may not be advertised.
Sessions may be advertised using multicast protocols such as SAP [5],
electronic mail, news groups, web pages or directories (LDAP), among
others. SIP serves the role of the invitation protocol.
SIP does not prescribe how a conference is to be managed, e.g.,
whether it uses a central server to manage conference and participant
state or distributes state via multicast.
SIP does not allocate multicast addresses, leaving this functionality
to protocols such as SAP [5].
SIP can invite users to conferences with and without resource
reservation. SIP does not reserve resources, but may convey to the
invited system the information necessary to do this. Quality-of-
service guarantees, if required, may depend on knowing the full
membership of the session; this information may or may not be known
Handley/Schulzrinne/Schooler [Page 3]
Internet Draft SIP July 31, 1997
to the agent performing session invitation.
SIP offers some of the same functionality as H.323, but may also be
used in conjunction with it. In this mode, H.323 is used to locate
the appropriate terminal identified by a H.245 address [TBD: what
does this look like?]. An H.323-capable terminal then proceeds with a
normal H.323/H.245 invitation [7].
1.3 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 [8] and
indicate requirement levels for compliant SIP implementations.
1.4 Definitions
This specification uses a number of terms to refer to the roles
played by participants in SIP communications. The definitions of
client, server and proxy are similar to those used by the Hypertext
Transport Protocol (HTTP) [9].
Client: An application program that establishes connections for the
purpose of sending requests. Clients may or may not interact
directly with a human user.
Final response: A response that terminates a -> SIP transaction, as
opposed to a -> provisional response 3xx, 4xx, and 5xx
responses are final.
Initiator, calling party: The party initiating a conference
invitation. Note that the calling party does not have to be the
same as the one creating a conference.
Invitation: A request sent to a user (or service) requesting
participation in a session.
Invitee, invited user, called party: The person or service that the
calling party is trying to invite to a conference.
Location server: A program that is contacted by a -> client and
that returns one or more possible locations of the called party
or service. Location servers may be invoked by SIP redirect and
proxy servers and may be Co-located with a SIP server.
Location service: A service used by a -> redirect or -> proxy
server to obtain information about a callee's possible location.
Handley/Schulzrinne/Schooler [Page 4]
Internet Draft SIP July 31, 1997
Provisional response: A response used by the server to indicate
progress, but that does not terminate a -> SIP transaction.
All 1xx and 6xx responses are provisional. Other responses are
considered -> final.
Proxy, proxy server: An intermediary program that acts as both a
server and a client for the purpose of making requests on behalf
of other clients. Requests are serviced internally or by passing
them on, possibly after translation, to other servers. A proxy
must interpret, and, if necessary, rewrite a request message
before forwarding it.
Redirect server: A server that accepts a SIP request, maps the
address into zero or more new addresses and returns these
addresses to the client. Unlike a -> proxy server, it does not
initiate its own SIP request.
Server: An application program that accepts requests in order to
service requests and sends back responses to those requests.
Servers are either proxy, redirect or user agent servers. An
application program may act as both server and client.
Session: "A multimedia session is a set of multimedia senders and
receivers and the data streams flowing from senders to
receivers. A multimedia conference is an example of a multimedia
session." [6] For SIP, a session is equivalent to a "call".
(Note: a session as defined here may comprise one or more RTP
sessions.)
(SIP) transaction: A SIP transaction occurs between a -> client and
a -> server and comprises all messages from the first request
sent from the client to the server up to a -> final (non-1xx)
response sent from the server to the client. A transaction is
for a single call (identified by a Call-ID, Section 6.11).
There can only be one pending transaction between a server and
client for each call id.
User agent server, called user agent: The server application that
contacts the user when a session request is received and that
returns a reply on behalf of the user. The reply may accept,
reject or redirect the call. (Note: in SIP, user agents can be
both clients and servers.)
An application program may be capable of acting both as a client and
a server. For example, a typical multimedia conference control
application would act as a client to initiate calls or to invite
others to conferences and as a user agent server to accept
invitations.
Handley/Schulzrinne/Schooler [Page 5]
Internet Draft SIP July 31, 1997
1.5 Protocol Properties
1.5.1 Minimal State
There is no concept of an ongoing SIP session that lasts for the
duration of the conference or call. Rather, a single conference
session or call may involve one or more SIP request-response
transactions. For example, a conference control protocol may use SIP
to add or remove a media stream, but again, once the information has
been successfully conveyed to the participants, SIP is then no longer
involved.
At most, a server has to maintain state for a single SIP transaction.
In some cases, it can process each message without regard to previous
messages ( stateless server ), as described in Section 12.
1.5.2 Transport-Protocol Neutral
SIP is able to utilize both UDP and TCP as transport protocols. UDP
allows the application to more carefully control the timing of
messages and their retransmission, to perform parallel searches
without requiring connection state for each outstanding request, and
to use multicast. TCP allows easier passage through existing
firewalls, and given the similar protocol design, allows common
servers for SIP, HTTP and the Real Time Streaming Protocol (RTSP)
[4].
When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing session. The
concept of a session is not implicitly bound to a TCP connection, so
the initial SIP request and a subsequent SIP request may use
different TCP connections or a single persistent connection as
appropriate.
Clients SHOULD implement both UDP and TCP transport, servers MUST.
1.5.3 Text-Based
SIP is text based. This allows easy implementation in languages such
as Tcl and Perl, allows easy debugging, and most importantly, makes
SIP flexible and extensible. As SIP is primarily used for session
initiation, it is believed that the additional overhead of using a
text-based protocol is not significant.
1.6 SIP Addressing
SIP uses two kinds of address identifiers, host-specific addresses
and host-independent addresses form user@host , where user is any
Handley/Schulzrinne/Schooler [Page 6]
Internet Draft SIP July 31, 1997
alphanumeric identifier and the form of host depends on the address
type. Note that SIP does not distinguish between the two and can,
while inviting a user, map repeatedly between the two address types.
For a host-specific address, the user part is an operating-system
user name. The host part is either a domain name having a DNS A
(address) record, or a numeric network address. Examples include:
mjh@metro.isi.edu
hgs@erlang.cs.columbia.edu
root@193.175.132.42
A user's host-specific address can be obtained out-of-band, can be
learned via existing media agents, can be included in some mailers'
message headers, or can be recorded during previous invitation
interactions.
Host-independent addresses contain a moniker (such as a civil name)
or user name and domain name that may not map into a single host.
[1]
Host-independent addresses may use any unambiguous user name,
including aliases, identifying the called party as the user part of
the address. They may use a domain name having an MX [10], SRV [11]
or A [12] record for the host part. These addresses may have
different degrees of location- and provider-independence and are
often chosen to be mnemonic. In many cases, the host-independent SIP
address can be the same as a user's electronic mail address, but this
is not required. SIP can thus leverage off the domain name system
(DNS) to provide a first-stage location mechanisms. Examples of
host-independent names include
M.Handley@cs.ucl.ac.uk
H.G.Schulzrinne@ieee.org
info@ietf.org
An address can designate an individual (possibly located at one of
several end systems), the first available person from a group of
individuals or a whole group. The form of the address, e.g.,
_________________________
[1] We avoid the term location-independent , since
the address may indeed refer to a specific location,
e.g., a company department.
Handley/Schulzrinne/Schooler [Page 7]
Internet Draft SIP July 31, 1997
sales@example.com , is not sufficient, in general, to determine the
intent of the caller.
If a user or service chooses to be reachable at an address that is
guessable from the person's name and organizational affiliation, the
traditional method of ensuring privacy by having an unlisted "phone"
number is compromised. However, unlike traditional telephony, SIP
offers authentication and access control mechanisms and can avail
itself of lower-layer security mechanisms, so that client software
can reject unauthorized or undesired call attempts.
1.7 Locating a SIP Server
Call setup may proceed in several phases. A SIP client MUST follow
the following steps to resolve the user part of a callee address. If
a client only supports TCP or UDP, but not both, the respective
address type is omitted.
1. If there is a SRV DNS resource record [11] of type sip.udp
, contact the listed SIP servers in order of preference
value using UDP as a transport protocol at the port number
listed in the DNS resource record.
2. If there is a SRV DNS resource record [11] of type sip.tcp
, contact the listed SIP servers in order of preference
value using TCP as a transport protocol at the port number
listed in the DNS resource record.
3. If there is a DNS MX record [10], contact the hosts listed
in their order of preference at the default port number
(TBD). For each host listed, first try to contact the
server using UDP, then TCP.
4. Finally, check if there is a DNS CNAME or A record for the
given host and try to contact a SIP server at the one or
more addresses listed, again trying first UDP, then TCP.
5. If all of the above methods fail, the caller MAY contact an
SMTP server at the user's host and use the SMTP EXPN
command to obtain an alternate address and repeat the steps
above. As a last resort, a client MAY choose to deliver the
session description to the callee using electronic mail.
If a server is found using one of the methods below, the other
methods are not tried. A client SHOULD rely on ICMP "Port
Unreachable" messages rather than time-outs to determine that a
server is not reachable at a particular address. A client MAY cache
the result of the reachability steps, but SHOULD start at the
Handley/Schulzrinne/Schooler [Page 8]
Internet Draft SIP July 31, 1997
beginning of the sequence when the cached address fails.
Implementation note for socket-based programs: For TCP, connect()
returns ECONNREFUSED if there is no server at the designated address;
for UDP, the socket should be bound to the destination address using
connect() rather than sendto() or similar.
This sequence is modeled after that described for SMTP,
where MX records are to be checked before A records [13].
1.8 SIP Transactions
Once the host part has been resolved to a SIP server, the client
sends one or more SIP requests to that server and receives one or
more responses from the server. If the invitation is SIP request is
an invitation, it contains a session description, for example written
in SDP format, that provides the called party with enough information
to join the session.
If TCP is used, request and responses within a single SIP transaction
are carried over the same TCP connection. Thus, the client SHOULD
maintain the connection until a final response has been received.
Several SIP requests from the same client to the same server may use
the same TCP connection or may open a new connection for each
request. If the client sent the request sends via unicast UDP, the
response is sent to the source address of the UDP request. If the
request is sent via multicast UDP, the response is directed to the
same multicast address and destination port. For UDP, reliability is
achieved using retransmission (Section 11).
Need motivation why we ALWAYS want to have a multicast
return.
The SIP message format and operation is independent of the transport
protocol.
The basic message flow is shown in Fig. 1 and Fig. 2, for proxy and
redirect modes, respectively.
1.9 Locating a User
A callee may move between a number of different end systems over
time. These locations can be dynamically registered with a location
server, typically for a single administrative domain, or a location
Handley/Schulzrinne/Schooler [Page 9]
Internet Draft SIP July 31, 1997
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@play :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | | 4: INVITE 5: ring :
: cz@cs.tu-berlin.de ========================> tune =========> play :
: <........................ <......... :
: : 7: 200 OK : 6: 200 OK :
+.....................+ +...............................+
====> SIP request
----> non-SIP protocols
Figure 1: Example of SIP proxy server
server may use other protocols, such as finger [14], rwho,
multicast-based protocols or operating-system dependent mechanism to
actively determine the end system where a user is reachable. The
location services yield a list of a zero or more possible locations,
possibly even sorted in order of likelihood of success.
The location server can be part of the SIP server or the SIP server
may use a different protocol (e.g., finger [14] or LDAP [15]) to map
addresses. A single user may be registered at different locations,
either because she is logged in at several hosts simultaneously or
because the location server has (temporarily) inaccurate information.
The action taken on receiving a list of locations varies with the
type of SIP server. A SIP redirect server simply returns the list to
the client sending the request as Location headers (Section 6.17). A
SIP proxy server can sequentially try the addresses until the call is
successful (2xx response) or the callee has declined the call (40x
response). Alternatively, the server may issue several requests in
parallel. A proxy server can only issue more than one sequential or
parallel connection request if it is the first in the chain of hosts
Handley/Schulzrinne/Schooler [Page 10]
Internet Draft SIP July 31, 1997
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@play :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | | :
: cz@cs.tu-berlin.de =======================> tune :
: ^ | <....................... :
: . | : 4: 302 Moved : :
+...........|.........+ hgs@play : :
. | : :
. | 5: INVITE hgs@play.cs.columbia.edu 6: ring :
. ==================================================> play :
..................................................... :
7: 200 OK : :
+...............................+
====> SIP request
----> non-SIP protocols
Figure 2: Example of SIP redirect server
noted in the Via header to do so. If it is not the first, it must
issue a redirect response.
If a proxy server forwards a SIP request, it SHOULD add itself to the
end of the list of forwarders noted in the Via (Section 6.31)
headers. A proxy server also notes whether it is attempting to reach
several possible locations at once ("connection forking"). The Via
trace ensures that replies can take the same path back, thus ensuring
correct operation through compliant firewalls and loop-free requests.
On the reply path, each host most remove its Via, so that routing
internal information is hidden from the callee and outside networks.
When a multicast request is made, first the host making the request,
then the multicast address itself are added to the path.
Handley/Schulzrinne/Schooler [Page 11]
Internet Draft SIP July 31, 1997
As discussed in Section 1.6, a SIP address may designate a group
rather than an individual. A client indicates using the Reach
request header whether it wants to reach the first available
individual or treat the address as a group, to be invited as a whole.
The default is to attempt to reach the first available callee. If
the address is designated as a group address, a proxy server MUST
return the list of individuals instead of attempting to connect to
these.
Otherwise, the proxy cannot report errors and call status
appropriately.
2 SIP Uniform Resource Locators
SIP URLs are used within SIP messages to indicate the originator and
recipient of a SIP request, and to specify redirection addresses. A
SIP URL may be embedded in web pages or other hyperlinks to indicate
that a user or service may be called. Within SIP messages, an email
address could have been used, but this would have made it more
difficult to gateway between SIP and other protocols with other
addressing schemes.
For greater functionality, because interaction with some resources
may require message headers or message bodies to be specified as well
as the SIP address, the sip URL scheme is extended to allow setting
SIP request-header fields and the SIP message-body.
A SIP URL follows the guidelines of RFC 1630 [16,17] and takes the
following form:
SIP-URL = short-sip-url | full-sip-url
full-sip-url = "sip://" user [ ":" password ] "@" host
url-parameters [ headers ]
short-sip-url = user [ ":" password ] "@" host : port
user = ; defined in RFC 1738 [18]
host = ; defined in RFC 1738
port = *digit
url-parameters = *( ";" url-parameter)
url-parameter = transport-param |
ttl-param | maddr-param
transport-param = "transport=" ( "udp" | "tcp" )
ttl-param = "ttl=" ttl
ttl = 1*3DIGIT ; 0 to 255
maddr-param = "maddr=" maddr
maddr = ; dotted decimal multicast address
headers = "?" header *( " " header )
Handley/Schulzrinne/Schooler [Page 12]
Internet Draft SIP July 31, 1997
header = hname "=" hvalue
hname = *urlc
hvalue = *urlc
urlc = ; defined in [17]
Thus a SIP URL can take either a short form or a full form. The short
form MAY only be used within SIP messages where the scheme (SIP) can
be assumed. In all other cases, and when parameters are required to
be specified, the full form MUST be used.
Note that all URL reserved characters must be encoded. The special
hname "body" indicates that the associated hvalue is the message-
body of the SIP INVITE request. Within sip URLs, the characters
"?", "=", "&" are reserved.
Examples of short and full form SIP URLs with identical address are:
j.doe@big.com
sip://j.doe@big.com
sip://j.doe:secret@big.com;transport=tcp
sip://j.doe@big.com?subject=project
The password parameter allows to easily specify a call-back address
on a secure web page, but carries the same security risks as all
URL-based passwords and should only be used under special
circumstances where security requirements are low or all transport
paths are secured.
Within a SIP message, URLs are used to indicate the source and
intended destination of a request, redirection addresses and the
current destination of a request. Normally all these fields will
contain SIP URLs. When additional parameters are not required, the
short form SIP URL can be used unambiguously.
In some circumstances a non-SIP URL may be used in a SIP message. An
example might be making a call from a telephone which is relayed by a
gateway onto the internet as a SIP request. In such a case, the
source of the call is really the telephone number of the caller, and
so a SIP URL is inappropriate and a phone URL might be used instead.
Thus where SIP specifies user addresses it allows these addresses to
be URLs.
Clearly not all URLs are appropriate to be used in a SIP message as a
user address. It is unlikely, for example, that HTTP or FTP URLs are
useful in this context. The correct behavior when an unknown scheme
Handley/Schulzrinne/Schooler [Page 13]
Internet Draft SIP July 31, 1997
is encountered by a SIP server is defined in the context of each of
the header fields that use a SIP URL.
SIP URLs can define specific parameters of the request, including the
transport mechanism (UDP or TCP) and the use of multicast to make a
request. These parameters are added after the host and are separated
by semi-colons. For example, to specify to call j.doe@big.com using
multicast to 239.255.255.1 with a ttl of 15, the following URL would
be used:
sip://j.doe@big.com;maddr=239.255.255.1;ttl=15
The transport protocol UDP is to be assumed when a multicast address
is given.
3 SIP Message Overview
Since much of the message syntax is identical to HTTP/1.1, rather
than repeating it here we use [HX.Y] to refer to Section X.Y of the
current HTTP/1.1 specification [9]. In addition, we describe SIP in
both prose and an augmented Backus-Naur form (BNF) [H2.1] described
in detail in [19].
All SIP messages are text-based and use HTTP/1.1 conventions [H4.1],
except for the additional ability of SIP to use UDP. When sent over
TCP or UDP, multiple SIP transactions can be carried in a single TCP
connection or UDP datagram. UDP datagrams, including all headers,
should not normally be larger than the path maximum transmission unit
(MTU) if the MTU is known, or 1500 bytes if the MTU is unknown.
The 1400 bytes accommodates lower-layer packet headers
within the "typical" MTU of around 1500 bytes. There are
few MTU values around 1 kB; the next value is 1006 bytes
for SLIP and 296 for low-delay PPP [20]. Recent studies
[21] indicate that an MTU of 1500 bytes is a reasonable
assumption. Thus, another reasonable value would be a
message size of 950 bytes, to accommodate packet headers
within the SLIP MTU without fragmentation.
A SIP message is either a request from a client to a server, or a
response from a server to a client.
SIP-message = Request | Response ; SIP messages
Handley/Schulzrinne/Schooler [Page 14]
Internet Draft SIP July 31, 1997
Both Request (section 4) and Response (section 5) messages use the
generic message format of RFC 822 [22] for transferring entities (the
payload of the message). Both types of message consist of a start-
line, one or more header fields (also known as "headers"), an empty
line (i.e., a line with nothing preceding the carriage-return line-
feed ( CRLF)) indicating the end of the header fields, and an
optional message-body. To avoid confusion with similar-named headers
in HTTP, we refer to the header describing the message body as entity
headers. These components are described in detail in the upcoming
sections.
generic-message = start-line
*message-header
CRLF
[ message-body ]
start-line = Request-Line | Status-Line
Request = Request-Line ; Section 4.1
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ]
Response = Status-Line ; Section 5.1
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ]
In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins
with a CRLF, the CRLF should be ignored.
4 Request
The Request message format is shown below:
Handley/Schulzrinne/Schooler [Page 15]
Internet Draft SIP July 31, 1997
general-header = Call-ID ; Section 6.11
| Date ; Section 6.14
| Expires ; Section 6.15
| From ; Section 6.16
| Sequence ; Section 6.26
| Via ; Section 6.31
entity-header = Content-Length ; Section 6.12
| Content-Type ; Section 6.13
request-header = Accept ; Section 6.6
| Accept-Language ; Section 6.7
| Authorization ; Section 6.9
| Organization ; Section 6.18
| Priority ; Section 6.20
| Proxy-Authorization ; Section 6.22
| Reach ; Section 6.24
| Subject ; Section 6.28
| To ; Section 6.29
| User-Agent ; Section 6.30
response-header = Location ; Section 6.17
| Proxy-Authenticate ; Section 6.21
| Public ; Section 6.23
| Retry-After ; Section 6.25
| Server ; Section 6.27
| Warning ; Section 6.32
| WWW-Authenticate ; Section 6.33
Table 1: SIP headers
Request = Request-Line ; Section 4.1
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ] ; Section 8
4.1 Request-Line
The Request-Line begins with a method token, followed by the
Request-URI and the protocol version, and ending with CRLF. The
elements are separated by SP characters. No CR or LF are allowed
except in the final CRLF sequence.
Request-Line = Method SP Request-URI SP SIP-Version CRLF
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4.1.1 Methods
The following methods are defined:
method = "INVITE" | "CONNECTED" | "OPTIONS" | "BYE"
| "REGISTER" | "UNREGISTER"
INVITE: The user or service is being invited to participate in the
session. This method MUST be supported by a SIP server.
CONNECTED: A CONNECTED request confirms that the client has received
a successful response to an INVITE request. See Section 11 for
details. This method MUST be supported by a SIP server.
OPTIONS: The client is being queried as to its capabilities. A server
that believes it can contact the user, such as a user agent
where the user is logged in and has been recently active, MAY
respond to this request with a capability set. Support of this
method is OPTIONAL.
BYE: The client indicates to the server that it wishes to abort the
call attempt. The leaving party can use a Location header field
to indicate that the recipient of request should contact the
named address. This implements the "call transfer" telephony
functionality. A client SHOULD also use this method to indicate
to the callee that it wishes to abort an on-going call attempt.
With UDP, the caller has no other way to signal her intent
to drop the call attempt and the callee side will keep
"ringing". When using TCP, a client MAY also close the
connection to abort a call attempt. Support of this method
is OPTIONAL.
REGISTER: A client uses the REGISTER method to register the address
listed in the request line to a SIP server. In the future, the
server MAY use the source address and port to forward SIP
requests to. A server SHOULD silently drop the registration
after one hour, unless refreshed by the client. A server may set
or lower or higher refresh interval and indicate the interval
through the Expires header (Section 6.15). A single address (if
host-independent) may be registered from several different
clients. Support of this method is OPTIONAL.
Beyond its use as a simple location service, this method is
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needed if there are several SIP servers on a single host,
so that some cannot use the default port number. Each such
server would register with a server for the administrative
domain.
UNREGISTER: A client cancels an existing registration established for
the Request-URI with REGISTER with the UNREGISTER method. If
it unregisters a Request-URI unknown to the servers, the server
returns a 200 (OK) response. Support of this method is OPTIONAL.
BYE and REGISTER are experimental and need to be discussed.
Methods that are not supported by a proxy server SHOULD be treated by
that proxy as if they were an INVITE method, and relayed through
unchanged or cause a redirection as appropriate.
Methods that are not supported by a server should cause a "501 Not
Implemented" response to be returned (Section 7).
4.1.2 Request-URI
The Request-URI field is a SIP URL as described in Section 2 or a
general URI. It indicates the user or service that this request is
being addressed to. Unlike the To field, the Request-URI field may
be re-written by proxies. For example, a proxy may perform a lookup
on the contents of the To field to resolve a username from a mail
alias, and then use this username as part of the Request-URI field
of requests it generates.
If a SIP server receives a request contain a URI indicating a scheme
other than SIP which that server does not understand, the server MUST
return a "400 Bad Request" response. It MUST do this even if the To
field contains a scheme it does understand.
4.1.3 SIP Version
Both request and response messages include the version of SIP in use,
and basically follow [H3.1], with HTTP replaced by SIP. To be
conditionally compliant with this specification, applications sending
SIP messages MUST include a SIP-Version of "SIP/2.0".
5 Response
After receiving and interpreting a request message, the recipient
responds with a SIP response message. The response message format is
shown below:
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Response = Status-Line ; Section 5.1
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ] ; Section 8
[H6] applies except that HTTP-Version is replaced by SIP-Version.
Also, SIP defines additional response codes and does not use some
HTTP codes.
5.1 Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version ((Section 4.1.3) followed by a numeric
Status-Code and its associated textual phrase, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
Status-Line = SIP-version SP Status-Code SP Reason-Phrase
CRLF
5.1.1 Status Codes and Reason Phrases
The Status-Code is a 3-digit integer result code that indicates the
outcome of the attempt to understand and satisfy the request. The
Reason-Phrase is intended to give a short textual description of the
Status-Code. The Status-Code is intended for use by automata,
whereas the Reason-Phrase is intended for the human user. The client
is not required to examine or display the Reason-Phrase.
We provide an overview of the Status-Code below, and provide full
definitions in section 7. The first digit of the Status-Code defines
the class of response. The last two digits do not have any
categorization role. SIP/2.0 allows 6 values for the first digit:
1xx: Informational -- request received, continuing process;
2xx: Success -- the action was successfully received, understood, and
accepted;
3xx: Redirection -- further action must be taken in order to complete
the request;
4xx: Client Error -- the request contains bad syntax or cannot be
fulfilled at this server;
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5xx: Server Error -- the server failed to fulfill an apparently valid
request;
6xx: Global Failure - the request is invalid at any server.
Presented below are the individual values of the numeric response
codes, and an example set of corresponding reason phrases for
SIP/2.0. These reason phrases are only recommended; they may be
replaced by local equivalents without affecting the protocol. Note
that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
codes in the range starting at x80 to avoid conflicts with newly
defined HTTP response codes, and extends these response codes in the
6xx range.
SIP response codes are extensible. SIP applications are not required
to understand the meaning of all registered response codes, though
such understanding is obviously desirable. However, applications MUST
understand the class of any response code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 response code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if a client
receives an unrecognized response code of 431, it can safely assume
that there was something wrong with its request and treat the
response as if it had received a 400 response code. In such cases,
user agents SHOULD present to the user the message body returned with
the response, since that message body is likely to include human-
readable information which will explain the unusual status.
6 Header Field Definitions
SIP header fields are similar to HTTP header fields in both syntax
and semantics [H4.2], [H14]. In general the ordering of the header
fields is not of importance (with the exception of Via fields, see
below), but proxies MUST NOT reorder or otherwise modify header
fields other than by adding a new Via field. This allows an
authentication field to be added after the Via fields that will not
be invalidated by proxies.
To, From, and Call-ID header MUST be present in each request with
method INVITE. The Content-Type and Content-Length headers are
required when there is a valid message body (of non-zero length)
associated with the message (Section 8).
A server MUST understand the PEP-Require header.
Other headers may be added as required; a server MAY ignore headers
that it does not understand. A compact form of these header fields is
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Status-Code = "100" ; Trying
| "180" ; Ringing
| "200" ; OK
| "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "305" ; Use Proxy
| "380" ; Alternative Service
| "400" ; Bad Request
| "401" ; Unauthorized
| "402" ; Payment Required
| "403" ; Forbidden
| "404" ; Not Found
| "405" ; Method Not Allowed
| "407" ; Proxy Authentication Required
| "408" ; Request Timeout
| "409" ; Conflict
| "410" ; Gone
| "411" ; Length Required
| "412" ; Precondition Failed
| "413" ; Request Message Body Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "420" ; Bad Extension
| "480" ; Temporarily not available
| "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Timeout
| "505" ; SIP Version not supported
| "600" ; Busy
| "603" ; Decline
| "604" ; Does not exist anywhere
| "606" ; Not Acceptable
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
Figure 3: Status Codes
also defined in Section 10 for use over UDP when the request has to
fit into a single packet and size is an issue.
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6.1 General Header Fields
There are a few header fields that have general applicability for
both request and response messages. These header fields apply only to
the message being transmitted.
General-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields may be given the semantics of general
header fields if all parties in the communication recognize them to
be general-header fields.
6.2 Entity Header Fields
Entity-header fields define meta-information about the message-body
or, if no body is present, about the resource identified by the
request. The term "entity header" is an HTTP 1.1 term where the reply
body may contain a transformed version of the message body. The
original message body is referred to as the "entity". We retain the
same terminology for header fields but usually refer to the "message
body" rather then the entity as the two are the same in SIP.
6.3 Request Header Fields
The request-header fields allow the client to pass additional
information about the request, and about the client itself, to the
server. These fields act as request modifiers, with semantics
equivalent to the parameters on a programming language method
invocation.
Request-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of request-
header fields if all parties in the communication recognize them to
be request-header fields. Unrecognized header fields are treated as
entity-header fields.
6.4 Response Header Fields
The response-header fields allow the server to pass additional
information about the response which cannot be placed in the Status-
Line. These header fields give information about the server and about
further access to the resource identified by the Request-URI.
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to
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be response-header fields. Unrecognized header fields are treated as
entity-header fields.
6.5 Header Field Format
Header fields ( general-header, request-header, response-header, and
entity-header) follow the same generic header format as that given in
Section 3.1 of RFC 822 [22].
Each header field consists of a name followed by a colon (":") and
the field value. Field names are case-insensitive. The field value
may be preceded by any amount of leading white space (LWS), though a
single space (SP) is preferred. Header fields can be extended over
multiple lines by preceding each extra line with at least one SP or
horizontal tab (HT). Applications SHOULD follow HTTP "common form"
when generating these constructs, since there might exist some
implementations that fail to accept anything beyond the common forms.
message-header = field-name ":" [ field-value ] CRLF
field-name = token
field-value = *( field-content | LWS )
field-content = < the OCTETs making up the field-value
and consisting of either *TEXT or combinations
of token, tspecials, and quoted-string>
The order in which header fields are received is not significant if
the header fields have different field names. Multiple header fields
with the same field-name may be present in a message if and only if
the entire field-value for that header field is defined as a comma-
separated list (i.e., #(values) ). It MUST be possible to combine the
multiple header fields into one "field-name: field-value" pair,
without changing the semantics of the message, by appending each
subsequent field-value to the first, each separated by a comma. The
order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field
value, and thus a proxy MUST NOT change the order of these field
values when a message is forwarded.
Field names are not case-sensitive, although their values may be.
6.6 Accept
See [H14.1]. This request header field is used only with the OPTIONS
request to indicate what description formats are acceptable.
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Example:
Accept: application/sdp;level=1, application/x-private
6.7 Accept-Language
See [H14.4]. The Accept-Language request header can be used to allow
the client to indicate to the server in which language it would
prefer to receive reason phrases. This may also be used as a hint by
the proxy as to which destination to connect the call to (e.g., for
selecting a human operator).
Example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.8 Allow
See [H14.7].
6.9 Authorization
See [H14.8].
6.10 Authentication
Authentication fields provide a digital signature of the remaining
fields for authentication purposes. They are not yet defined The use
of authentication headers is optional. If used, authentication
headers MUST be added to the header after the Via fields and before
the rest of the fields.
HS: Should probably re-use S/MIME here rather than invent
our own. Maybe better to fold into Authorization field.
6.11 Call-ID
The Call-ID uniquely identifies a particular invitation. Note that a
single multimedia conference may give rise to several calls, e.g., if
a user invites several different people. Calls to different callee
MUST always use different Call-IDs unless they are the result of a
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proxy server "forking" a single request.
The Call-ID may be any URL-encoded string that can be guaranteed to
be globally unique for the duration of the request. Using the
initiator's IP-address, process id, and instance (if more than one
request is being made simultaneously) satisfies this requirement.
The form local-id@host is recommended, where host is either the
fully qualified domain name or a globally routable IP address, and
local-id depends on the application and operating system of the host,
but is an ID that can be guaranteed to be unique during this session
initiation request.
Call-ID = ( "Call-ID" | "i" ) ":" atom "@" host
Example:
Call-ID: 9707211351.AA08181@foo.bar.com
6.12 Content-Length
The Content-Length entity-header field indicates the size of the
message-body, in decimal number of octets, sent to the recipient.
Content-Length = "Content-Length" ":" 1*DIGIT
An example is
Content-Length: 3495
Applications SHOULD use this field to indicate the size of the
message-body to be transferred, regardless of the media type of the
entity. Any Content-Length greater than or equal to zero is a valid
value. If no body is present in a message, then the Content-Length
header MAY be omitted or set to zero. Section 8 describes how to
determine the length of the message body.
6.13 Content-Type
The Content-Type entity-header field indicates the media type of the
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message-body sent to the recipient.
Content-Type = "Content-Type" ":" media-type
An example of the field is
Content-Type: application/sdp
6.14 Date
See [H14.19].
The Date header field is useful for simple devices without
their own clock.
6.15 Expires
The Expires header field gives the date/time after which the
registration expires.
This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a response-header field and
allows the server to indicate when the client has to re-register. For
INVITE, it is a request-header with which the callee can limit the
validity of an invitation. (For example, if a client wants to limit
how long a search should take at most or when a conference being
invited to is time-limited. A user interface may take this is as a
hint to leave the invitation window on the screen even if the user is
not currently at the workstation.)
The value of this field can be either an HTTP-date or an integer
number of seconds (in decimal), measured from the receipt of the
request.
Expires = "Expires" ":" ( HTTP-date | delta-seconds )
Two example of its use are
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5
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6.16 From
Requests MUST and responses SHOULD contain a From header field,
indicating the invitation initiator. The field MUST be a SIP URL as
defined in Section 2. Only a single initiator and a single invited
user are allowed to be specified in a single SIP request. The sense
of To and From header fields is maintained from request to
response, i.e., if the From header is sip://bob@example.edu in the
request then it is MUST also be sip://bob@example.edu in the response
to that request.
The From field is a URL and not a simple SIP address (Section 1.6
address to allow a gateway to relay a call into a SIP request and
still produce an appropriate From field. An example might be a
telephone call relayed into a SIP request where the from field might
contain a phone:// URL. Normally however this field will contain a
sip:// URL in either the long or short form.
If a SIP agent or proxy receives a request sourced From a URL
indicating a scheme other that SIP that is unknown to it, this MUST
NOT be treated as an error.
From = ( "From" | "f" ) ":" *1( ( SIP-URL | URL ) [ comment
] )
Example:
From: mjh@isi.edu (Mark Handley)
6.17 Location
The Location response header can be used with a 2xx or 3xx response
codes to indicate a new location to try. It contains a SIP URL giving
the new location or username to try, or may simply specify addition
transport parameters. For example, a "301 Moved Permanently" response
SHOULD contain a Location field containing the SIP URL giving the
new location and username to try. However, a "302 Moved Temporarily"
MAY give simply the same location and username that was being tried
but specify additional transport parameters such as a multicast
address to try or a change of transport from UDP to TCP or vice
versa.
A user agent or redirect server sending a definitive, positive
response (2xx), SHOULD insert a Location response header indicating
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the SIP address under which it is reachable most directly for future
SIP requests. This may be the address of the server itself or that of
a proxy (e.g., if the host is behind a firewall).
Location = ( "Location" | "m" ) ( SIP-URL | URL )
*( ";" location-params )
extension-name = token
extension-value = *( token | quoted-string | LWS | extension-specials)
extension-specials = < any element of tspecials except <"> >
language-tag = < see [H3.10] >
service-tag = "fax" | "IP" | "PSTN" | "ISDN" | "pager" | "voice-mail
| "attendant"
media-tag = < see SDP: "audio" | "video" | ...
feature-list = to be determined
location-params = "q" "=" qvalue
| "mobility" "=" ( "fixed" | "mobile" )
| "class" "=" ( "personal" | "business" )
| "language" "=" 1# language-tag
| "service" "=" 1# service-tag
| "media" "=" 1# media-tag
| "features" "=" 1# feature-list
| "description" "=" quoted-string
| "duplex" "=" ( "full" | "half" | "receive-only" |
"send-only" )
| extension-attributes
extension-attribute = extension-name "=" extension-value
Examples:
Location: sip://hgs@erlang.cs.columbia.edu ;service=IP,voice-mail
;media=audio ;duplex=full ;q=0.7
Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
language=en,es,iw ;q=0.5
Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
duplex=send-only;media=text; q=0.1
Attributes which are unknown should be omitted. New tags for class-
tag and service-tag can be registered with IANA. The media tag uses
Internet media types, e.g., audio, video, application/x-wb, etc. This
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is meant for indicating general communication capability, sufficient
for the caller to choose an appropriate address.
6.18 Organization
The Organization request-header fields conveys the name of the
organization to which the callee belongs. It may be inserted by
proxies at the boundary of an organization and may be used by client
software to filter calls.
6.19 PEP
This corresponds to the PEP header in the "Protocol Extension
Protocol" defined in RFC XXXX. The Protocol Extension Protocol (PEP)
is an extension mechanism designed to accommodate dynamic extension
of applications such as SIP clients and servers by software
components. The PEP general header declares new headers and whether
an application must or may understand them. Servers MUST parse this
field and MUST return "420 Bad Extension" when there is a PEP
extension of strength "must" (see RFC XXXX) that they do not
understand.
6.20 Priority
The priority request header signals the urgency of the call to the
callee.
Priority = "Priority" ":" priority-value
priority-value = "urgent" | "normal" | "non-urgent"
Example:
Subject: A tornado is heading our way!
Priority: urgent
6.21 Proxy-Authenticate
See [H14.33].
6.22 Proxy-Authorization
See [H14.34].
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6.23 Public
See [H14.35].
6.24 Reach
The Reach request header field allows the client to indicate whether
it wants to reach the group identified by the user part of the
address (value "all") or the first available individual (value
"first"). If not present, a value of "first" is implied. The "do-
not-forward" request prohibits proxies from forwarding the call to
another individual (e.g., the call is personal or the caller does not
want to be shunted to a secretary if the line is busy.) Section 1.6
describes the behavior of proxy servers when resolving group aliases.
Reach = "Reach" ":" 1#( "first" | "all" ) ( "do-not-
forward" )
Example:
Reach: first, do-not-forward
HS: This header is experimental.
6.25 Retry-After
The Retry-After response header field can be used with a "503
Service Unavailable" response to indicate how long the service is
expected to be unavailable to the requesting client and with a "404
Not Found" or "451 Busy" response to indicate when the called party
may be available again. The value of this field can be either an
HTTP-date or an integer number of seconds (in decimal) after the time
of the response.
Retry-After = "Retry-After" ":" ( HTTP-date | delta-seconds
)
Two examples of its use are
Retry-After: Mon, 21 Jul 1997 18:48:34 GMT
Retry-After: 120
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In the latter example, the delay is 2 minutes.
6.26 Sequence
The Sequence header field MAY be added by a SIP client making a
request if it needs to distinguish responses to several consecutive
requests sent with the same Call-ID. A Sequence field contains a
single decimal sequence number chosen by the requesting client.
Consecutive different requests made with the same Call-ID MUST
contain strictly monotonically increasing sequence numbers although
the sequence space MAY NOT be contiguous. A server responding to a
request containing a sequence number MUST echo the sequence number
back in the response.
Sequence = "Sequence" ":" 1*DIGIT
Sequence header fields are NOT needed for SIP requests using the
INVITE or OPTIONS methods but may be needed for future methods.
Example:
Sequence: 4711
6.27 Server
See [H14.39].
6.28 Subject
This is intended to provide a summary, or indicate the nature, of the
call, allowing call filtering without having to parse the session
description. (Also, the session description may not necessarily use
the same subject indication as the invitation.)
Subject = ( "Subject" | "s" ) ":" *text
Example:
Subject: Tune in - they are talking about your work!
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6.29 To
The To request header field specifies the invited user, with the
same SIP URL syntax as the From field.
To = ( "To" | "t" ) ":" ( SIP-URL | URL ) [ comment ]
If a SIP server receives a request destined To a URL indicating a
scheme other than SIP and that is unknown to it, the server returns a
"400 bad request" response.
Example:
To: sip://operator@cs.columbia.edu (The Operator)
6.30 User-Agent
See [H14.42].
6.31 Via
The Via field indicates the path taken by the request so far. This
prevents request looping and ensures replies take the same path as
the requests, which assists in firewall traversal and other unusual
routing situations.
In the request path, an initiator MUST add its own Via field to each
request. This Via field MUST be the first field in the request. Each
subsequent client or proxy that sends the message onwards MUST add
its own additional Via field, which MUST be added before any
existing Via fields. Additionally, if the message goes to a
multicast address, an extra Via field is added before all the others
giving the multicast address and TTL.
In the return path, Via fields are processed by a proxy or client
according to the following rules:
o If the first Via field in the reply received is the client's
or server's local address, remove the Via field and process
the reply.
o If the first Via field in a reply you are going to send is a
multicast address, remove that Via field before sending to the
multicast address.
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These rules ensure that a client or proxy server only has to check
the first Via field in a reply to see if it needs processing.
When a reply passes through a proxy on the reverse path, that proxies
Via field MUST be removed from the reply.
The format for a Via header is:
Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] )
via-params = "ttl" "=" ttl
| "fanout"
sent-protocol = [ protocol-name "/" ] protocol-version
[ "/" transport ]
protocol-name = "SIP" | token
protocol-version = token
transport = "UDP" | "TCP"
sent-by = host [ ":" port ]
ttl = 1*3DIGIT ; 0 to 255
The "ttl" parameter is included only if the address is a multicast
address. The "fanout" parameter indicates that this proxy has
initiated several connection attempts and that subsequent proxies
should not do the same.
Example:
Via: SIP/2.0/UDP first.example.com:4000 ;fanout
6.32 Warning
The Warning response-header field is used to carry additional
information about the status of a response. Warning headers are sent
with responses using:
Warning = "Warning" ":" 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text
warn-code = 2DIGIT
warn-agent = ( host [ ":" port ] ) | pseudonym
; the name or pseudonym of the server adding
; the Warning header, for use in debugging
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warn-text = quoted-string
A response may carry more than one Warning header.
The warn-text should be in a natural language and character set that
is most likely to be intelligible to the human user receiving the
response. This decision may be based on any available knowledge, such
as the location of the cache or user, the Accept-Language field in a
request, the Content-Language field in a response, etc. The default
language is English and the default character set is ISO- 8859-1.
Any server may add Warning headers to a response. New Warning
headers should be added after any existing Warning headers. A proxy
server MUST NOT delete any Warning header that it received with a
response.
When multiple Warning headers are attached to a response, the user
agent SHOULD display as many of them as possible, in the order that
they appear in the response. If it is not possible to display all of
the warnings, the user agent should follow these heuristics:
o Warnings that appear early in the response take priority over
those appearing later in the response.
o Warnings in the user's preferred character set take priority
over warnings in other character sets but with identical
warn-codes and warn-agents.
Systems that generate multiple Warning headers should order them
with this user agent behavior in mind.
Example:
Warning: 606.4 isi.edu Multicast not available
Warning: 606.2 isi.edu Incompatible protocol (RTP/XXP)
6.33 WWW-Authenticate
See [H14.46].
7 Status Code Definitions
The response codes are consistent with, and extend, HTTP/1.1 response
codes. Not all HTTP/1.1 response codes are appropriate, and only
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those that are appropriate are given here. Response codes not defined
by HTTP/1.1 have codes x80 upwards to avoid clashes with future HTTP
response codes. Also, SIP defines a new class, 6xx. The default
behavior for unknown response codes is given for each category of
codes.
7.1 Informational 1xx
Informational responses indicate that the server or proxy contacted
is performing some further action and does not yet have a definitive
response. The client SHOULD wait for a further response from the
server, and the server SHOULD send such a response without further
prompting. If UDP transport is being used, the client SHOULD
periodically re-send the request in case the final response is lost.
Typically a server should send a "1xx" response if it expects to take
more than one second to obtain a final reply.
7.1.1 100 Trying
Some further action is being taken (e.g., the request is being
forwarded) but the user has not yet been located.
7.1.2 180 Ringing
The user agent or conference server has located a possible location
where the user has been recently and is trying to alert them.
7.2 Successful 2xx
The request was successful and MUST terminate a search.
7.2.1 200 OK
The request was successful in contacting the user, and the user has
agreed to participate.
7.3 Redirection 3xx
3xx responses give information about the user's new location, or
about alternative services that may be able to satisfy the call.
They SHOULD terminate an existing search, and MAY cause the initiator
to begin a new search if appropriate.
7.3.1 300 Multiple Choices
The requested resource corresponds to any one of a set of
representations, each with its own specific location, and agent-
driven negotiation (i.e., controlled by the SIP client) is being
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provided so that the user (or user agent) can select a preferred
communication end point and redirect its request to that location.
The response SHOULD include an entity containing a list of resource
characteristics and location(s) from which the user or user agent can
choose the one most appropriate. The entity format is specified by
the media type given in the Content-Type header field. Depending
upon the format and the capabilities of the user agent, selection of
the most appropriate choice may be performed automatically. However,
this specification does not define any standard for such automatic
selection.
The choices SHOULD also be listed as Location fields (Section 6.17).
Unlike HTTP, the SIP response may contain several Location fields.
User agents MAY use the Location field value for automatic
redirection or MAY ask the user to confirm a choice.
7.3.2 301 Moved Permanently
The requesting client should retry on the new address given by the
Location field because the user has permanently moved and the address
this response is in reply to is no longer a current address for the
user. A 301 response MUST NOT suggest any of the hosts in the Via
path of the request as the user's new location.
7.3.3 302 Moved Temporarily
The requesting client should retry on the new address(es) given by
the Location header. A 302 response MUST NOT suggest any of the hosts
in the Via path of the request as the user's new location.
7.3.4 380 Alternative Service
The call was not successful, but alternative services are possible.
The alternative services are described in the message body of the
response.
7.4 Request Failure 4xx
4xx responses are definite failure responses from a particular
server. The client SHOULD NOT retry the same request without
modification (e.g., adding appropriate authorization). However, the
same request to a different server may be successful.
7.4.1 400 Bad Request
The request could not be understood due to malformed syntax.
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7.4.2 401 Unauthorized
The request requires user authentication.
7.4.3 402 Payment Required
Reserved for future use.
7.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request should not be repeated.
7.4.5 404 Not Found
The server has definitive information that the user does not exist at
the domain specified in the Request-URI.
7.4.6 405 Method Not Allowed
The method specified in the Request-Line is not allowed for the
address identified by the Request-URI. The response MUST include an
Allow header containing a list of valid methods for the indicated
address.
7.4.7 407 Proxy Authentication Required
This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with the proxy. The proxy MUST
return a Proxy-Authenticate header field (section 6.21) containing a
challenge applicable to the proxy for the requested resource. The
client MAY repeat the request with a suitable Proxy-Authorization
header field (section 6.22). SIP access authentication is explained
in section [H11].
This status code should be used for applications where access to the
communication channel (e.g., a telephony gateway) rather than the
callee herself requires authentication.
7.4.8 408 Request Timeout
The client did not produce a request within the time that the server
was prepared to wait. The client MAY repeat the request without
modifications at any later time.
7.4.9 420 Bad Extension
The server did not understand the protocol extension specified with
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strength "must".
7.4.10 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is
currently unavailable (e.g., not logged in or logged in in such a
manner as to preclude communication with the callee). The response
may indicate a better time to call in the Retry-After header. The
user may also be available elsewhere (unbeknownst to this host),
thus, this response does terminate any searches.
7.5 Server Failure 5xx
5xx responses are failure responses given when a server itself has
erred. They are not definitive failures, and SHOULD NOT terminate a
search if other possible locations remain untried.
7.5.1 500 Server Internal Error
The server encountered an unexpected condition that prevented it from
fulfilling the request.
7.5.2 501 Not implemented
The server does not support the functionality required to fulfill the
request. This is the appropriate response when the server does not
recognize the request method and is not capable of supporting it for
any user.
7.5.3 502 Bad Gateway
The server, while acting as a gateway or proxy, received an invalid
response from the upstream server it accessed in attempting to
fulfill the request.
7.5.4 503 Service Unavailable
The server is currently unable to handle the request due to a
temporary overloading or maintenance of the server. The implication
is that this is a temporary condition which will be alleviated after
some delay. If known, the length of the delay may be indicated in a
Retry-After header. If no Retry-After is given, the client SHOULD
handle the response as it would for a 500 response.
Note: The existence of the 503 status code does not imply that a
server must use it when becoming overloaded. Some servers may wish to
simply refuse the connection.
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7.5.5 504 Gateway Timeout
The server, while acting as a gateway, did not receive a timely
response from the upstream server (e.g., a location server) it
accessed in attempting to complete the request.
7.6 Global Failures
6xx responses indicate that a server has definitive information about
a particular user, not just the particular instance indicated in the
Request-URI. All further searches for this user are doomed to failure
and pending searches SHOULD be terminated.
7.6.1 600 Busy
The callee's end system was contacted successfully but the callee is
busy and does not wish to take the call at this time. The response
may indicate a better time to call in the Retry-After header. If the
callee does not wish to reveal the reason for declining the call, the
callee should use status code 680 instead.
7.6.2 603 Decline
The callee's machine was successfully contacted but the user
explicitly does not wish to participate. The response may indicate a
better time to call in the Retry-After header.
7.6.3 604 Does not exist anywhere
The server has authoritative information that the user indicated in
the To request field does not exist anywhere. Searching for the user
elsewhere will not yield any results.
7.6.4 606 Not Acceptable
The user's agent was contacted successfully but some aspects of the
session profile (the requested media, bandwidth, or addressing style)
were not acceptable.
A "606 Not Acceptable" reply means that the user wishes to
communicate, but cannot adequately support the session described. The
"604 Not Acceptable" reply MAY contain a list of reasons in a Warning
header describing why the session described cannot be supported.
These reasons can be one or more of:
606.1 Insufficient Bandwidth: The bandwidth specified in the session
description or defined by the media exceeds that known to be
available.
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606.2 Incompatible Protocol: One or more protocols described in the
request are not available.
606.3 Incompatible Format: One or more media formats described in the
request is not available.
606.4 Multicast not available: The site where the user is located
does not support multicast.
606.5 Unicast not available: The site where the user is located does
not support unicast communication (usually due to the presence
of a firewall).
Other reasons are likely to be added later. It is hoped that
negotiation will not frequently be needed, and when a new user is
being invited to join a pre-existing lightweight session, negotiation
may not be possible. It is up to the invitation initiator to decide
whether or not to act on a "606 Not Acceptable" reply.
8 SIP Message Body
The session description body gives details of the session the user is
being invited to join. Its Internet media type MUST be given by the
Content-type header field, and the body length in bytes MUST be given
by the Content-Length header field. If the body has undergone any
encoding (such as compression) then this MUST be indicated by the
Content-encoding header field, otherwise Content-encoding MUST be
omitted.
If required, the session description can be encrypted using public
key cryptography, and then can also carry private session keys for
the session. If this is the case, four random bytes are added to the
beginning of the session description before encryption and are
removed after decryption but before parsing.
8.1 Body Inclusion
For a request message, the presence of a body is signaled by the
inclusion of a Content-Length header. A body may be included in a
request only when the request method allows one.
For response messages, whether or not a body is included is dependent
on both the request method and the response message's response code.
All 1xx informational responses MUST NOT include a body. All other
responses MAY include a payload, although it may be of zero length.
8.2 Message Body Length
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If no body is present in a message, then the Content-Length header
MAY be omitted or set to zero. When a body is included, its length in
bytes is indicated in the Content-Length header and is determined by
one of the following:
1. Any response message which MUST NOT include a body (such as
the 1xx responses) is always terminated by the first empty
line after the header fields, regardless if any entity-
header fields are present.
2. Otherwise, a Content-Length header MUST be present (this
requirement differs from HTTP/1.1). Its value in bytes
represents the length of the message body.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
9 Examples
9.1 Invitation
9.1.1 Request
The example below is a request message en route from initiator to
invitee:
C->S: INVITE schooler@vlsi.cs.caltech.edu SIP/2.0
Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu (Mark Handley)
Subject: SIP will be discussed, too
To: schooler@cs.caltech.edu (Eve Schooler)
Call-ID: 62729-27@oregon.isi.edu
Content-type: application/sdp
Content-Length: 187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
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The first line above states that this is a SIP version 2.0 request.
The Via fields give the hosts along the path from invitation
initiator (the first element of the list) towards the invitee. In the
example above, the message was last multicast to the administratively
scoped group 239.128.16.254 with a ttl of 16 from the host
131.215.131.131
The request header above states that the request was initiated by
mjh@isi.edu the host 128.16.64.19 schooler@cs.caltech.edu is being
invited; the message is currently being routed to
schooler@vlsi.cs.caltech.edu
In this case, the session description is using the Session
Description Protocol (SDP), as stated in the Content-type header.
The header is terminated by an empty line and is followed by a
message body containing the session description.
9.1.2 Reply
The called user agent, directly or indirectly through proxy servers,
indicates that it is alerting ("ringing") the called party:
S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 1
From: mjh@isi.edu
Call-ID: 62729-27@128.16.64.19
Location: sip://es@jove.cs.caltech.edu
A sample reply to the invitation is given below. The first line of
the reply states the SIP version number, that it is a "200 OK" reply,
which means the request was successful. The Via headers are taken
from the request, and entries are removed hop by hop as the reply
retraces the path of the request. A new authentication field MAY be
added by the invited user's agent if required. The Call-ID is taken
directly from the original request, along with the remaining fields
of the request message. The original sense of From field is
preserved (i.e., it is the session initiator).
In addition, the Location header gives details of the host where the
user was located, or alternatively the relevant proxy contact point
which should be reachable from the caller's host.
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S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 1
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
Location: sip://es@jove.cs.caltech.edu
For two-party Internet phone calls, the response must contain a
description of where to send data to, for example the reply from
schooler to mjh :
S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 1
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
Location: sip://es@jove.cs.caltech.edu
Content-Length: 102
v=0
o=schooler 4858949 4858949 IN IP4 192.1.2.3
t=0 0
m=audio 5004 RTP/AVP 0
c=IN IP4 131.215.131.147
The caller confirms the invitation by sending a request to the
location named in the Location header:
C->S: CONNECTED schooler@jove.cs.caltech.edu SIP/2.0
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
9.1.3 Aborting a Call
If the caller wants to abort a pending call, it sends a BYE request.
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C->S: BYE schooler@jove.cs.caltech.edu
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
9.1.4 Redirects
Replies with response codes "301 Moved Permanently" or "302 Moved
Temporarily" SHOULD specify another location using the Location
field.
S->C: SIP/2.0 302 Moved temporarily
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
Location: sip://239.128.16.254;ttl=16;transport=udp
Content-length: 0
In this example, the proxy located at 131.215.131.131 is being
advised to contact the multicast group 239.128.16.254 with a ttl of
16 and UDP transport. In normal situations, a server would not
suggest a redirect to a local multicast group unless, as in the above
situation, it knows that the previous proxy or client is within the
scope of the local group. If the request is redirected to a multicast
group, a proxy server SHOULD query the multicast address itself
rather than sending the redirect back towards the client as multicast
may be scoped; this allows a proxy within the appropriate scope
regions to make the query.
9.1.5 Alternative Services
An example of a "350 Alternative Service" reply is:
S->C: SIP/2.0 350 Alternative Service
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
Location: recorder@131.215.131.131
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Content-type: application/sdp
Content-length: 146
v=0
o=mm-server 2523535 0 IN IP4 131.215.131.131
s=Answering Machine
i=Leave an audio message
c=IN IP4 131.215.131.131
t=0 0
m=audio 12345 RTP/AVP 0
In this case, the answering server provides a session description
that describes an "answering machine". If the invitation initiator
decides to take advantage of this service, it should send an
invitation request to the answering machine at 131.215.131.131 with
the session description provided (modified as appropriate for a
unicast session to contain the appropriate local address and port for
the invitation initiator). This request SHOULD contain a different
Call-ID from the one in the original request. An example would be:
C->S: INVITE mm-server@131.215.131.131 SIP/2.0
Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-28@128.16.64.19
Content-type: application/sdp
Content-length: 146
v=0
o=mm-server 2523535 0 IN IP4 131.215.131.131
s=Answering Machine
i=Leave an audio message
c=IN IP4 128.16.64.19
t=0 0
m=audio 26472 RTP/AVP 0
Invitation initiators MAY choose to treat a "350 Alternative Service"
reply as a failure if they wish to do so.
9.1.6 Negotiation
An example of a "606 Not Acceptable" reply is:
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S->C: SIP/2.0 606 Not Acceptable
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID:62729-27@128.16.64.19
Location: mjh@131.215.131.131
Warning: 606.1 Insufficient bandwidth (only have ISDN),
606.3 Incompatible format,
606.4 Multicast not available
Content-Type: application/sdp
Content-Length: 50
v=0
s=Lets talk
b=CT:128
c=IN IP4 131.215.131.131
m=audio 3456 RTP/AVP 7 0 13
m=video 2232 RTP/AVP 31
In this example, the original request specified 256 kb/s total
bandwidth, and the reply states that only 128 kb/s is available. The
original request specified GSM audio, H.261 video, and WB whiteboard.
The audio coding and whiteboard are not available, but the reply
states that DVI, PCM or LPC audio could be supported in order of
preference. The reply also states that multicast is not available.
In such a case, it might be appropriate to set up a transcoding
gateway and re-invite the user.
9.2 OPTIONS Request
A caller Alice can use an OPTIONS request to find out the
capabilities of a potential callee Bob, without "ringing" the
designated address. In this case, Bob indicates that he can be
reached at three different addresses, ranging from voice-over-IP to a
PSTN phone to a pager.
C->S: OPTIONS bob@example.com SIP/2.0
From: alice@anywhere.org (Alice)
To: bob@example.com (Bob)
Accept: application/sdp
S->C: SIP/2.0 200 OK
Location: sip://bob@host.example.com ;service=IP,voice-mail
;media=audio ;duplex=full ;q=0.7
Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
language=en,es,iw ;q=0.5
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Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
duplex=send-only;media=text; q=0.1
Alternatively, Bob could have returned a description of
C->S: OPTIONS bob@example.com SIP/2.0
From: alice@anywhere.org (Alice)
To: bob@example.com (Bob)
Accept: application/sdp
S->C: SIP/2.0 200 OK
Content-Length: 81
Content-Type: application/sdp
v=0
m=audio 0 RTP/AVP 0 1 3 99
m=video 0 RTP/AVP 29 30
a:rtpmap:98 SX7300/8000
10 Compact Form
When SIP is carried over UDP with authentication and a complex
session description, it may be possible that the size of a request or
reply is larger than the MTU. To reduce this problem, a more compact
form of SIP is also defined by using alternative names for common
header fields. These short forms are NOT abbreviations, they are
field names. No other abbreviations are allowed.
short field name long field name note
c Content-Type
e Content-Encoding
f From
i Call-ID
l Content-Length
m Location from "moved"
s Subject
t To
v Via
Thus the header in section 9.1 could also be written:
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INVITE schooler@vlsi.caltech.edu SIP/2.0
v:SIP/2.0/UDP 239.128.16.254 16
v:SIP/2.0/UDP 131.215.131.131
v:SIP/2.0/UDP 128.16.64.19
f:mjh@isi.edu
t:schooler@cs.caltech.edu
i:62729-27@128.16.64.19
c:application/sdp
l:187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
Mixing short field names and long field names is allowed, but not
recommended. Servers MUST accept both short and long field names for
requests. Proxies MUST NOT translate a request between short and long
forms if authentication fields are present.
11 SIP Transport
SIP is defined so it can use either UDP or TCP as a transport
protocol.
11.1 Achieving Reliability For UDP Transport
11.1.1 General Operation
SIP assumes no additional reliability from IP. Requests or replies
may be lost. A SIP client SHOULD simply retransmit a SIP request
periodically with timer T1 (default value of T1: once a second) until
it receives a response, or until it has reached a set limit on the
number of retransmissions. The default limit is 20.
SIP requests and replies are matched up by the client using the
Call-ID header field; thus, a server can only have one outstanding
request per call at any given time.
HS: A transaction or request ID would remove this
limitation.
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If the reply is a provisional response, the initiating client SHOULD
continue retransmitting the request, albeit less frequently, using
timer T2. The default retransmission interval T2 is 5 seconds.
After the server sends a final response, it cannot be sure the client
has received the response, and thus SHOULD cache the results for at
least 30 seconds to avoid having to, for example, contact the user or
user location server again upon receiving a retransmission.
11.1.2 INVITE
Special considerations apply for the INVITE method.
1. After receiving an invitation, considerable time may elapse
before the server can determine the outcome. For example,
the called party may be "rung" or extensive searches may be
performed, so delays can reach several tens of seconds.
2. It is possible that the invitation request reaches the
callee and the callee is willing to take the call, but that
the final response (200 OK, in this case) is lost on the
way to the caller. If the session still exists but the
initiator gives up on including the user, the contacted
user has sufficient information to be able to join the
session. However, if the session no longer exists because
the invitation initiator "hung up" before the reply arrived
and the session was to be two-way, the conferencing system
should be prepared to deal with this circumstance.
3. If a telephony user interface is modeled or if we need to
interface to the PSTN, the caller will provide "ringback",
a signal that the callee is being alerted. Once the callee
picks up, the caller needs to know so that it can enable
the voice path and stop ringback. The callee's response to
the invitation could get lost. Unless the response is
transmitted reliably, the caller will continue to hear
ringback while the callee assumes that the call exists.
4. The client has to be able to terminate an on-going request,
e.g., because it is no longer willing to wait for the
connection or search to succeed. One cannot rely on the
absence of request retransmission, since the server would
have to continue honoring the request for several request
retransmission periods, that is, possible tens of seconds
if only one or two packets can be lost.
The first problem is solved by indicating progress to the caller: the
server returns a provisional response indicating it is searching or
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ringing the user.
The server retransmits the final response at intervals of T3 (default
value of T3 = 2 seconds) until it receives a CONNECTED request for
the same Call-ID or until it has retransmitted the final response 10
times. The CONNECTED request is acknowledged only once. If the
request is syntactically valid and the Request-URI matches that in
the INVITED request with the same Call-ID, the server answers with
status code 200, otherwise with status code 400.
Fig. 4 and 5 show the client and server state diagram for
invitations.
11.2 Connection Management for TCP
A single TCP connection can serve one or more SIP transactions. A
transaction contains zero or more provisional responses followed by
exactly one final response.
The client MAY close the connection at any time. Closing the
connection before receiving a final response signals that the client
wishes to abort the request.
The server SHOULD NOT close the TCP connection until it has sent its
final response, at which point it MAY close the TCP connection if it
wishes to. However, normally it is the client's responsibility to
close the connection.
If the server leaves the connection open, and if the client so
desires it may re-use the connection for further SIP requests or for
requests from the same family of protocols (such as HTTP or stream
control commands).
12 Behavior of SIP Servers
This section describes behavior of a SIP server in detail. Servers
can operate in proxy or redirect mode. Proxy servers can "fork"
connections, i.e., a single incoming request spawns several outgoing
(client) requests.
A proxy server always inserts a Via header field containing their
own address into requests it issues that are caused by an incoming
request.
We define an "A--B proxy" as a proxy that receives SIP requests over
transport protocol A and issues requests, acting as a SIP client,
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+===========+
| Initial |
+===========+
|
|
| -
| ------
| INVITE
+------v v
T1 +-----------+
------ | Calling |-------------------+
INVITE +-----------+ |
+------| | | |
+----------------+ | |
| | |
| | |
| | |
| | |
| +------v v v-----| |
| T2 +-----------+ 1xx |
| ------ | Ringing | --- |
| INVITE +-----------+ - |
| +------| | | |-----+ |
| | +--------------+ |
| 2xx | | >=300 |
| --------- | 2xx | ----- |
| CONNECTED | --------- | - |
| | CONNECTED | |
+----------------+ | | |
+------v | v v v
2xx +-----------+ +-----------+
--------- | Connected | | Failure |
CONNECTED +-----------+ +-----------+
+------|
event
-------
message
Figure 4: State transition diagram of client for INVITE method
using transport protocol B. If not stated explicitly, rules apply to
any combination of transport protocols. For conciseness, we only
describe behavior with UDP and TCP, but the same rules apply for any
unreliable datagram or reliable protocol, respectively.
Handley/Schulzrinne/Schooler [Page 51]
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+===========+
+------------>| Initial |<-------------+
| +===========+ |
| | |
| failure | |
| ----------- | INVITE |
| 3xx,4xx,5xx | ------ |
| | 1xx |
| +------v v |
| INVITE +-----------+ |
| ------ | Searching | |
| 1xx +-----------+ |
| +------| | | +---------------->+
| | | |
| | | callee picks up |
+----------------+ | --------------- |
| 200 |
| | BYE
+------v v v-----| | ---
INVITE +-----------+ T3 | 200
------ | Answered | --- |
1xx +-----------+ 200 |
+------| | | |-----+ |
| +---------------->+
| |
| CONNECTED |
| --------- |
| 200 |
| |
+------v v |
CONNECTED +-----------+ |
--------- | Connected | |
200 +-----------+ |
+------| | |
+-----------------+
event
-------
message
Figure 5: State transition diagram of server for INVITE method
The detailed connection behavior for UDP and TCP is described in
Section 11.
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12.1 Redirect Server
A redirect server does not issue any SIP requests of its own. It can
return a response that accepts, refuses or redirects the request.
After receiving a request, a redirect server proceeds through the
following steps:
1. If the request cannot be answered immediately (e.g.,
because a location server needs to be contacted), it
returns one or more provisional responses.
2. Once the server has gathered the list of alternative
locations or has decided to accept or refuse the call, it
returns the final response. This ends the SIP transaction.
The redirect server maintains transaction state for the whole SIP
transaction. Servers in user agents are redirect servers.
12.2 Proxies Issuing Single Unicast Requests
Proxies in this category issue at most a single unicast request for
each incoming SIP request, that is, they do not "fork" requests.
Servers may choose to always operate in the mode described in Section
12.3.
12.2.1 UDP--UDP Proxy Server
The UDP--UDP server can forward the request and any responses. It
does not have to maintain any state for the SIP transaction. UDP
reliability is assured by the next redirect server in the server
chain.
12.2.2 UDP--TCP Proxy Server
A proxy server issuing a single request over TCP maintains state for
the whole SIP transaction indexed by the Call-ID.
If it receives a UDP retransmission of the same request for an
existing session, it retransmits the last response received from the
TCP side. Any changes in the message body compared to the last
request for the Call-ID are silently ignored. (Otherwise, the proxy
would have to remember and compare the message body; this also
violates the notion of a SIP transaction. TBD) The server SHOULD
cache the final response for a particular Call-ID after the SIP
transaction on the TCP side has completed.
After the cache entry has been expired, the server cannot tell
whether an incoming request is actually a retransmission of an older
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request, where the TCP side has terminated. It will treat it as a new
request.
12.3 Proxy Server Issuing Several Requests
All requests carry the same Call-ID. For unicast, each of the
requests has a different (host-dependent) Request-URI. For
multicast, a single request is issued, likely with a host-independent
Request-URI. A client receiving a multicast query does not have to
check whether the host part of the Request-URI matches its own host
or domain name. To avoid response implosion, servers SHOULD NOT
answer multicast requests with a 404 (Not Found) status code.
Servers MAY decide not to answer multicast requests if their response
would be 5xx.
The server MAY respond to the request immediately with a "100 Trying"
response; otherwise it MAY wait until either the first response to
its requests or the UDP retransmission interval.
The following pseudo-code describes the behavior of a proxy server
issuing several requests in response to an incoming request. The
function request(a) sends a SIP request to address a.
await_response() waits until a response is received and returns the
response. request_close(a) closes the TCP connection to client with
address a; this is optional. response(s, l, L) sends a response to
the client with status s and list of locations L, with l entries.
ismulticast() returns 1 if the location is a multicast address and
zero otherwise. The variable timeleft indicates the amount of time
left until the maximum response time has expired. The variable
recurse indicates whether the server will recursively try addresses
returned through a 3xx response. A server MAY decide to recursively
try only certain addresses, e.g., those which are within the same
domain as the proxy server. Thus, an initial multicast request may
trigger additional unicast requests.
int N = 0; /* number of connection attempts */
address_t address[]; /* list of addresses */
location[]; /* list of locations */
int heard = 0; /* number of sites heard from */
int class; /* class of status code */
int best = 1000; /* best response so far */
int timeleft = 120; /* sample timeout value */
int loc = 0; /* number of locations */
struct { /* response */
int status; /* response status */
char *location; /* redirect locations */
address_t a; /* address of respondent */
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} r;
int i;
if (multicast) {
request(address[0]);
} else {
N = /* number of addresses to try */
for (i = 0; i < N; i++) {
request(address[i]);
}
}
while (timeleft > 0 && (heard < N || multicast)) {
r = await_response();
class = r.status / 100;
if (class >= 2) {
heard++;
if (tcp) request_close(a);
}
if (class == 2) {
best = r.status;
break;
}
else if (class == 3) {
/* A server may optionally recurse. The server MUST check whether
* it has tried this location before and whether the location is
* part of the Via path of the incoming request. This check is
* omitted here for brevity. Multicast locations MUST NOT be
* returned to the client if the server is not recursing.
*/
if (recurse) {
multicast = 0;
N++;
request(r.location);
} else if (!ismulticast(r.location)) {
locations[loc++] = r.location;
best = r.status;
}
}
else if (class == 4) {
if (best >= 400) best = r.status;
}
else if (class == 5) {
if (best >= 500) best = r.status;
}
else if (class == 6) {
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best = r.status;
break;
}
}
/* We haven't heard anything useful from anybody. */
if (best == 1000) {
best = 404;
}
if (best/100 != 3) locs = 0;
response(best, locs, locations);
When operating in this mode, a proxy server MUST ignore any responses
received for Call-IDs that it does not have a pending transaction
for. (If server were to forward responses not belonging to a current
transaction using the Via field, the requesting client would get
confused if it has just issued another request using the same Call-
ID.)
13 Security Considerations
13.1 Confidentiality
Unless SIP transactions are protected by lower-layer security
mechanisms such as SSL , an attacker may be able to eavesdrop on call
establishment and invitations and, through the Subject header field
or the session description, gain insights into the topic of
conversation.
13.2 Integrity
Unless SIP transactions are protected by lower-layer security
mechanisms such as SSL , an active attacker may be able to modify SIP
requests.
13.3 Access Control
SIP requests are not authenticated unless the SIP Authorization and
WWW-Authenticate headers are being used. The strengths and weaknesses
of these authentication mechanisms are the same as for HTTP.
13.4 Privacy
User location and SIP-initiated calls may violate a callee's privacy.
An implementation SHOULD be able to restrict, on a per-user basis,
what kind of location and availability information is given out to
certain classes of callers.
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A Summary of Augmented BNF
In this specification we use the Augmented Backus-Naur Form notation
described in [19]. For quick reference, the following is a brief
summary of the main features of this ABNF.
"abc"
The case-insensitive string of characters "abc" (or "Abc",
"aBC", etc.);
%d32
The character with ASCII code decimal 32 (space);
*term
zero of more instances of term;
3*term
three or more instances of term;
2*4term
two, three or four instances of term;
[ term ]
term is optional;
term1 term2 term3
set notation: term1, term2 and term3 must all appear but
their order is unimportant;
term1 | term2
either term1 or term2 may appear but not both;
#term
a comma separated list of term;
2#term
a comma separated list of term containing at least 2 items;
2#4term
a comma separated list of term containing 2 to 4 items.
Common Tokens
Certain tokens are used frequently in the BNF this document, and not
defined elsewhere. Their meaning is well understood but we include it
here for completeness.
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CR = %d13 ; carriage return character
LF = %d10 ; line feed character
CRLF = CR LF ; typically the end of a line
SP = %d32 ; space character
TAB = %d09 ; tab character
LWS = *( SP | TAB) ; linear whitespace
DIGIT = "0" .. "9" ; a single decimal digit
Changes
Since version -01, the following things have changed:
o Added personal note to "Searching" section indicating that 6xx
codes may not be necessary. Added figures.
o Initial author's note removed; dated.
o Introduction rewritten to give quick, concise overview as to
what SIP does.
o Conference control (tight vs. loose) seems less and less
appropriate. All share some state such as notions of
membership; some (ITU versions) tend to keep it in a central
server, others distribute it. Some state is synchronized at
larger timescales than other. (After all, even a server won't
know if a participant disconnects from the network until TCP
keep-alive, if any, kicks in.)
o Added list of related protocols to emphasize that this is part
of a whole architecture.
o Terminology: user always reminds me of controlled substances;
thus, this term is avoided where better terminology exists.
Since this protocol sits at the boundary between traditional
Internet and telephony services, some of the terminology
familiar in that realm is introduced.
o Terminology: user location server replaced by redirect server,
since a proxy server may also invoke user location. Also, the
actual user location server (e.g., an LDAP, ULS or similar
directory) may be invoked using protocols other than SIP.
o Rearranged ordering of address resolution to correspond to
host requirements for MX and suggestions in DNS SRV RFC. Adding
note about caching and socket implementation. Added note about
using SMTP EXPN to get an alternate address.
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o Defined SIP transaction, provisional and final responses.
o Assigned values to timeout parameters; otherwise, there will
be unnecessary retransmissions between different
implementations.
o Retransmission was greatly simplified; there does not seem to
be a need for all the rules governing transitions between TCP
and UDP domains. A proxy should look just like a server to one
side and like a client to the other. Proxies need to maintain
transaction state in any event since they need to remember
where they forwarded the last SIP request to ( Confirm wouldn't
work otherwise, for example.). Invoking a location service may
yield inconsistent results, introduces additional failure modes
(what if the location server is temporarily unavailable?),
increases delay and processing overhead. UDP--UDP proxies can
still be built without state; they just forward packets and
responses. Proxies with TCP on one and UDP on the other side
will have to act like a normal UDP server and issue 100
responses.
o Removed redundancies and contradictions from request and
response definitions (space vs. SP, duplicate CRLF definition,
recursive request-header, ...).
o Added the experimental methods CONNECTED, REGISTER,
UNREGISTER and BYE.
o Re-engineered the invitation reliability mechanism to use a
separate confirmation message.
o Tentative increase of MTU to 1500 bytes, as per discussion in
Stevens.
o Added Reach, Organization, Subject, Priority,
Authorization, WWW-Authentication headers for improved call
handling. WWW "basic" authentication isn't great, but it is
widely deployed and probably sufficient for giving out
"private" telephone numbers, particularly those where the
callee incurs a charge. (I want to be able to give somebody a
password to call my 800 number via an Internet gateway;
authenticating who that person is requires that I modify a
script on my server to add another distinguished name to the
list of allowable callees.)
o Renamed Reason to Warning (to align with HTTP) header since
the response line already offers a failure reason.
Unfortunately, listing several failures is not all that helpful
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Internet Draft SIP July 31, 1997
since the calling party cannot determine which of the media
within the description causes the difficulty or whether it was
the set of media as a whole, but it may give the user agent
some indication as to what's going on.
o SEP and CRLF in headers removed, since this is always implied
between items. Missing ":" added. CRLF was already in the
message definition. Also, unlike RFC 822 and HTTP, the
definition did not allow spaces between the field name and the
colon.
o Added (reluctantly) password to URL. It's no worse than ftp
and needed to easily call from a secure web page, without
having to type in a password manually.
o Added port to SIP URL to specify non-standard port.
o CAPABILITIES to OPTIONS for closer alignment with HTTP and
RTSP;
o Path to Via for closer alignment with HTTP and RTSP;
o Content type meta changed to application, since "meta" doesn't
exist as a top-level Internet media type.
o Formatting closer to HTTP and RTSP.
o Explain relationship to H.323.
B Open Issues
RELIABLE: How to provide reliability?
BYE: Use of BYE method?
REGISTER: Use of REGISTER method?
H.323: Interaction with H.323 and H.245.
TRANSACTION: Should we have a transaction id in addition to a call
ID? Call-IDs are for the end system, but a transaction ID is for
a single SIP exchange. This is useful for Internet telephony,
where a single call may trigger several transactions.
C Acknowledgments
We wish to thank the members of the IETF MMUSIC WG for their comments
and suggestions. This work is based, inter alia, on [23,24].
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Parameters of the terminal negotiation mechanism were influenced by
Scott Petrack's CMA design.
D Authors' Addresses
Mark Handley
USC Information Sciences Institute
c/o MIT Laboratory for Computer Science
545 Technology Square
Cambridge, MA 02139
USA
electronic mail: mjh@isi.edu
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Eve Schooler
Computer Science Department 256-80
California Institute of Technology
Pasadena, CA 91125
USA
electronic mail: schooler@cs.caltech.edu
E Bibliography
[1] R. Pandya, "Emerging mobile and personal communication systems,"
IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.
[2] R. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin,
"Resource reservation protocol (RSVP) -- version 1 functional
specification," Internet Draft, Internet Engineering Task Force, June
1997. Work in progress.
[3] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
transport protocol for real-time applications," RFC 1889, Internet
Engineering Task Force, Jan. 1996.
[4] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
Mar. 1997. Work in progress.
[5] M. Handley, "SAP: Session announcement protocol," Internet Draft,
Internet Engineering Task Force, Nov. 1996. Work in progress.
Handley/Schulzrinne/Schooler [Page 61]
Internet Draft SIP July 31, 1997
[6] M. Handley and V. Jacobson, "SDP: Session description protocol,"
Internet Draft, Internet Engineering Task Force, Mar. 1997. Work in
progress.
[7] P. Lantz, "Usage of H.323 on the Internet," Internet Draft,
Internet Engineering Task Force, Feb. 1997. Work in progress.
[8] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[9] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-Lee,
"Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
Engineering Task Force, Jan. 1997.
[10] C. Partridge, "Mail routing and the domain system," STD 14, RFC
974, Internet Engineering Task Force, Jan. 1986.
[11] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the
location of services (DNS SRV)," RFC 2052, Internet Engineering Task
Force, Oct. 1996.
[12] P. Mockapetris, "Domain names - implementation and
specification," STD 13, RFC 1035, Internet Engineering Task Force,
Nov. 1987.
[13] R. Braden, "Requirements for internet hosts - application and
support," STD 3, RFC 1123, Internet Engineering Task Force, Oct.
1989.
[14] D. Zimmerman, "The finger user information protocol," RFC 1288,
Internet Engineering Task Force, Dec. 1991.
[15] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
protocol," RFC 1777, Internet Engineering Task Force, Mar. 1995.
[16] T. Berners-Lee, "Universal resource identifiers in WWW: a
unifying syntax for the expression of names and addresses of objects
on the network as used in the world-wide web," RFC 1630, Internet
Engineering Task Force, June 1994.
[17] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
locators (URL): Generic syntax and semantics," Internet Draft,
Internet Engineering Task Force, May 1997. Work in progress.
[18] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
1994.
Handley/Schulzrinne/Schooler [Page 62]
Internet Draft SIP July 31, 1997
[19] D. Crocker, "Augmented BNF for syntax specifications: ABNF,"
Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in
progress.
[20] J. Mogul and S. Deering, "Path MTU discovery," RFC 1191,
Internet Engineering Task Force, Nov. 1990.
[21] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1.
Reading, Massachusetts: Addison-Wesley, 1994.
[22] D. Crocker, "Standard for the format of ARPA internet text
messages," STD 11, RFC 822, Internet Engineering Task Force, Aug.
1982.
[23] E. M. Schooler, "Case study: multimedia conference control in a
packet-switched teleconferencing system," Journal of Internetworking:
Research and Experience , vol. 4, pp. 99--120, June 1993. ISI
reprint series ISI/RS-93-359.
[24] H. Schulzrinne, "Personal mobility for multimedia services in
the Internet," in European Workshop on Interactive Distributed
Multimedia Systems and Services , (Berlin, Germany), Mar. 1996.
Table of Contents
1 Introduction ........................................ 2
1.1 Overview of SIP Functionality ....................... 2
1.2 Finding Multimedia Sessions ......................... 3
1.3 Terminology ......................................... 4
1.4 Definitions ......................................... 4
1.5 Protocol Properties ................................. 6
1.5.1 Minimal State ....................................... 6
1.5.2 Transport-Protocol Neutral .......................... 6
1.5.3 Text-Based .......................................... 6
1.6 SIP Addressing ...................................... 6
1.7 Locating a SIP Server ............................... 8
1.8 SIP Transactions .................................... 9
1.9 Locating a User ..................................... 9
2 SIP Uniform Resource Locators ....................... 12
3 SIP Message Overview ................................ 14
4 Request ............................................. 15
4.1 Request-Line ........................................ 16
4.1.1 Methods ............................................. 17
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Internet Draft SIP July 31, 1997
4.1.2 Request-URI ......................................... 18
4.1.3 SIP Version ......................................... 18
5 Response ............................................ 18
5.1 Status-Line ......................................... 19
5.1.1 Status Codes and Reason Phrases ..................... 19
6 Header Field Definitions ............................ 20
6.1 General Header Fields ............................... 22
6.2 Entity Header Fields ................................ 22
6.3 Request Header Fields ............................... 22
6.4 Response Header Fields .............................. 22
6.5 Header Field Format ................................. 23
6.6 Accept .............................................. 23
6.7 Accept-Language ..................................... 24
6.8 Allow ............................................... 24
6.9 Authorization ....................................... 24
6.10 Authentication ...................................... 24
6.11 Call-ID ............................................. 24
6.12 Content-Length ...................................... 25
6.13 Content-Type ........................................ 25
6.14 Date ................................................ 26
6.15 Expires ............................................. 26
6.16 From ................................................ 27
6.17 Location ............................................ 27
6.18 Organization ........................................ 29
6.19 PEP ................................................. 29
6.20 Priority ............................................ 29
6.21 Proxy-Authenticate .................................. 29
6.22 Proxy-Authorization ................................. 29
6.23 Public .............................................. 30
6.24 Reach ............................................... 30
6.25 Retry-After ......................................... 30
6.26 Sequence ............................................ 31
6.27 Server .............................................. 31
6.28 Subject ............................................. 31
6.29 To .................................................. 32
6.30 User-Agent .......................................... 32
6.31 Via ................................................. 32
6.32 Warning ............................................. 33
6.33 WWW-Authenticate .................................... 34
7 Status Code Definitions ............................. 34
7.1 Informational 1xx ................................... 35
7.1.1 100 Trying .......................................... 35
7.1.2 180 Ringing ......................................... 35
7.2 Successful 2xx ...................................... 35
7.2.1 200 OK .............................................. 35
7.3 Redirection 3xx ..................................... 35
7.3.1 300 Multiple Choices ................................ 35
7.3.2 301 Moved Permanently ............................... 36
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7.3.3 302 Moved Temporarily ............................... 36
7.3.4 380 Alternative Service ............................. 36
7.4 Request Failure 4xx ................................. 36
7.4.1 400 Bad Request ..................................... 36
7.4.2 401 Unauthorized .................................... 37
7.4.3 402 Payment Required ................................ 37
7.4.4 403 Forbidden ....................................... 37
7.4.5 404 Not Found ....................................... 37
7.4.6 405 Method Not Allowed .............................. 37
7.4.7 407 Proxy Authentication Required ................... 37
7.4.8 408 Request Timeout ................................. 37
7.4.9 420 Bad Extension ................................... 37
7.4.10 480 Temporarily Unavailable ......................... 38
7.5 Server Failure 5xx .................................. 38
7.5.1 500 Server Internal Error ........................... 38
7.5.2 501 Not implemented ................................. 38
7.5.3 502 Bad Gateway ..................................... 38
7.5.4 503 Service Unavailable ............................. 38
7.5.5 504 Gateway Timeout ................................. 39
7.6 Global Failures ..................................... 39
7.6.1 600 Busy ............................................ 39
7.6.2 603 Decline ......................................... 39
7.6.3 604 Does not exist anywhere ......................... 39
7.6.4 606 Not Acceptable .................................. 39
8 SIP Message Body .................................... 40
8.1 Body Inclusion ...................................... 40
8.2 Message Body Length ................................. 40
9 Examples ............................................ 41
9.1 Invitation .......................................... 41
9.1.1 Request ............................................. 41
9.1.2 Reply ............................................... 42
9.1.3 Aborting a Call ..................................... 43
9.1.4 Redirects ........................................... 44
9.1.5 Alternative Services ................................ 44
9.1.6 Negotiation ......................................... 45
9.2 OPTIONS Request ..................................... 46
10 Compact Form ........................................ 47
11 SIP Transport ....................................... 48
11.1 Achieving Reliability For UDP Transport ............. 48
11.1.1 General Operation ................................... 48
11.1.2 INVITE .............................................. 49
11.2 Connection Management for TCP ....................... 50
12 Behavior of SIP Servers ............................. 50
12.1 Redirect Server ..................................... 53
12.2 Proxies Issuing Single Unicast Requests ............. 53
12.2.1 UDP--UDP Proxy Server ............................... 53
12.2.2 UDP--TCP Proxy Server ............................... 53
12.3 Proxy Server Issuing Several Requests ............... 54
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13 Security Considerations ............................. 56
13.1 Confidentiality ..................................... 56
13.2 Integrity ........................................... 56
13.3 Access Control ...................................... 56
13.4 Privacy ............................................. 56
A Summary of Augmented BNF ............................ 57
B Open Issues ......................................... 60
C Acknowledgments ..................................... 60
D Authors' Addresses .................................. 61
E Bibliography ........................................ 61
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