http://stupid.domain.name/ietf/

One document matched: draft-ietf-mmusic-sip-04.txt

Differences from draft-ietf-mmusic-sip-03.txt

Internet Engineering Task Force MMUSIC WG
Internet Draft Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-04.txt ISI/Columbia U./Caltech
November 11, 1997
Expires: April 1, 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 November 11, 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 invite
a person to both unicast and multicast sessions; the initiator does
not necessarily have to be a member of the session it is inviting to.
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 [1].)

SIP can be used to initiate sessions as well as invite members to
sessions that have been advertised and established by other means.
(Sessions may be advertised using multicast protocols such as SAP
[2], electronic mail, news groups, web pages or directories (LDAP),
among others.)

SIP transparently supports name mapping and redirection services,
allowing the implementation of ISDN and Intelligent Network telephony
subscriber services. Section 14 discusses these services in detail.

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')." [3]. 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 five facets of establishing and
terminating multimedia communications:

User location: determination of the end system to be used for
communication;

User capabilities: determination of the media and media parameters to
be used;

User availability: determination of the willingness of the called
party to engage in communications;

Handley/Schulzrinne/Schooler [Page 2]

Internet Draft SIP November 11, 1997

Call setup: "ringing", establishment of call parameters at both
called and calling party;

Call handling: including transfer and termination of calls.

SIP may also be used in conjunction with other call setup and
signaling protocols. In that mode, an end system uses SIP protocol
exchanges to determine the appropriate end system address and
protocol from a given address that is protocol-independent. For
example, SIP may be used to determine that the party may be reached
via H.323, obtain the H.245 gateway and user address and then use
H.225.0 to establish the call [4]. In another example, it may be used
to determine that the callee is reachable via the public switched
telephone network (PSTN) and indicate the phone number to be called,
possibly suggesting an Internet-to-PSTN gateway to be used.

SIP can also initiate multi-party calls using a multipoint control
unit (MCU) or fully-meshed interconnection instead of multicast.

Internet telephony gateways that connect PSTN parties may also use
SIP to set up calls between them.

SIP does not offer conference control services such as floor control
or voting and does not prescribe how a conference is to be managed,
but SIP can be used to introduce conference control protocols.

SIP does not allocate multicast addresses, leaving this functionality
to protocols such as SAP [2].

SIP can invite users to sessions 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
to the agent performing session invitation.

SIP is designed as part of the overall IETF multimedia data and
control architecture [5] currently incorporating protocols such as
RSVP [6] for reserving network resources, the real-time transport
protocol (RTP) [7] for transporting real-time data and providing QOS
feedback, the real-time streaming protocol (RTSP) [8] for controlling
delivery of streaming media, the session announcement protocol (SAP)
[2] for advertising multimedia sessions via multicast and the session
description protocol (SDP) [9] for describing multimedia sessions,
but the functionality and operation of SIP does not depend on any of
these protocols.

1.2 Terminology

Handley/Schulzrinne/Schooler [Page 3]

Internet Draft SIP November 11, 1997

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 [10]
and indicate requirement levels for compliant SIP implementations.

1.3 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) [11]. The following terms have special
significance for SIP.

Call: A call consists of a single invitation attempt from a single
user. A SIP call is identified by a globally unique call-id
(Section 6.12. Thus, if a user is, for example, invited to the
same multicast session by several people, each of these
invitations will be a unique call. A point-to-point Internet
telephony conversation maps into a single SIP call. In a MCU-
based conference, each participant uses a separate call to
invite himself to the MCU.

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 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. A successful SIP invitation consists
of two transactions: an INVITE request followed by a ACK
request.

Invitee, invited user, called party: The person or service that the
calling party is trying to invite to a conference.

Location server: See location service

Location service: A service used by a SIP redirect or proxy server to
obtain information about a callee's possible location(s).
Location services are offered by location servers. Location
servers may be co-located with a SIP server, but the manner in
which a SIP server requests location services is beyond the

Handley/Schulzrinne/Schooler [Page 4]

Internet Draft SIP November 11, 1997

scope of the document.

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. Unlike a user agent server, it
does not accept calls.

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." [9] (Note: a session as defined here may comprise one
or more RTP sessions.) Since the word session is used
differently by protocols relevant to SIP, this document avoids
the term altogether.

(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 6.12). 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 SIP 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

Handley/Schulzrinne/Schooler [Page 5]

Internet Draft SIP November 11, 1997

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. The properties of the different SIP server types are
summarized in Table 1.

property redirect proxy user agent
server server server
_______________________________________________________
also acts as client no yes no
return 1xx status yes yes yes
return 2xx status no yes yes
return 3xx status yes yes yes
return 4xx status yes yes yes
return 5xx status yes yes yes
return 6xx status no yes yes
insert Via header no yes no
accept ACK no yes yes

Table 1: Properties of the different SIP server types

1.4 Summary of SIP Operation

This section explains the basic protocol functionality and operation.
Callers and callees are identified by SIP addresses, described in
Section 1.4.1. When making a SIP call, a caller first locates the
appropriate server (Section 1.4.2) and then sends a SIP request
(Section 1.4.3). The most common SIP operation is the invitation
(Section 1.4.4). Instead of directly reaching the intended callee, a
SIP request may be redirected or trigger a chain of new SIP requests
by proxies (Section 1.4.5). Users can register with SIP servers
(Section 4.2.5).

1.4.1 SIP Addressing

SIP addresses contain a user and host part. 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]
root@193.175.132.42

Handley/Schulzrinne/Schooler [Page 6]

Internet Draft SIP November 11, 1997

A user's 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.

SIP addresses may contain a moniker (such as a civil name) or user
name and domain name that may not map into a single host. [1]

SIP 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 [12], SRV [13] or A [14] 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 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 SIP 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.,
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.

When used within SIP, SIP addresses are written as SIP URLs (Section
sec:url), e.g., sip://info@ietf.org as SIP requests and responses may
also contain non-SIP addresses, e.g., telephone numbers.

1.4.2 Locating a SIP Server
_________________________

[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 November 11, 1997

A SIP client MUST follow the following steps to resolve the host part
of a callee address. If a client only supports TCP or UDP, but not
both, the respective address type is omitted. If the SIP address
contains a port number, that number is to be used, otherwise, the the
default port number. The default port number for UDP and TCP is the
same.

1. If the SIP address is a numeric IP address, contact a SIP
server at that address.

2. If the SIP address does not contain a port number and if
there is a SRV DNS resource record [13] of type sip.udp,
contact the listed SIP servers in the order of the
preference values contained in those resource records,
using UDP as a transport protocol at the port number listed
in the DNS resource record. [TBD: What if the SIP URL
contains a port number?]

3. If the SIP address does not contain a port number and if
there is a SRV DNS resource record [13] of type sip.tcp,
contact the listed SIP servers in the order of the
preference value contained in those resource records, using
TCP as a transport protocol at the port number listed in
the DNS resource record.

4. If there is a DNS MX record [12], contact the hosts listed
in their order of preference at the default port number
(TBD). For each host listed, first try to contact the SIP
server using UDP, then TCP.

5. 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.

6. 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 for a
particular address and retry that host address for the next call. If
the client does not find a SIP server at the cached address, it MUST

Handley/Schulzrinne/Schooler [Page 8]

Internet Draft SIP November 11, 1997

start the search at the beginning of the sequence.

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 [15].

1.4.3 SIP Transaction

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. A request (and its retransmissions)
together with the responses triggered by that request make up a SIP
transaction.

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.
(Implementation note: use recvfrom() to obtain the source address and
port of the 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.

1.4.4 SIP Invitation

A successful SIP invitation consists of two requests, INVITE
followed by ACK. The INVITE (Section 4.2.1) request asks the callee
to join a particular conference or establish a two-party
conversation. After the callee has agreed to participate in the call,
the caller confirms that it has received that response by sending an
ACK (Section 4.2.2) request. If the call is rejected or otherwise
unsuccessful, the caller sends a BYE request instead of an ACK.

Handley/Schulzrinne/Schooler [Page 9]

Internet Draft SIP November 11, 1997

The INVITE request typically contains a session description, for
example written in SDP format, that provides the called party with
enough information to join the session. For multicast sessions, the
session description enumerates the media types and formats that may
be distributed to that session. For unicast session, the session
description enumerates the media types and formats that the caller is
willing to receive and where it wishes the media data to be sent. In
either case, if the callee wishes to accept the call, it responds to
the invitation by returning a similar description listing the media
it wishes to receive. For a multicast session, the callee should only
return a session description if it is unable to receive the media
indicated in the caller's description. The caller may ignore the
session description returned or use it to change the global session
description.

The session description may refer to a session start time in the
future. Actual transmission of data SHOULD not start until the time
indicated in the session description.

The protocol exchanges for the INVITE method are shown in Fig. 1 for
a proxy server and in Fig. 2 for a redirect server. The proxy server
accepts the INVITE request (step 1), contacts the location service
with all or parts of the address (step 2) and obtains a more precise
location (step 3). The proxy server then issues a SIP INVITE request
to the address(es) returned by the location service (step 4). The
user agent server alerts the user (step 5) and returns a success
indication to the proxy server (step 6). The proxy server then
returns the success result to the original caller (step 7). The
receipt of this message is confirmed by the caller using an ACK
message, which is forwarded to the callee (steps 8 and 9), with a
response returned (steps 10 and 11). All requests have the same
Call-ID.

The redirect server accepts the INVITE request (step 1), contacts
the location service as before (steps 2 and 3) and, instead of
contacting the newly found address itself, returns the address to the
caller (step 4). The caller issues a new request, with a new call-ID,
to the address returned by the first server (step 6). In the example,
the call succeeds (step 7). The caller and callee complete the
handshanke with an ACK (steps 8 and 9).

The next section discusses what happens if the location service
returns more than one possible alternative.

1.4.5 Locating a User

Handley/Schulzrinne/Schooler [Page 10]

Internet Draft SIP November 11, 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

A callee may move between a number of different end systems over
time. These locations can be dynamically registered with the SIP
server (Section 4.2.5) or a location server, typically for a single
administrative domain, or a location server may use other protocols,
such as finger [16], rwho, multicast-based protocols or operating-
system dependent mechanism to actively determine the end system where
a user might be 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 [16] or LDAP [17]) 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 6.18). A
SIP proxy server can sequentially or in parallel try the addresses

Handley/Schulzrinne/Schooler [Page 11]

Internet Draft SIP November 11, 1997

====> SIP request
----> non-SIP protocols

Figure 2: Example of SIP redirect server

until the call is successful (2xx response) or the callee has
declined the call (60x response). With sequential attempts, a proxy
server can implement an "anycast" service.

If a proxy server forwards a SIP request, it MUST add itself to the
end of the list of forwarders noted in the Via (Section 6.33)
headers. 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. A proxy server MUST check that it does not generate a
request to a host listed in the Via list. (Note: If a host has

Handley/Schulzrinne/Schooler [Page 12]

Internet Draft SIP November 11, 1997

several names or network addresses, this may not always work. Thus,
each host also checks if it is part of the Via list.)

A SIP invitation may traverse more than one SIP proxy server. If one
of these "forks" the request, i.e., issues more than one request in
response to receiving the invitation request, it is possible that a
client is reached, independently, by more than one copy of the
invitation request. Each of these copies bears the same Call-ID.
The user agent MUST return the appropriate status response, but
SHOULD NOT alert the user.

As discussed in Section 1.4.1, 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, redirections and
call status individually. For example, some may be contacted
successfully, while one of the group may be reachable under a
different address.)

1.4.6 Changing an Existing Session

In some circumstances, it may be necessary to change the parameters
of an existing session. For example, two parties may have been
conversing and then want to add a third party, switching to multicast
for efficiency. One of the participants invites the third party with
the new multicast address and simultaneously sends an INVITE to the
second party, with the new multicast session description, but the old
call identifier.

1.4.7 Registration Services

The REGISTER and UNREGISTER requests allow a client to let a proxy
or redirect server know which address it may be reached under. A
client may also use it to install call handling features at the
server.

1.5 Protocol Properties

1.5.1 Minimal State

A single conference session or call may involve one or more SIP
request-response transactions. Proxy server do not have to keep state
for a particular call, however, they maintain state for a single SIP
transaction, as discussed in Section 12.

Handley/Schulzrinne/Schooler [Page 13]

Internet Draft SIP November 11, 1997

For efficiency, a server may cache the results of location service
requests.

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 TCP connection state for each outstanding request,
and to use multicast. Routers can more readily snoop SIP UDP
packets. 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) [1].

When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing conference.
Different SIP requests for the same SIP call 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 used for initiating multimedia
conferences rather than delivering media data, it is believed that
the additional overhead of using a text-based protocol is not
significant.

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 also be embedded in web pages or other hyperlinks to
indicate that a user or service may be called.

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 defined to allow setting SIP request-header fields and
the SIP message-body. (This is similar to the mailto: URL.)

A SIP URL follows the guidelines of RFC 1630 [18,19] and takes the
following form:

SIP-URL = short-sip-url | full-sip-url

Handley/Schulzrinne/Schooler [Page 14]

Internet Draft SIP November 11, 1997

full-sip-url = "sip://" ( user | phone ) [ ":" password ]
"@" [ host | nhost ]
url-parameters [ headers ]
short-sip-url = ( user | phone) [ ":" password ]
"@" [ host | nhost ] : port
user = ; defined in RFC 1738 [20]
phone = "+" DIGIT *( DIGIT | "-" | "." )
host = ; defined in RFC 1738
nhost = "[" hostnumber "]" | hostnumber
hostnumber = digits "." digits "." digits "." digits
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 )
header = hname "=" hvalue
hname = *urlc
hvalue = *urlc
urlc = ; defined in [19]
digits = 1*digit

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.

The mailto: URL and RFC 822 email addresses require that numeric
host addresses ("host numbers") are enclosed in square brackets
(presumably, since host names might be numeric), while host numbers
without brackets are used for all other URLs. The SIP URL allows both
forms.

The password parameter can be used for a basic authentication
mechanism that takes the place of an unlisted telephone number. Also,
for Internet telephony gateways, it may serve as a PIN. Including
just the password in the URL is more convenient than including a
whole authentication header. This approach may be reasonably secure

Handley/Schulzrinne/Schooler [Page 15]

Internet Draft SIP November 11, 1997

if the URL is part of a secure web page. Unless the SIP transaction
is carried over a secure network connection, this carries the same
security risks as all URL-based passwords and should only be used
when security requirements are low. In almost all circumstances, use
of the Authorization (Section 6.10) header is preferred.

The phone identifier is to be used when connecting to a telephony
gateway. The phone number follows the rules for international numbers
in ITU Recommendation E.123, with only numbers and hyphens allowed.

Examples of short and full-form SIP URLs 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
sip://+1-212-555-1212:1234@gateway.com
sip://alice@[10.1.2.3]
sip://alice@10.1.2.3

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. The correct behavior when an unknown scheme 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:

Handley/Schulzrinne/Schooler [Page 16]

Internet Draft SIP November 11, 1997

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 [11]. In addition, we describe SIP in
both prose and an augmented Backus-Naur form (BNF) [H2.1] described
in detail in [21].

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 [22]. Recent studies
[23] 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

Both Request (section 4) and Response (section 5) messages use the
generic message format of RFC 822 [24] 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

Handley/Schulzrinne/Schooler [Page 17]

Internet Draft SIP November 11, 1997

headers. These components are described in detail in the upcoming
sections.

generic-message = start-line
*message-header
CRLF
[ message-body ]

start-line = Request-Line | Section 4.1
Status-Line Section 5.1

message-header = *( general-header
| request-header
| entity-header )

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:

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.

Handley/Schulzrinne/Schooler [Page 18]

Internet Draft SIP November 11, 1997

Table 2: SIP headers

Request-Line ___ Method SP Request-URI SP SIP-Version CRLF

4.2 Methods

The methods are defined below. Methods that are not supported by a
proxy or redirect server SHOULD be treated by that server as if they
were an INVITE method and forwarded accordingly.

Methods that are not supported by a user agent server should cause a
"501 Not Implemented" response to be returned (Section 7).

method = "INVITE" | "ACK" | "OPTIONS"

Handley/Schulzrinne/Schooler [Page 19]

Internet Draft SIP November 11, 1997

| "BYE" | "REGISTER" | "UNREGISTER"

4.2.1 INVITE

The INVITE method indicates that the user or service is being
invited to participate in a session. The message body contains a
description of the session the callee is being invited to. For two-
party calls, the caller indicates the type of media it is able to
receive as well as their parameters such as network destination. If
the session description format allows this, it may also indicate
"send-only" media. A success response indicates in its message body
which media the callee wishes to receive.

A server MAY automatically respond to an invitation for a conference
the user is already participating in, identified either by the SIP
Call-ID or a globally unique identifier within the session
description, with a "200 OK" response.

A user agent MUST check any version identifiers in the session
description to see if it has changed. If the version number has
changed, the user agent server MUST adjust the session parameters
accordingly, possibly after asking the user for confirmation.
(Versioning of the session description may be used to accomodate the
capabilities of new arrivals to a conference or change from a unicast
to a multicast conference.)

This method MUST be supported by a SIP server.

4.2.2 ACK

ACK request confirms that the client has received a final response to
an INVITE request. See Section 11 for details. This method MUST be
supported by a SIP server and client.

4.2.3 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.

4.2.4 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

Handley/Schulzrinne/Schooler [Page 20]

Internet Draft SIP November 11, 1997

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.

Support of this method is OPTIONAL.

4.2.5 REGISTER

A client uses the REGISTER method to register the address listed in
the request line to a SIP server. The host part of the request-URI
SHOULD correspond to (one of the aliases of) name of the server or to
the domain that it represents, if location-independent. After
registration, the server MAY forward incoming SIP requests to the the
network source address and port from the registration request. A
server SHOULD silently drop the registration after one hour, unless
refreshed by the client. A client may request and a server may
indicate or lower or higher refresh interval and indicate the
interval through the Expires header (Section 6.16). A single address
(if host-independent) may be registered from several different
clients.

If the request contains a Location header, requests for the
request-URI will be directed to the address(es) given.

Support of this method is OPTIONAL.

Beyond its use as a simple location service, this method is
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.

4.2.6 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.

4.3 Request-URI

The Request-URI field is a SIP URL as described in Section 2 or a

Handley/Schulzrinne/Schooler [Page 21]

Internet Draft SIP November 11, 1997

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.

4.3.1 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".

4.4 Option Tags

Option tags are unique identifiers used to designate new options in
SIP. These tags are used in Require (Section 6.24) and Unsupported
(Section 6.29) fields.

Syntax:

option-tag ___ 1*OCTET ; LWS must be URL-escaped

The creator of a new SIP option should either prefix the option with
a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name
for a feature whose inventor can be reached at "foo.com"), or
register the new option with the Internet Assigned Numbers Authority
(IANA).

4.4.1 Registering New Option Tags with IANA

When registering a new SIP option, the following information should
be provided:

oName and description of option. The name may be of any length,
but SHOULD be no more than twenty characters long. The name
should not contain any spaces, control characters or periods.

oIndication of who has change control over the option (for
example, IETF, ISO, ITU-T, other international standardization

Handley/Schulzrinne/Schooler [Page 22]

Internet Draft SIP November 11, 1997

bodies, a consortium or a particular company or group of
companies);

oA reference to a further description, if available, for example
(in order of preference) an RFC, a published paper, a patent
filing, a technical report, documented source code or a
computer manual;

oFor proprietary options, contact information (postal and email
address);

Borrowed from RTSP and the RTP AVP.

5 Response

After receiving and interpreting a request message, the recipient
responds with a SIP response message. The response message format is
shown below:

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 4.3.1) 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

Handley/Schulzrinne/Schooler [Page 23]

Internet Draft SIP November 11, 1997

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;

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.

Handley/Schulzrinne/Schooler [Page 24]

Internet Draft SIP November 11, 1997

Informational = "100" ; Trying
| "180" ; Ringing
| "181" ; Queued

Success = "200" ; OK

Figure 3: Informational and success status codes

Figure 4: Redirection status codes

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

Handley/Schulzrinne/Schooler [Page 25]

Internet Draft SIP November 11, 1997

Figure 5: Client error status codes

Figure 6: Server error status codes

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.

The header fields required, optional and not applicable for each

Handley/Schulzrinne/Schooler [Page 26]

Internet Draft SIP November 11, 1997

Global-Failure | "600" ; Busy
| "603" ; Decline
| "604" ; Does not exist anywhere
| "606" ; Not Acceptable

Figure 7: Global failure status Codes

method are listed in Table 3. 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).

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
also defined in Section 10 for use over UDP when the request has to
fit into a single packet and size is an issue.

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

Handley/Schulzrinne/Schooler [Page 27]

Internet Draft SIP November 11, 1997

type ACK BYE INV OPT REG UNR
_________________________________________________________________
Accept R o - o o o o
Accept-Language R o o o o o o
Allow 405 o o o o o o
Also R - - o - - -
Authorization R o o o o o o
Call-Disposition R - o o - - -
Call-ID g m m m o - -
Content-Length g - - * * - -
Content-Type g - - * * - -
CSeq g o o o o o o
Date g o o o o o o
Expires g - - o o o -
From R m m m m o o
Location R - o - - o -
Location r - - o o - -
Organization R - - o o - -
Proxy-Authenticate R o o o o o o
Proxy-Authorization R o o o o o o
Priority R - - o - - -
Public r - - - o - -
Require R o o o o o o
Retry-After 600,603 - - o - - -
Server r o o o o o o
Subject R - - o - - -
Timestamp g o o o o o o
To g m m m m m m
Unsupported r o o o o o o
User-Agent R o o o o o o
Via g m m m m m m
Warning r o o o o o o
WWW-Authenticate 401 o o o o o o

Table 3: Summary of header fields. "o": optional, "m": mandatory, "-
": not applicable, "R': request header, "r": response header, "g":
general header, "*": needed if message body is not empty. A numeric
value in the "type" column indicates the status code the header field
is used with.

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

Handley/Schulzrinne/Schooler [Page 28]

Internet Draft SIP November 11, 1997

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
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 [24].

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,

Handley/Schulzrinne/Schooler [Page 29]

Internet Draft SIP November 11, 1997

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] for syntax. This request header field is used only with
the OPTIONS and INVITE request methods to indicate what description
formats are acceptable in the response.

Example:

Accept: application/sdp;level=1, application/x-private

6.7 Accept-Language

See [H14.4] for syntax. 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 Also

The Also request header advises the callee to send invitations to
the addresses listed. This supports third-party call initiation
(Section 13).

Handley/Schulzrinne/Schooler [Page 30]

Internet Draft SIP November 11, 1997

Also ___ "Also" ":" 1#( SIP-URL ) [ comment ]

Example:

Also: sip://jones@foo.com, sip://mueller@bar.edu

6.10 Authorization

See [H14.8].

6.11 Call-Disposition

The Call-Disposition request header field allows the client to
indicate how the server is to handle the call. The following options
can be used singly or in combination:

all: If the user part of the SIP request address identifies a group
rather than an individual, the " all" feature indicates that all
members of the group should be alerted rather than the default
of locating the first available individual from that group.
Section 1.4.1 describes the behavior of proxy servers when
resolving group aliases.

do-not-forward: 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.)

queue: If the called party is temporarily unreachable, e.g., because
it is in another call, the caller can indicate that it wants to
have its call queued rather than rejected immediately. If the
call is queued, the server returns "181 Queued" (see Section
7.1.3). A pending call be terminated by a BYE request (Section
4.2.4).

Call-Disposition ___ "Call-Disposition" ":" 1#( "all" | "do-not-forward"
| "queue" )

Example:

Call-Disposition: all, do-not-forward, queue

Handley/Schulzrinne/Schooler [Page 31]

Internet Draft SIP November 11, 1997

HS: This header is experimental. The name is based on the
SMTP Content-Disposition header.

6.12 Call-ID

The Call-ID general header uniquely identifies a particular
invitation. Note that a single multimedia conference may give rise to
several calls with different Call-IDs, e.g., if a user invites
several different people. Since the Call-ID is unique for each
caller, a user may invited to the same conference using several
different Call-IDs. If desired, it must use identifiers within the
session description to detect this duplication. Calls to different
callee MUST always use different Call-IDs unless they are the result
of a 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

6.13 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

Handley/Schulzrinne/Schooler [Page 32]

Internet Draft SIP November 11, 1997

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.

6.14 Content-Type

The Content-Type entity-header field indicates the media type of the
message-body sent to the recipient.

Content-Type ___ "Content-Type" ":" media-type

An example of the field is

Content-Type: application/sdp

6.15 Date

General header field. See [H14.19].

The Date header field is useful for simple devices without
their own clock.

6.16 Expires

The Expires entity-header field gives the date and time after which
the message content expires.

This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a request and response-header
field and allows the client to indicate how long the registration
should be valid; the server uses it to indicate when the client has
to re-register. The server's choice overrides that of the client. The
server MAY choose a shorter time interval than that requested by the
client, but SHOULD not choose a longer one.

Handley/Schulzrinne/Schooler [Page 33]

Internet Draft SIP November 11, 1997

For INVITE, it is a request and response-header field. In a request,
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.) In a 302
response, a server can advise the client of the maximal duration of
the redirection.

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

6.17 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 1.4.1
address to allow a gateway to relay a call into a SIP request and
still produce an appropriate From field.

From ___ ( "From" | "f" ) ":" *1( ( SIP-URL | URL ) [ comment ] )

Examples:

Handley/Schulzrinne/Schooler [Page 34]

Internet Draft SIP November 11, 1997

From: agb@bell-telephone.com (A. G. Bell)
From: +12125551212@server.phone2net.com

6.18 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 URL giving the
new location or username to try, or may simply specify additional
transport parameters. A "301 Moved Permanently" or "302 Moved
Temporarily" response SHOULD contain a Location field containing the
URL giving a new address to try. A 301 or 302 response may also give
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 SIP 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
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).

A Location response header may contain any suitable URL indicating
where the called party may be reached, not limited to SIP URLs. For
example, it may contain a phone or fax URL [25], a mailto: URL [26]
or irc.

The following parameters are defined:

q: The qvalue indicates the relative preference among the locations
given. qvalue values are decimal numbers from 0.0 to 1.0, with
higher values indicating higher preference.

class: The class parameter whether this terminal is found in a
residential or business setting. (A caller may defer a personal
call if only a business line is available, for example.)

description: The description field further describes, as text, the
terminal. It is expected that the user interface will render
this text.

duplex: The duplex parameter lists whether the terminal can
simultaneously send and receive ("full"), alternate between
sending and receiving ("half"), can only receive ("receive-
only") or only send ("send-only"). Typically, a caller will
prefer a full-duplex terminal over a half-duplex terminal and
these over receive-only or send-only terminals.

Handley/Schulzrinne/Schooler [Page 35]

Internet Draft SIP November 11, 1997

features: The feature list enumerates additional features of this
terminal. Values for this field are for further study.

language: The language parameter lists, in order of preference, the
languages spoken by the person answering. This feature may be
used to have a caller automatically select the appropriate
attendant or customer service representative, without having to
declare its own language skills.

media: The media tag lists the media types supported by the terminal.
Currently, the names defined in SDP may be used [9]: "audio",
"video", "whiteboard", "text" and "data".

mobility: The mobility parameter indicates if the terminal is fixed
or mobile. In some locales, this may affect voice quality or
charges.

priority: The priority tag indicates the minimum priority level this
terminal is to be used for. It can be used for automatically
restricting the choice of terminals available to the user.

service: The service tag describes what service is being provided by
the terminal.

Handley/Schulzrinne/Schooler [Page 36]

Internet Draft SIP November 11, 1997

Examples:

Location: sip://watson@worcester.bell-telephone.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
Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
duplex=send-only;media=text; q=0.1; priority=urgent;
description="For emergencies only"
Location: mailto:watson@bell-telephone.com
Location: http://www.bell-telephone.com/~watson

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
is meant for indicating general communication capability, sufficient
for the caller to choose an appropriate address.

6.19 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.

6.20 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:

Handley/Schulzrinne/Schooler [Page 37]

Internet Draft SIP November 11, 1997

Subject: A tornado is heading our way!
Priority: urgent

6.21 Proxy-Authenticate

See [H14.33].

6.22 Proxy-Authorization

See [H14.34].

6.23 Public

See [H14.35].

6.24 Require

The Require header is used by clients to query the server about
options that it may or may not support. The server MUST respond to
this header by returning status code "420 Bad Extension" and list
those options it does not understand in the Unsupported header.

Require ___ "Require" ":" 1#option-tag

Example:

C->S: INVITE sip:watson@bell-telephone.com SIP/2.0
Require: com.example.billing
Payment: sheep_skins, conch_shells

S->C: SIP/2.0 420 Bad Extension
Unsupported: com.example.billing

This is to make sure that the client-server interaction will proceed
optimally when all options are understood by both sides, and only
slow down if options are not understood (as in the example above).
For a well-matched client-server pair, the interaction proceeds
quickly, saving a round-trip often required by negotiation
mechanisms. In addition, it also removes ambiguity when the client
requires features that the server does not understand.

Handley/Schulzrinne/Schooler [Page 38]

Internet Draft SIP November 11, 1997

We explored using the W3C's PEP proposal for this
functionality. However, Require, Proxy-Require, and
Unsupported allow the addition of extensions with far less
complexity.

This field roughly corresponds to the PEP field in the PEP draft.

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", "600 Busy", "603 Decline" 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. An optional comment can be used to
indicate additional information about the time of callback. An
optional duration parameter indicates how long the called party will
be reachable starting at the initial time of availability.

Retry-After ___ "Retry-After" ":" ( HTTP-date | delta-seconds )
[ comment ] [ ";duration" "=" delta-seconds

Examples of its use are

Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
Retry-After: Mon, 1 Jan 9999 00:00:00 GMT
(Dear John: Don't call me back, ever)
Retry-After: Fri, 26 Sep 1997 21:00:00 GMD;duration=3600
Retry-After: 120

In the third example, the callee is reachable for one hour starting
at 21:00 GMT. In the last example, the delay is 2 minutes.

6.26 CSeq

The CSeq (command 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 CSeq
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; the sequence space MAY NOT be contiguous. Retransmissions of

Handley/Schulzrinne/Schooler [Page 39]

Internet Draft SIP November 11, 1997

the same request carry the same sequence number. A server responding
to a request containing a sequence number MUST echo the sequence
number back in the response. The ACK request MUST contain the same
CSeq value as the INVITE request that it refers to.

CSeq = "CSeq" ":" 1*DIGIT

CSeq header fields are NOT needed for SIP requests using the INVITE
or OPTIONS methods but may be needed for future methods.

Example:

CSeq: 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!

6.29 Unsupported

The Unsupported response header lists the features not supported by
the server.

Handley/Schulzrinne/Schooler [Page 40]

Internet Draft SIP November 11, 1997

See Section 6.24 for a usage example and motivation.

6.30 Timestamp

The timestamp general header describes when the client sent the
request to the server. The value of the timestamp is of significance
only to the client and may use any timescale. The server MUST echo
the exact same value and MAY, if it has accurate information about
this, add a floating point number indicating the number of seconds
that has elapsed since it has received the request. The timestamp is
used by the client to compute the round-trip time to the server so
that it can adjust the timeout value for retransmissions.

Timestamp ___ "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay ___ *(DIGIT) [ "." *(DIGIT) ]

6.31 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)

6.32 User-Agent

See [H14.42].

6.33 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

Handley/Schulzrinne/Schooler [Page 41]

Internet Draft SIP November 11, 1997

routing situations.

The client originating the request MUST insert a Via field
containing its address to the request. Each subsequent proxy server
that sends the request 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.

If a proxy server receives a request which contains its own address,
it MUST respond with a "482 Loop Detected" status code. (This
prevents a malfunctioning proxy server from causing loops. Also, it
cannot be guaranteed that a proxy server can always detect that the
address returned by a location service refers to a host listed in the
Via list, as a single host may have aliases or several network
interfaces.)

In the return path, Via fields are processed by a proxy or client
according to the following rules:

oIf 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.

oIf the first Via field in a reply is a multicast address,
remove that Via field before sending to the multicast address.

These rules ensure that a proxy server only has to check the first
Via field in a reply to see if it needs processing.

The format for a Via header is:

Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] )
via-params = "ttl" "=" ttl
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.

Handley/Schulzrinne/Schooler [Page 42]

Internet Draft SIP November 11, 1997

Example:

Via: SIP/2.0/UDP first.example.com:4000

6.34 Warning

The Warning response-header field is used to carry additional
information about the status of a response. Warning headers are sent
with responses and have the following format:

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
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.

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:

oWarnings that appear early in the response take priority over
those appearing later in the response.

oWarnings in the user's preferred character set take priority
over warnings in other character sets but with identical

Handley/Schulzrinne/Schooler [Page 43]

Internet Draft SIP November 11, 1997

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)

6.35 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
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.

Handley/Schulzrinne/Schooler [Page 44]

Internet Draft SIP November 11, 1997

7.1.3 181 Queued

The called party was temporarily unavailable, but the caller
indicated via a "Call-Disposition: Queue" directive (Section 6.11) to
queue the call rather than reject it. When the callee becomes
available, it will return the appropriate final status response. The
reason phrase MAY give further details about the status of the call,
e.g., "5 calls queued; expected waiting time is 15 minutes". The
server may issue several 181 responses to update the caller about the
status of the queued call.

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
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 6.18).
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.

Handley/Schulzrinne/Schooler [Page 45]

Internet Draft SIP November 11, 1997

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
(Section 6.33) 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 (Section 6.33) path of the request as the user's new
location. The duration of the redirection can be indicated through
an Expires (Section 6.16) header.

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.

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

Handley/Schulzrinne/Schooler [Page 46]

Internet Draft SIP November 11, 1997

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
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 not terminate any searches. The reason
phrase SHOULD indicate the more precise cause as to why the callee is
unavailable. This value SHOULD be setable by the user agent.

7.4.11 481 Invalid Call-ID

Handley/Schulzrinne/Schooler [Page 47]

Internet Draft SIP November 11, 1997

The server received a BYE or ACK request with a Call-ID value it
does not recognize.

7.4.12 482 Loop Detected

The server received a request with a Via path containing itself.

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.

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

Handley/Schulzrinne/Schooler [Page 48]

Internet Draft SIP November 11, 1997

accessed in attempting to complete the request.

7.6 Global Failures 6xx

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
"606 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.

606.2 Incompatible Protocol: One or more protocols described in the
request are not available.

Handley/Schulzrinne/Schooler [Page 49]

Internet Draft SIP November 11, 1997

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.

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

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.

Handley/Schulzrinne/Schooler [Page 50]

Internet Draft SIP November 11, 1997

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.
(Note: The chunked encoding modifies the body of a message in order
to transfer it as a series of chunks, each with its own size
indicator.)

9 Examples

9.1 Invitation to Multimedia Conference

The first example invites schooler@vlsi.cs.caltech.edu to a multicast
session.

9.1.1 Request

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
CSeq: 4711
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

The Via fields list 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

Handley/Schulzrinne/Schooler [Page 51]

Internet Draft SIP November 11, 1997

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
CSeq: 4711

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.

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

Handley/Schulzrinne/Schooler [Page 52]

Internet Draft SIP November 11, 1997

To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
Location: sip://es@jove.cs.caltech.edu
CSeq: 4711

The caller confirms the invitation by sending a request to the
location named in the Location header:

C->S: ACK 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
CSeq: 4711

9.2 Two-party Call

A two-party call proceeds as above. The only difference is

For two-party Internet phone calls, the response must contain a
description of where to send data to. In the example below, Bell
calls Watson. Bell indicates that he can receive RTP audio codings 0
(PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).

C->S: INVITE watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
From: a.g.bell@bell-telephone.com (A. Bell)
To: watson@bell-telephone.com (T. A. Watson)
Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here.
Content-type: application/sdp
Content-Length: ...

v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5

S->C: SIP/2.0 200 OK
From: a.g.bell@bell-telephone.com (A. Bell)
To: watson@bell-telephone.com
Call-ID: 187602141351@worcester.bell-telephone.com
Location: sip://watson@boston.bell-telephone.com

Handley/Schulzrinne/Schooler [Page 53]

Internet Draft SIP November 11, 1997

Content-Length: ...

v=0
o=watson 4858949 4858949 IN IP4 192.1.2.3
c=IN IP4 135.180.161.25
m=audio 5004 RTP/AVP 0 3

Watson can only receive PCMU and GSM. Note that Watson's list of
codecs may or may not be a subset of the one offered by Bell, as each
party indicates the data types it is willing to receive. Watson will
send audio data to port 3456 at 135.180.144.94, Bell will send to
port 5004 at 135.180.161.25.

9.3 Aborting a Call

If the caller wants to abort a pending call, it sends a BYE request.

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

9.3.1 Redirects

Replies with status 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

Handley/Schulzrinne/Schooler [Page 54]

Internet Draft SIP November 11, 1997

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.

9.3.2 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
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

Handley/Schulzrinne/Schooler [Page 55]

Internet Draft SIP November 11, 1997

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.

9.3.3 Negotiation

An example of a "606 Not Acceptable" reply is:

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

Handley/Schulzrinne/Schooler [Page 56]

Internet Draft SIP November 11, 1997

gateway and re-invite the user.

9.4 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
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

Handley/Schulzrinne/Schooler [Page 57]

Internet Draft SIP November 11, 1997

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:

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

Handley/Schulzrinne/Schooler [Page 58]

Internet Draft SIP November 11, 1997

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.

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 between the request and a definitive
response can reach several tens of seconds. If either
caller or callee are automated servers not directly
controlled by a human being, a call attempt may be
unbounded in time.

It is undesirable to retransmit the INVITE request, as this
introduces additional network traffic. The retransmission
interval would have to be no more than about a second, since the
callee would encounter a "dead" voice path if the "200 OK"
response is lost.

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

Handley/Schulzrinne/Schooler [Page 59]

Internet Draft SIP November 11, 1997

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
ringing the user.

The second and third problems are solved by having the server
retransmit the final response at intervals of T3 (default value of T3
= 2 seconds) until it receives an ACK request for the same Call-ID
and CSeq or until it has retransmitted the final response 10 times.
The ACK 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. 8 and 9 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.

Handley/Schulzrinne/Schooler [Page 60]

Internet Draft SIP November 11, 1997

+===========+
| Initial |
+===========+
|
|
| -
| ------
| INVITE
+------v v
T1 +-----------+
------ | Calling |--------+
INVITE +-----------+ |
+------| | | |
+----------------+ | |
| | 1xx | >= 200
| | --- | ------
| | - | ACK
| | |
| +------v v v-----| |
| T2 +-----------+ 1xx |
| ------ | Ringing | --- |
| INVITE +-----------+ - |
| +------| | |-----+ |
| | |
| 2xx | |
| --- | 2xx |
| ACK | --- |
| | ACK |
+----------------+ | |
+------v | v |
xxx +-----------+ |
--- | Completed |<-------+
ACK +-----------+
+------|

event
-------
message

Figure 8: State transition diagram of client for INVITE method

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.

Handley/Schulzrinne/Schooler [Page 61]

Internet Draft SIP November 11, 1997

+===========+
| Initial |<-------------+
+===========+ |
| |
| |
| INVITE |
| ------ |
| 1xx |
+------v v |
INVITE +-----------+ |
------ | Searching | |
1xx +-----------+ |
+------| | | +---------------->+
| | |
failure | | callee picks up |
------- | | --------------- |
>= 300 | | 200 |
| | | BYE
+------v v v v-----| | ---
INVITE +-----------+ T3 | 200
------ | Answered | ------ |
status +-----------+ status |
+------| | | |-----+ |
| +---------------->+
| |
| ACK |
| --- |
| 200 |
| |
+------v v |
ACK +-----------+ |
--- | Connected | |
200 +-----------+ |
+------| | |
+-----------------+

event
-------
message

Figure 9: State transition diagram of server for INVITE method

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

Handley/Schulzrinne/Schooler [Page 62]

Internet Draft SIP November 11, 1997

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 its own
address into those requests that are caused by an incoming request.
To prevent loops, a server MUST check if its own address is already
contained in the Via header of the 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,
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.

The detailed connection behavior for UDP and TCP is described in
Section 11.

12.1 Redirect Server

A redirect server does not issue any SIP requests of its own. It can
return a response that refuses or redirects the request. After
receiving an INVITE request, a redirect server proceeds through the
following steps:

1. If the INVITE 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 refuse the call, it returns the
final response. This ends the SIP transaction.

The redirect server maintains transaction state for the whole SIP
transaction.

12.2 User Agent Server

Handley/Schulzrinne/Schooler [Page 63]

Internet Draft SIP November 11, 1997

Servers in user agents behave similarly to redirect servers, except
that they may also accept a call.

12.3 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
12.4.

The server can forward the request and any responses. It does not
have to maintain any state for the SIP transaction. Reliability is
assured by the next redirect server in the server chain.

A proxy server SHOULD cache the result of any address translations
and the response to speed forwarding. After the cache entry has been
expired, the server cannot tell whether an incoming request is
actually a retransmission of an older request, where the TCP side has
terminated. It will treat it as a new request.

12.4 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"
or "180 Ringing" 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(r, a) sends a SIP request r to address a.
await_response() waits until a response is received and returns the
response. close(a) closes the TCP connection to client with address
a. 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.,

Handley/Schulzrinne/Schooler [Page 64]

Internet Draft SIP November 11, 1997

those which are within the same domain as the proxy server. Thus, an
initial multicast request may trigger additional unicast requests.

enum {INVITE, /* request type */
ACK, OPTIONS, BYE, REGISTER, UNREGISTER} R;
int N = 0; /* number of connection attempts */
address_t address[]; /* list of addresses */
int done[]; /* address has responded */
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 */
} r;
int i;

if (multicast) {
request(R, address[0]);
} else {
N = /* number of addresses to try */
for (i = 0; i < N; i++) {
request(R, address[i]);
done[i] = 0;
}
}

while (timeleft > 0 && (heard < N || multicast)) {
r = await_response();
class = r.status / 100;

if (class >= 2) {
heard++;
for (i = 0; i < N; i++) {
if (address[i] == r.a) {
done[i] = 1;
break;
}
}
}

if (class == 2) {
best = r.status;

Handley/Schulzrinne/Schooler [Page 65]

Internet Draft SIP November 11, 1997

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, 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) {
best = r.status;
break;
}
}
/* We haven't heard anything useful from anybody. */
if (best == 1000) {
best = 404;
}
if (best/100 != 3) loc = 0;
response(best, loc, locations);

/*
* Close the other pending transactions by sending BYE.
*/
for (i = 0; i < N; i++) {
if (!done[i]) {
request(BYE, address[i]);
if (tcp) close(a);
}
}

Handley/Schulzrinne/Schooler [Page 66]

Internet Draft SIP November 11, 1997

After receiving a 2xx or 6xx response, the server SHOULD terminate
all other pending requests by sending a BYE request and closing the
TCP connection, if applicable. (Terminating pending requests is
advisable as searches consume resources. Also, INVITE requests may
"ring" on a number of workstations if the callee is currently logged
in more than once.)

[TBD: How do we cancel multicast requests? Force receivers to listen
for a 200/6xx response and hope that they don't miss one?]

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 Third-Party Call Initiation

In some circumstances, third-party call control is required, where
the calling party suggests to the called party to invite a (small)
number of other parties. Third-party call control can be used to
implement the following features:

Multipoint-control unit (MCU): Some conferences use a multipoint
control unit to mix, convert and replicate media streams. While
this solution has scaling problems, it is widely deployed in
traditional telephony and ISDN conferencing settings, as so-
called conference bridges. In a MCU-based conference, the
conference initiator or any authorized member invites a new
participant and indicate the address of the MCU in the Also
header. The invitee then contacts the MCU using the same session
description and requests to be added to the call, just like a
normal two-party call.

Telephony call initiation ("click-to-call"): A SIP INVITE request
containing two addresses in the Also header is sent to a PSTN
service node that connects these two addresses by a telephone
call.

Fully-meshed small conference: For small conferences, such as adding
a third party to a two-party call, multicast may not always be
appropriate or available. Instead, when inviting a new
participant, the caller asks the new member to call the
remaining members. TBD: Should the call-id be the same or
different? Need to distinguish between new INVITE for same call
and adding a party to a call. Include conference identifier?

Handley/Schulzrinne/Schooler [Page 67]

Internet Draft SIP November 11, 1997

TBD: How about just transferring an SDP description with multiple
addresses?

The Also: header (Section 6.9) is used to indicate a list of parties
that the callee should invite.

14 ISDN and Intelligent Network Services

SIP may be used to support a number of ISDN [27] and Intelligent
Network [28] telephony services, described below. Due to the
fundamental differences between Internet-based telephony and
conferencing as compared to public switched telephone network
(PSTN)-based services, service definitions cannot be precisely the
same. Where large differences beyond addressing and location of
implementation exist, this is indicated below. The term address
implies any SIP address. (Section 1.4.1).

Call transfer (TRA) enables a user to transfer an established (i.e.,
active) call to a third party. SIP signals this via the Location
header in the BYE (Section 4.2.4) method.

Call forwarding (CF) permits the called user to forward particular
pre-selected calls to another address. Unlike telephony, the
choice of calls to be forwarded depends on program logic
contained in any of the SIP servers and can thus be made
dependent on time-of-day, subject of call, media types, urgency
or caller identity, rather than being restricted to matching
list entries. This forwarding service encompasses:

Call forwarding busy/don't answer (CFB/CFNR, SCF-BY/DA) allows the
called user to forward particular pre-selected calls if the
called user is busy or does not answer within a set time.

Selective call forwarding (SCF) permits the user to have her incoming
calls addressed to another network destination, no matter what
the called party status is, if the calling address is included
in, or excluded from, a screening list. The user's originating
service is unaffected.

Completion of calls to busy subscriber (CCBS) allows a calling user
encountering a busy destination to be informed when the busy
destination becomes free, without having to make a new call
attempt. SIP supports services close to CCBS by allowing a
callee to indicate a more opportune time to call back (Section
6.25). Also, calling and called user agents can easily record
the URL of outcoming and incoming calls, so that a user can re-
try or return calls with a single mouse click.

Handley/Schulzrinne/Schooler [Page 68]

Internet Draft SIP November 11, 1997

Conferencing (CON) allows the user to communicate simultaneously with
multiple parties, which may also communicate among themselves.
SIP can initiate IP multicast conferences with any number of
participants, conferences where media are mixed by a conference
bridge (multipoint control unit or MCU) and, for exceptional
applications with a small number of participants, fully-meshed
conferences, where each participant sends and receives data to
all other participants.

Conference calling add-on allows a user to add and drop participants
once the conference is active.

Conference calling meet-me (MMC) allows the user to set up a
conference or multi-party call, indicating the date, time,
conference duration, conference media and other parameters. The
conference session description included in the SIP invitation
may indicate a time in the future. For multicast conferences,
participants do not have to connect using SIP at the actual time
of the conference; instead, they simply subscribe to the
multicast addresses listed in the announcement. For MCU-based
conferences, the session description may contain the address of
the MCU to be called at the time of the conference.

Destination call routing (DCR) allows customers to specify the
routing of their incoming calls to destinations according to

-time of day, day of week, etc.;

-area of call origination;

-network address of caller;

-service attributes;

-priority (e.g., from input of a PIN or password);

-charge rates applicable for the destination;

-proportional routing of traffic.

In SIP, destination call routing is implemented by proxy and redirect
servers that implement custom call handling logic, with parameters
including, but not limited to the set listed above.

Follow-me diversion (FMD) allows the service subscriber to remotely
control the redirection (diversion) of calls from his primary
network address to other locations.

Handley/Schulzrinne/Schooler [Page 69]

Internet Draft SIP November 11, 1997

In SIP, finding the current network-reachable location of a callee is
left to the location service and is outside the scope of this
specification. However, users may use the REGISTER method (Section
4.2.5) to appraise their "home" SIP server of their new location.

Originating call screening (OCS) controlls the ability of a node to
originate calls. In a fashion similar to closed user groups, a
firewall would have to be used to restrict the ability to
initiate SIP invitations outside a designated part of the
network. In many cases, gateways to the PSTN will require
appropriate authentication.

Premium rate (PRM) allows to pay back part of the call cost to the
called party, considered as an added value provider. See
discussion on billing services below.

Split charging (SPL) allows the calling and called party being each
charged for one part of the call. See discussion on billing
services below.

Universal access number (UAN) allows a subscriber with several
network addresses to be reached with a single, unique address.
The subscriber may specify which incoming calls are to be routed
to which address. SIP offers this functionality through proxies
and redirection.

Universal personal telecommunications (UPT) is a mobility service
which enables subscribers to be reached with a unique personal
telecommunication number (PTN) across multiple networks at any
network access. The PTN will be translated to an appropriate
destination address for routing based on the capabilities
subscribed to by each service subscriber. A person may have
multiple PTNs, e.g., a business and private PTN. In SIP, the
host-independent address (Section 1.4.1) of the form user@host
serves as the PTN, which is translated into one or more host-
dependent addresses.

User-defined routing (UDR) allows a subscriber to specify how
outgoing calls, from the subscriber's location, shall be routed.
SIP cannot specify routing preferences; this is presumed to be
handled by a policy-based routing protocol, source routing or
similar mechanisms.

Some telephony services can be provided by the end system, without
involvement by SIP:

Abbreviated dialing allows users to reach local subscribers without
specifying the full address (domain or host name). For SIP, the

Handley/Schulzrinne/Schooler [Page 70]

Internet Draft SIP November 11, 1997

user application completes the address to be a fully qualified
domain name.

Call waiting (CW) allows the called party to receive a notification
that another party is trying to reach her while she is busy
talking to another calling party.

For SIP-based telephony, the called party can maintain several call
presences at the same time, limited by local resources. Thus, it is
up to the called party to decide whether to accept another call. The
separation of resource reservation and call control may lead to the
situation that the called party accepts the incoming call, but that
the network or system resource allocation fails. This cannot be
completely prevented, but if the likely resource bottleneck is at the
local system, the user agent may be able to determine whether there
are sufficient resources available or roughly track its own resource
consumption.

Consultation calling (COC) allows a subscriber to place a call on
hold, in order to initiate a new call for consultation. In
systems using SIP, consultation calling can be implemented as
two separate SIP calls, possibly with the temporary release of
reserved resources for the call being put on hold.

Customized ringing (CRG) allows the subscriber to allocate a
distinctive ringing to a list of calling parties. In a SIP-based
system, this feature is offered by the user application, based
on caller identification ( From, Section 6.17) provided by the
SIP INVITE request (Section 4.2.1).

Malicious call identification (MCI) allows the service subscriber to
control the logging (making a record) of calls that received
that are of a malicious nature. In SIP, by default, all calls
identify the calling party and the SIP servers that have
forwarded the call. In addition, calls may be authenticated
using standard HTTP methods or transport-layer security. A
callee may decide only to accept calls that are authenticated.

Multiway calling (MWC) allows the user to establish multiple,
simultaneous calls with other parties. For a SIP-based end
system, the considerations for consultation calling apply.

Terminating call screening (TCS) allows the subscriber to specify
that incoming calls either be restricted or allowed, according
to a screening list and/or by time of day or other parameters.

Billing features such as account card dialing , automatic alternative
billing , credit card calling (CCC) , reverse charging , freephone

Handley/Schulzrinne/Schooler [Page 71]

Internet Draft SIP November 11, 1997

(FPH) , premium rate (PRM) and split charging are supported through
authentication. However, mechanisms for indicating billing
preferences and capabilities have not yet been specified for SIP.

Advice of charge allows the user paying for a call to be informed of
usage-based charging information. Charges incurred by reserving
resources in the network are probably best indicated by a protocol
closely affiliated with the reservation protocol. Advice of charge
when using Internet-to-PSTN gateways through SIP appears feasible,
but is for further study. Desirable facilities include indication of
charges at call setup time, during the call and at the end of the
call

Closed user groups (CUGs) that restrict members to communicate only
within the group can be implemented using firewalls and SIP proxies.

User-to-user signaling is supported within SIP through the addition
of headers, with predefined header fields such as Subject or
Organization.

Third-party signaling is optionally supported within SIP (Section
6.9). Third-party signaling can be used to indicate to callees who
else to invite to a call for MCU and fully-meshed conferences.
Third-party signaling, combined with appropriate URLs, may be used to
initiate PSTN phone calls from an Internet host.

15 Security Considerations

15.1 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.

15.2 Integrity

Unless SIP transactions are protected by lower-layer security
mechanisms such as SSL , an active attacker may be able to modify SIP
requests.

15.3 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.

Handley/Schulzrinne/Schooler [Page 72]

Internet Draft SIP November 11, 1997

15.4 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.

A Minimal Implementation

A.1 Client

All clients MUST be able to generate the INVITE and ACK requests
and MUST be able to include the Call-ID, Content-Length, Content-
Type, From and To headers. A minimal implementation MUST understand
SDP [9]. In responses, it must be able to parse the Call-ID,
Content-Length, Content-Type, Require headers. It must be able to
recognize the status code classes 1 through 6 and act accordingly.

The following capability sets build on top of a minimal
implementation:

Basic: A basic implementation SHOULD add support for the BYE method
to allow the interruption of a pending call attempt. It SHOULD
include a User-Agent header in its requests and indicate its
preferred language in the Accept-Language header.

Redirection: To support call forwarding, a client needs to be able to
understand the Location header, but only the SIP-URL part, not
the parameters.

Negotiation: A client MUST be able to request the OPTIONS method and
understand the 380 "Alternative Service" status and the Location
parameters to participate in terminal and media negotiation. It
SHOULD be able to parse the Warning response header to provide
useful feedback to the caller.

Authentication: If a client wishes to invite callees that require
caller authentication, it must be able to recognize the 401
"Unauthorized" status code, must be able to generate the
Authorization request header and understand the WWW-
Authenticate response header.

If a client wishes to use proxies that require caller authentication,
it must be able to recognize the 407 "Proxy Authentication Required"
status code, must be able to generate the Proxy-Authorization
request header and understand the Proxy-Authenticate response
header.

Handley/Schulzrinne/Schooler [Page 73]

Internet Draft SIP November 11, 1997

A.2 Server

A minimally compliant server implementation MUST understand the
INVITE, ACK and BYE requests. It MUST parse the generate, as
appropriate, the Call-ID, Content-Length, Content-Type, From,
PEP, To and Via headers. It must echo the Sequence header in the
response. It SHOULD include the Server header in its responses.

B Summary of Augmented BNF

In this specification we use the Augmented Backus-Naur Form notation
described in [21]. 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.

Handley/Schulzrinne/Schooler [Page 74]

Internet Draft SIP November 11, 1997

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.

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 in Version -04

Since version -03, the following changes have been made.

oThe introduction has been reorganized and large parts
rewritten.

oCONNECTED changed to ACK, as it applies to all responses, not
just 200.

oStatus code 181 (Queued) and Call-Disposition: Queue added.

oStatus code 481 (Invalid Call-ID) added.

oStatus code 482 (Loop Detected) added. Via description contains
motivation.

oAllow phone numbers in SIP URL for easy connection to Internet
telephony gateways.

oAdded Also header for third-party connectivity.

oWhen doing parallel searches, pending searches should be
aborted when one address was successful. The phone call may be
ringing on a number of workstations where the user is logged in
and would keep ringing.

oAdded duration parameter to Retry-After to indicate how long
the callee is likely to be reachable at the address given.

Handley/Schulzrinne/Schooler [Page 75]

Internet Draft SIP November 11, 1997

oChanged Sequence to CSeq for consistency with RTSP.

C Open Issues

Full meshes: How about just transferring an SDP description with
multiple addresses?

H.323: Detailed 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. Also,
avoids BYE race condition: Proxy doing parallel search cancels
pending search with BYE after one of the addresses responds with
200. Through another proxy, this BYE reaches the same end system
and cancels the successful call.

D Acknowledgments

We wish to thank the members of the IETF MMUSIC WG for their comments
and suggestions. Detailed comments were provided by Jonathan
Rosenberg. This work is based, inter alia, on [29,30]. Parameters of
the terminal negotiation mechanism were influenced by Scott Petrack's
CMA design.

E 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

Handley/Schulzrinne/Schooler [Page 76]

Internet Draft SIP November 11, 1997

USA
electronic mail: schooler@cs.caltech.edu

F Bibliography

[1] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
Mar. 1997. Work in progress.

[2] M. Handley, "SAP: Session announcement protocol," Internet Draft,
Internet Engineering Task Force, Nov. 1996. Work in progress.

[3] R. Pandya, "Emerging mobile and personal communication systems,"
IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.

[4] P. Lantz, "Usage of H.323 on the Internet," Internet Draft,
Internet Engineering Task Force, Feb. 1997. Work in progress.

[5] M. Handley, J. Crowcroft, C. Bormann, and J. Ott, "The internet
multimedia conferencing architecture," Internet Draft, Internet
Engineering Task Force, July 1997. Work in progress.

[6] 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.

[7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
transport protocol for real-time applications," Tech. Rep. RFC 1889,
Internet Engineering Task Force, Jan. 1996.

[8] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
July 1997. Work in progress.

[9] M. Handley and V. Jacobson, "SDP: Session description protocol,"
Internet Draft, Internet Engineering Task Force, Mar. 1997. Work in
progress.

[10] S. Bradner, "Key words for use in RFCs to indicate requirement
level," Tech. Rep. RFC 2119, Internet Engineering Task Force, Mar.
1997.

[11] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-
Lee, "Hypertext transfer protocol -- HTTP/1.1," Tech. Rep. RFC 2068,
Internet Engineering Task Force, Jan. 1997.

[12] C. Partridge, "Mail routing and the domain system," Tech. Rep.

Handley/Schulzrinne/Schooler [Page 77]

Internet Draft SIP November 11, 1997

RFC 974, Internet Engineering Task Force, Jan. 1986.

[13] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the
location of services (DNS SRV)," Tech. Rep. RFC 2052, Internet
Engineering Task Force, Oct. 1996.

[14] P. V. Mockapetris, "Domain names - implementation and
specification," Tech. Rep. RFC 1035, Internet Engineering Task
Force, Nov. 1987.

[15] R. T. Braden, "Requirements for internet hosts - application and
support," Tech. Rep. RFC 1123, Internet Engineering Task Force, Oct.
1989.

[16] D. Zimmerman, "The finger user information protocol," Tech. Rep.
RFC 1288, Internet Engineering Task Force, Dec. 1991.

[17] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
protocol," Tech. Rep. RFC 1777, Internet Engineering Task Force, Mar.
1995.

[18] 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," Tech. Rep. RFC 1630,
Internet Engineering Task Force, June 1994.

[19] 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.

[20] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
locators (URL)," Tech. Rep. RFC 1738, Internet Engineering Task
Force, Dec. 1994.

[21] D. Crocker, "Augmented BNF for syntax specifications: ABNF,"
Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in
progress.

[22] J. C. Mogul and S. E. Deering, "Path MTU discovery," Tech. Rep.
RFC 1191, Internet Engineering Task Force, Nov. 1990.

[23] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1.
Reading, Massachusetts: Addison-Wesley, 1994.

[24] D. Crocker, "Standard for the format of ARPA internet text
messages," Tech. Rep. Also STD0011, RFC 822, Internet Engineering
Task Force, Aug. 1982.

Handley/Schulzrinne/Schooler [Page 78]

Internet Draft SIP November 11, 1997

[25] A. Vaha-Sipila, "URLs for telephony," Internet Draft, Internet
Engineering Task Force, Aug. 1997. Work in progress.

[26] L. Masinter, P. Hoffman, and J. Zawinski, "The mailto URL
scheme," Internet Draft, Internet Engineering Task Force, Oct. 1997.
Work in progress.

[27] International Telecommunication Union, "Integrated services
digital network (ISDN) service capabilities -- definition of
supplementary services," Recommendation I.250, Telecommunication
Standardization Sector of ITU, Geneva, Switzerland, 1993.

[28] International Telecommunication Union, "General recommendations
on telephone switching and signaling -- intelligent network:
Introduction to intelligent network capability set 1," Recommendation
Q.1211, Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, Mar. 1993.

[29] 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.

[30] H. Schulzrinne, "Personal mobility for multimedia services in
the Internet," in European Workshop on Interactive Distributed
Multimedia Systems and Services , (Berlin, Germany), Mar. 1996.

Full Copyright Statement

This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind,
provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.

The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.

Handley/Schulzrinne/Schooler [Page 79]

Internet Draft SIP November 11, 1997

This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Table of Contents

1 Introduction ........................................ 2
1.1 Overview of SIP Functionality ....................... 2
1.2 Terminology ......................................... 3
1.3 Definitions ......................................... 4
1.4 Summary of SIP Operation ............................ 6
1.4.1 SIP Addressing ...................................... 6
1.4.2 Locating a SIP Server ............................... 7
1.4.3 SIP Transaction ..................................... 9
1.4.4 SIP Invitation ...................................... 9
1.4.5 Locating a User ..................................... 10
1.4.6 Changing an Existing Session ........................ 13
1.4.7 Registration Services ............................... 13
1.5 Protocol Properties ................................. 13
1.5.1 Minimal State ....................................... 13
1.5.2 Transport-Protocol Neutral .......................... 14
1.5.3 Text-Based .......................................... 14
2 SIP Uniform Resource Locators ....................... 14
3 SIP Message Overview ................................ 17
4 Request ............................................. 18
4.1 Request-Line ........................................ 18
4.2 Methods ............................................. 19
4.2.1 INVITE ............................................. 20
4.2.2 ACK ................................................ 20
4.2.3 OPTIONS ............................................ 20
4.2.4 BYE ................................................ 20
4.2.5 REGISTER ........................................... 21
4.2.6 UNREGISTER ......................................... 21
4.3 Request-URI ......................................... 21
4.3.1 SIP Version ......................................... 22
4.4 Option Tags ......................................... 22
4.4.1 Registering New Option Tags with IANA ............... 22
5 Response ............................................ 23
5.1 Status-Line ......................................... 23
5.1.1 Status Codes and Reason Phrases ..................... 23

Handley/Schulzrinne/Schooler [Page 80]

Internet Draft SIP November 11, 1997

6 Header Field Definitions ............................ 25
6.1 General Header Fields ............................... 27
6.2 Entity Header Fields ................................ 27
6.3 Request Header Fields ............................... 27
6.4 Response Header Fields .............................. 29
6.5 Header Field Format ................................. 29
6.6 Accept .............................................. 30
6.7 Accept-Language ..................................... 30
6.8 Allow ............................................... 30
6.9 Also ................................................ 30
6.10 Authorization ....................................... 31
6.11 Call-Disposition .................................... 31
6.12 Call-ID ............................................. 32
6.13 Content-Length ...................................... 32
6.14 Content-Type ........................................ 33
6.15 Date ................................................ 33
6.16 Expires ............................................. 33
6.17 From ................................................ 34
6.18 Location ............................................ 35
6.19 Organization ........................................ 37
6.20 Priority ............................................ 37
6.21 Proxy-Authenticate .................................. 38
6.22 Proxy-Authorization ................................. 38
6.23 Public .............................................. 38
6.24 Require ............................................. 38
6.25 Retry-After ......................................... 39
6.26 CSeq ................................................ 39
6.27 Server .............................................. 40
6.28 Subject ............................................. 40
6.29 Unsupported ......................................... 40
6.30 Timestamp ........................................... 41
6.31 To .................................................. 41
6.32 User-Agent .......................................... 41
6.33 Via ................................................. 41
6.34 Warning ............................................. 43
6.35 WWW-Authenticate .................................... 44
7 Status Code Definitions ............................. 44
7.1 Informational 1xx ................................... 44
7.1.1 100 Trying .......................................... 44
7.1.2 180 Ringing ......................................... 44
7.1.3 181 Queued .......................................... 45
7.2 Successful 2xx ...................................... 45
7.2.1 200 OK .............................................. 45
7.3 Redirection 3xx ..................................... 45
7.3.1 300 Multiple Choices ................................ 45
7.3.2 301 Moved Permanently ............................... 46
7.3.3 302 Moved Temporarily ............................... 46
7.3.4 380 Alternative Service ............................. 46

Handley/Schulzrinne/Schooler [Page 81]

Internet Draft SIP November 11, 1997

7.4 Request Failure 4xx ................................. 46
7.4.1 400 Bad Request ..................................... 46
7.4.2 401 Unauthorized .................................... 46
7.4.3 402 Payment Required ................................ 46
7.4.4 403 Forbidden ....................................... 46
7.4.5 404 Not Found ....................................... 46
7.4.6 405 Method Not Allowed .............................. 47
7.4.7 407 Proxy Authentication Required ................... 47
7.4.8 408 Request Timeout ................................. 47
7.4.9 420 Bad Extension ................................... 47
7.4.10 480 Temporarily Unavailable ......................... 47
7.4.11 481 Invalid Call-ID ................................. 47
7.4.12 482 Loop Detected ................................... 48
7.5 Server Failure 5xx .................................. 48
7.5.1 500 Server Internal Error ........................... 48
7.5.2 501 Not implemented ................................. 48
7.5.3 502 Bad Gateway ..................................... 48
7.5.4 503 Service Unavailable ............................. 48
7.5.5 504 Gateway Timeout ................................. 48
7.6 Global Failures 6xx ................................. 49
7.6.1 600 Busy ............................................ 49
7.6.2 603 Decline ......................................... 49
7.6.3 604 Does not exist anywhere ......................... 49
7.6.4 606 Not Acceptable .................................. 49
8 SIP Message Body .................................... 50
8.1 Body Inclusion ...................................... 50
8.2 Message Body Length ................................. 50
9 Examples ............................................ 51
9.1 Invitation to Multimedia Conference ................. 51
9.1.1 Request ............................................. 51
9.1.2 Reply ............................................... 52
9.2 Two-party Call ...................................... 53
9.3 Aborting a Call ..................................... 54
9.3.1 Redirects ........................................... 54
9.3.2 Alternative Services ................................ 55
9.3.3 Negotiation ......................................... 56
9.4 OPTIONS Request ..................................... 57
10 Compact Form ........................................ 57
11 SIP Transport ....................................... 58
11.1 Achieving Reliability For UDP Transport ............. 59
11.1.1 General Operation ................................... 59
11.1.2 INVITE .............................................. 59
11.2 Connection Management for TCP ....................... 60
12 Behavior of SIP Servers ............................. 63
12.1 Redirect Server ..................................... 63
12.2 User Agent Server ................................... 63
12.3 Proxies Issuing Single Unicast Requests ............. 64
12.4 Proxy Server Issuing Several Requests ............... 64

Handley/Schulzrinne/Schooler [Page 82]

Internet Draft SIP November 11, 1997

13 Third-Party Call Initiation ......................... 67
14 ISDN and Intelligent Network Services ............... 68
15 Security Considerations ............................. 72
15.1 Confidentiality ..................................... 72
15.2 Integrity ........................................... 72
15.3 Access Control ...................................... 72
15.4 Privacy ............................................. 73
A Minimal Implementation .............................. 73
A.1 Client .............................................. 73
A.2 Server .............................................. 74
B Summary of Augmented BNF ............................ 74
C Open Issues ......................................... 76
D Acknowledgments ..................................... 76
E Authors' Addresses .................................. 76
F Bibliography ........................................ 77

Handley/Schulzrinne/Schooler [Page 83]

PAFTECH AB 2003-2026

2026-04-22 23:25:47