draft-ietf-mmusic-sip-03.txt   draft-ietf-mmusic-sip-04.txt 
Internet Engineering Task Force MMUSIC WG Internet Engineering Task Force MMUSIC WG
Internet Draft Handley/Schulzrinne/Schooler Internet Draft Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-03.txt ISI/Columbia U./Caltech draft-ietf-mmusic-sip-04.txt ISI/Columbia U./Caltech
July 31, 1997 November 11, 1997
Expires: January 20, 1998 Expires: April 1, 1998
SIP: Session Initiation Protocol SIP: Session Initiation Protocol
STATUS OF THIS MEMO STATUS OF THIS MEMO
This document is an Internet-Draft. Internet-Drafts are working This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas, documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts. working documents as Internet-Drafts.
skipping to change at page 2, line 12 skipping to change at page 2, line 12
should be addressed to the working group's mailing list should be addressed to the working group's mailing list
at confctrl@isi.edu and/or the authors. at confctrl@isi.edu and/or the authors.
1 Introduction 1 Introduction
1.1 Overview of SIP Functionality 1.1 Overview of SIP Functionality
The Session Initiation Protocol (SIP) is an application-layer The Session Initiation Protocol (SIP) is an application-layer
protocol that can establish and control multimedia sessions or calls. protocol that can establish and control multimedia sessions or calls.
These multimedia sessions include multimedia conferences, distance These multimedia sessions include multimedia conferences, distance
learning, Internet telephony and similar applications. SIP can learning, Internet telephony and similar applications. SIP can invite
initiate both unicast and multicast sessions; the initiator does not a person to both unicast and multicast sessions; the initiator does
necessarily have to be a member of the session. Media and not necessarily have to be a member of the session it is inviting to.
participants can be added to an existing session. SIP can be used to Media and participants can be added to an existing session. SIP can
"call" both persons and "robots", for example, to invite a media be used to "call" both persons and "robots", for example, to invite a
storage device to record an ongoing conference or to invite a video- media storage device to record an ongoing conference or to invite a
on-demand server to play a video into a conference. (SIP does not video-on-demand server to play a video into a conference. (SIP does
directly control these services, however; see RTSP [4].) 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, SIP transparently supports name mapping and redirection services,
allowing the implementation of telephony services such as selective allowing the implementation of ISDN and Intelligent Network telephony
call forwarding, selective call rejection, conditional and subscriber services. Section 14 discusses these services in detail.
unconditional call forwarding, call forwarding busy, call forwarding
no response. SIP may use multicast to try several possible callee
locations at the same time.
SIP supports personal mobility telecommunications intelligent network SIP supports personal mobility telecommunications intelligent network
services, this is defined as: "Personal mobility is the ability of services, this is defined as: "Personal mobility is the ability of
end users to originate and receive calls and access subscribed end users to originate and receive calls and access subscribed
telecommunication services on any terminal in any location, and the telecommunication services on any terminal in any location, and the
ability of the network to identify end users as they move. Personal 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., mobility is based on the use of a unique personal identity (i.e.,
'personal number')." [1]. Personal mobility complements terminal 'personal number')." [3]. Personal mobility complements terminal
mobility, i.e., the ability to maintain communications when moving a mobility, i.e., the ability to maintain communications when moving a
single end system from one network to another. single end system from one network to another.
SIP supports some or all of four facets of establishing multimedia SIP supports some or all of five facets of establishing and
communications: terminating multimedia communications:
1. user location: determination of the end system to be used
for communication;
2. user capabilities: determination of the media and media
parameters to be used;
3. user availability: determination of the willingness of the
called party to engage in communications;
4. call setup ("ringing", establishment of call parameters at
both called and calling party)
In particular, SIP can be used to locate a user and determine
the appropriate end system, leaving the actual call
establishment to other protocols such as H.323.
SIP may also be used to terminate and transfer a call. SIP can also
initiate multi-party calls using a multipoint control unit (MCU) or
fully-meshed interconnection instead of multicast.
These features are for further study.
SIP is not a conference control protocol, but can be used to User location: determination of the end system to be used for
introduce conference control protocols to a session. communication;
SIP is designed as part of the overall IETF multimedia data and User capabilities: determination of the media and media parameters to
control architecture currently incorporating protocols such as RSVP be used;
[2] for reserving network resources, RTP [3] for transporting real-
time data and providing QOS feedback, RTSP [4] for controlling
delivery of streaming media, SAP [5] for advertising multimedia
sessions via multicast and SDP [6] for describing multimedia
sessions, but SIP does not depend for its operation on any of these
protocols.
1.2 Finding Multimedia Sessions User availability: determination of the willingness of the called
party to engage in communications;
Call setup: "ringing", establishment of call parameters at both
called and calling party;
There are two basic ways to locate and to participate in a multimedia Call handling: including transfer and termination of calls.
session:
Advertisement: The session is advertised, potential participants see SIP may also be used in conjunction with other call setup and
the advertisement, then join the session address to participate. 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.
Invitation: Users are invited by others to participate in a session, SIP can also initiate multi-party calls using a multipoint control
which may or may not be advertised. unit (MCU) or fully-meshed interconnection instead of multicast.
Sessions may be advertised using multicast protocols such as SAP [5], Internet telephony gateways that connect PSTN parties may also use
electronic mail, news groups, web pages or directories (LDAP), among SIP to set up calls between them.
others. SIP serves the role of the invitation protocol.
SIP does not prescribe how a conference is to be managed, e.g., SIP does not offer conference control services such as floor control
whether it uses a central server to manage conference and participant or voting and does not prescribe how a conference is to be managed,
state or distributes state via multicast. but SIP can be used to introduce conference control protocols.
SIP does not allocate multicast addresses, leaving this functionality SIP does not allocate multicast addresses, leaving this functionality
to protocols such as SAP [5]. to protocols such as SAP [2].
SIP can invite users to conferences with and without resource SIP can invite users to sessions with and without resource
reservation. SIP does not reserve resources, but may convey to the reservation. SIP does not reserve resources, but may convey to the
invited system the information necessary to do this. Quality-of- invited system the information necessary to do this. Quality-of-
service guarantees, if required, may depend on knowing the full service guarantees, if required, may depend on knowing the full
membership of the session; this information may or may not be known membership of the session; this information may or may not be known
to the agent performing session invitation. to the agent performing session invitation.
SIP offers some of the same functionality as H.323, but may also be SIP is designed as part of the overall IETF multimedia data and
used in conjunction with it. In this mode, H.323 is used to locate control architecture [5] currently incorporating protocols such as
the appropriate terminal identified by a H.245 address [TBD: what RSVP [6] for reserving network resources, the real-time transport
does this look like?]. An H.323-capable terminal then proceeds with a protocol (RTP) [7] for transporting real-time data and providing QOS
normal H.323/H.245 invitation [7]. feedback, the real-time streaming protocol (RTSP) [8] for controlling
delivery of streaming media, the session announcement protocol (SAP)
1.3 Terminology [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
In this document, the key words "MUST", "MUST NOT", "REQUIRED", In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [8] and and "OPTIONAL" are to be interpreted as described in RFC 2119 [10]
indicate requirement levels for compliant SIP implementations. and indicate requirement levels for compliant SIP implementations.
1.4 Definitions 1.3 Definitions
This specification uses a number of terms to refer to the roles This specification uses a number of terms to refer to the roles
played by participants in SIP communications. The definitions of played by participants in SIP communications. The definitions of
client, server and proxy are similar to those used by the Hypertext client, server and proxy are similar to those used by the Hypertext
Transport Protocol (HTTP) [9]. 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 Client: An application program that establishes connections for the
purpose of sending requests. Clients may or may not interact purpose of sending requests. Clients may or may not interact
directly with a human user. directly with a human user.
Final response: A response that terminates a -> SIP transaction, as Final response: A response that terminates a SIP transaction, as
opposed to a -> provisional response 3xx, 4xx, and 5xx opposed to a provisional response responses are final.
responses are final.
Initiator, calling party: The party initiating a conference Initiator, calling party: The party initiating a conference
invitation. Note that the calling party does not have to be the invitation. Note that the calling party does not have to be the
same as the one creating a conference. same as the one creating a conference.
Invitation: A request sent to a user (or service) requesting Invitation: A request sent to a user (or service) requesting
participation in a session. 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 Invitee, invited user, called party: The person or service that the
calling party is trying to invite to a conference. calling party is trying to invite to a conference.
Location server: A program that is contacted by a -> client and Location server: See location service
that returns one or more possible locations of the called party
or service. Location servers may be invoked by SIP redirect and
proxy servers and may be Co-located with a SIP server.
Location service: A service used by a -> redirect or -> proxy Location service: A service used by a SIP redirect or proxy server to
server to obtain information about a callee's possible location. 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
scope of the document.
Provisional response: A response used by the server to indicate Provisional response: A response used by the server to indicate
progress, but that does not terminate a -> SIP transaction. progress, but that does not terminate a SIP transaction. All 1xx
All 1xx and 6xx responses are provisional. Other responses are and 6xx responses are provisional. Other responses are
considered -> final. considered final.
Proxy, proxy server: An intermediary program that acts as both a Proxy, proxy server: An intermediary program that acts as both a
server and a client for the purpose of making requests on behalf server and a client for the purpose of making requests on behalf
of other clients. Requests are serviced internally or by passing of other clients. Requests are serviced internally or by passing
them on, possibly after translation, to other servers. A proxy them on, possibly after translation, to other servers. A proxy
must interpret, and, if necessary, rewrite a request message must interpret, and, if necessary, rewrite a request message
before forwarding it. before forwarding it.
Redirect server: A server that accepts a SIP request, maps the Redirect server: A server that accepts a SIP request, maps the
address into zero or more new addresses and returns these address into zero or more new addresses and returns these
addresses to the client. Unlike a -> proxy server, it does not addresses to the client. Unlike a proxy server, it does not
initiate its own SIP request. 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 Server: An application program that accepts requests in order to
service requests and sends back responses to those requests. service requests and sends back responses to those requests.
Servers are either proxy, redirect or user agent servers. An Servers are either proxy, redirect or user agent servers. An
application program may act as both server and client. application program may act as both server and client.
Session: "A multimedia session is a set of multimedia senders and Session: "A multimedia session is a set of multimedia senders and
receivers and the data streams flowing from senders to receivers and the data streams flowing from senders to
receivers. A multimedia conference is an example of a multimedia receivers. A multimedia conference is an example of a multimedia
session." [6] For SIP, a session is equivalent to a "call". session." [9] (Note: a session as defined here may comprise one
(Note: a session as defined here may comprise one or more RTP or more RTP sessions.) Since the word session is used
sessions.) differently by protocols relevant to SIP, this document avoids
the term altogether.
(SIP) transaction: A SIP transaction occurs between a -> client and (SIP) transaction: A SIP transaction occurs between a client and a
a -> server and comprises all messages from the first request server and comprises all messages from the first request sent
sent from the client to the server up to a -> final (non-1xx) from the client to the server up to a final (non-1xx) response
response sent from the server to the client. A transaction is sent from the server to the client. A transaction is for a
for a single call (identified by a Call-ID, Section 6.11). single call (identified by a Call-ID, Section 6.12). There can
There can only be one pending transaction between a server and only be one pending transaction between a server and client for
client for each call id. each call id.
User agent server, called user agent: The server application that User agent server, called user agent: The server application that
contacts the user when a session request is received and that contacts the user when a SIP request is received and that
returns a reply on behalf of the user. The reply may accept, returns a reply on behalf of the user. The reply may accept,
reject or redirect the call. (Note: in SIP, user agents can be reject or redirect the call. (Note: in SIP, user agents can be
both clients and servers.) both clients and servers.)
An application program may be capable of acting both as a client and An application program may be capable of acting both as a client and
a server. For example, a typical multimedia conference control a server. For example, a typical multimedia conference control
application would act as a client to initiate calls or to invite application would act as a client to initiate calls or to invite
others to conferences and as a user agent server to accept others to conferences and as a user agent server to accept
invitations. invitations. The properties of the different SIP server types are
summarized in Table 1.
1.5 Protocol Properties
1.5.1 Minimal State
There is no concept of an ongoing SIP session that lasts for the
duration of the conference or call. Rather, a single conference
session or call may involve one or more SIP request-response
transactions. For example, a conference control protocol may use SIP
to add or remove a media stream, but again, once the information has
been successfully conveyed to the participants, SIP is then no longer
involved.
At most, a server has to maintain state for a single SIP transaction.
In some cases, it can process each message without regard to previous
messages ( stateless server ), as described in Section 12.
1.5.2 Transport-Protocol Neutral
SIP is able to utilize both UDP and TCP as transport protocols. UDP
allows the application to more carefully control the timing of
messages and their retransmission, to perform parallel searches
without requiring connection state for each outstanding request, and
to use multicast. TCP allows easier passage through existing
firewalls, and given the similar protocol design, allows common
servers for SIP, HTTP and the Real Time Streaming Protocol (RTSP)
[4].
When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing session. The
concept of a session is not implicitly bound to a TCP connection, so
the initial SIP request and a subsequent SIP request may use
different TCP connections or a single persistent connection as
appropriate.
Clients SHOULD implement both UDP and TCP transport, servers MUST. 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
1.5.3 Text-Based Table 1: Properties of the different SIP server types
SIP is text based. This allows easy implementation in languages such 1.4 Summary of SIP Operation
as Tcl and Perl, allows easy debugging, and most importantly, makes
SIP flexible and extensible. As SIP is primarily used for session
initiation, it is believed that the additional overhead of using a
text-based protocol is not significant.
1.6 SIP Addressing 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).
SIP uses two kinds of address identifiers, host-specific addresses 1.4.1 SIP Addressing
and host-independent addresses form user@host , where user is any
alphanumeric identifier and the form of host depends on the address
type. Note that SIP does not distinguish between the two and can,
while inviting a user, map repeatedly between the two address types.
For a host-specific address, the user part is an operating-system SIP addresses contain a user and host part. The user part is an
user name. The host part is either a domain name having a DNS A operating-system user name. The host part is either a domain name
(address) record, or a numeric network address. Examples include: having a DNS A (address) record, or a numeric network address.
Examples include:
mjh@metro.isi.edu mjh@metro.isi.edu
hgs@erlang.cs.columbia.edu hgs@erlang.cs.columbia.edu
root@[193.175.132.42]
root@193.175.132.42 root@193.175.132.42
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.
A user's host-specific address can be obtained out-of-band, can be SIP addresses may contain a moniker (such as a civil name) or user
learned via existing media agents, can be included in some mailers' name and domain name that may not map into a single host. [1]
message headers, or can be recorded during previous invitation
interactions.
Host-independent addresses contain a moniker (such as a civil name)
or user name and domain name that may not map into a single host.
[1]
Host-independent addresses may use any unambiguous user name, SIP addresses may use any unambiguous user name, including aliases,
including aliases, identifying the called party as the user part of identifying the called party as the user part of the address. They
the address. They may use a domain name having an MX [10], SRV [11] may use a domain name having an MX [12], SRV [13] or A [14] record
or A [12] record for the host part. These addresses may have for the host part. These addresses may have different degrees of
different degrees of location- and provider-independence and are location- and provider-independence and are often chosen to be
often chosen to be mnemonic. In many cases, the host-independent SIP mnemonic. In many cases, the SIP address can be the same as a user's
address can be the same as a user's electronic mail address, but this electronic mail address, but this is not required. SIP can thus
is not required. SIP can thus leverage off the domain name system leverage off the domain name system (DNS) to provide a first-stage
(DNS) to provide a first-stage location mechanisms. Examples of location mechanisms. Examples of SIP names include
host-independent names include
M.Handley@cs.ucl.ac.uk M.Handley@cs.ucl.ac.uk
H.G.Schulzrinne@ieee.org H.G.Schulzrinne@ieee.org
info@ietf.org info@ietf.org
An address can designate an individual (possibly located at one of An address can designate an individual (possibly located at one of
several end systems), the first available person from a group of several end systems), the first available person from a group of
individuals or a whole group. The form of the address, e.g., individuals or a whole group. The form of the address, e.g.,
_________________________
[1] We avoid the term location-independent , since
the address may indeed refer to a specific location,
e.g., a company department.
sales@example.com , is not sufficient, in general, to determine the sales@example.com , is not sufficient, in general, to determine the
intent of the caller. intent of the caller.
If a user or service chooses to be reachable at an address that is If a user or service chooses to be reachable at an address that is
guessable from the person's name and organizational affiliation, the guessable from the person's name and organizational affiliation, the
traditional method of ensuring privacy by having an unlisted "phone" traditional method of ensuring privacy by having an unlisted "phone"
number is compromised. However, unlike traditional telephony, SIP number is compromised. However, unlike traditional telephony, SIP
offers authentication and access control mechanisms and can avail offers authentication and access control mechanisms and can avail
itself of lower-layer security mechanisms, so that client software itself of lower-layer security mechanisms, so that client software
can reject unauthorized or undesired call attempts. can reject unauthorized or undesired call attempts.
1.7 Locating a SIP Server 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.
Call setup may proceed in several phases. A SIP client MUST follow 1.4.2 Locating a SIP Server
the following steps to resolve the user part of a callee address. If _________________________
a client only supports TCP or UDP, but not both, the respective
address type is omitted.
1. If there is a SRV DNS resource record [11] of type sip.udp [1] We avoid the term location-independent , since
, contact the listed SIP servers in order of preference the address may indeed refer to a specific location,
value using UDP as a transport protocol at the port number e.g., a company department.
listed in the DNS resource record.
2. If there is a SRV DNS resource record [11] of type sip.tcp A SIP client MUST follow the following steps to resolve the host part
, contact the listed SIP servers in order of preference of a callee address. If a client only supports TCP or UDP, but not
value using TCP as a transport protocol at the port number both, the respective address type is omitted. If the SIP address
listed in the DNS resource record. 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.
3. If there is a DNS MX record [10], contact the hosts listed 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 in their order of preference at the default port number
(TBD). For each host listed, first try to contact the (TBD). For each host listed, first try to contact the SIP
server using UDP, then TCP. server using UDP, then TCP.
4. Finally, check if there is a DNS CNAME or A record for the 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 given host and try to contact a SIP server at the one or
more addresses listed, again trying first UDP, then TCP. more addresses listed, again trying first UDP, then TCP.
5. If all of the above methods fail, the caller MAY contact an 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 SMTP server at the user's host and use the SMTP EXPN
command to obtain an alternate address and repeat the steps command to obtain an alternate address and repeat the steps
above. As a last resort, a client MAY choose to deliver the above. As a last resort, a client MAY choose to deliver the
session description to the callee using electronic mail. session description to the callee using electronic mail.
If a server is found using one of the methods below, the other If a server is found using one of the methods below, the other
methods are not tried. A client SHOULD rely on ICMP "Port methods are not tried. A client SHOULD rely on ICMP "Port
Unreachable" messages rather than time-outs to determine that a Unreachable" messages rather than time-outs to determine that a
server is not reachable at a particular address. A client MAY cache server is not reachable at a particular address.
the result of the reachability steps, but SHOULD start at the
beginning of the sequence when the cached address fails. 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
start the search at the beginning of the sequence.
Implementation note for socket-based programs: For TCP, connect() Implementation note for socket-based programs: For TCP, connect()
returns ECONNREFUSED if there is no server at the designated address; returns ECONNREFUSED if there is no server at the designated address;
for UDP, the socket should be bound to the destination address using for UDP, the socket should be bound to the destination address using
connect() rather than sendto() or similar. connect() rather than sendto() or similar.
This sequence is modeled after that described for SMTP, This sequence is modeled after that described for SMTP,
where MX records are to be checked before A records [13]. where MX records are to be checked before A records [15].
1.8 SIP Transactions 1.4.3 SIP Transaction
Once the host part has been resolved to a SIP server, the client 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 sends one or more SIP requests to that server and receives one or
more responses from the server. If the invitation is SIP request is more responses from the server. A request (and its retransmissions)
an invitation, it contains a session description, for example written together with the responses triggered by that request make up a SIP
in SDP format, that provides the called party with enough information transaction.
to join the session.
If TCP is used, request and responses within a single 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 are carried over the same TCP connection. Thus, the client SHOULD
maintain the connection until a final response has been received. maintain the connection until a final response has been received.
Several SIP requests from the same client to the same server may use 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 the same TCP connection or may open a new connection for each
request. If the client sent the request sends via unicast UDP, the request. If the client sent the request sends via unicast UDP, the
response is sent to the source address of the UDP request. If the response is sent to the source address of the UDP request.
request is sent via multicast UDP, the response is directed to the (Implementation note: use recvfrom() to obtain the source address and
same multicast address and destination port. For UDP, reliability is port of the request.) If the request is sent via multicast UDP, the
achieved using retransmission (Section 11). 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 Need motivation why we ALWAYS want to have a multicast
return. return.
The SIP message format and operation is independent of the transport The SIP message format and operation is independent of the transport
protocol. protocol.
The basic message flow is shown in Fig. 1 and Fig. 2, for proxy and 1.4.4 SIP Invitation
redirect modes, respectively.
1.9 Locating a User 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.
A callee may move between a number of different end systems over The INVITE request typically contains a session description, for
time. These locations can be dynamically registered with a location example written in SDP format, that provides the called party with
server, typically for a single administrative domain, or a location 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
+....... cs.columbia.edu .......+ +....... cs.columbia.edu .......+
: : : :
: (~~~~~~~~~~) : : (~~~~~~~~~~) :
: ( location ) : : ( location ) :
: ( service ) : : ( service ) :
: (~~~~~~~~~~) : : (~~~~~~~~~~) :
: ^ | : : ^ | :
: | hgs@play : : | hgs@play :
: 2| 3| : : 2| 3| :
: | | : : | | :
skipping to change at page 10, line 27 skipping to change at page 11, line 27
: cz@cs.tu-berlin.de ========================> tune =========> play : : cz@cs.tu-berlin.de ========================> tune =========> play :
: <........................ <......... : : <........................ <......... :
: : 7: 200 OK : 6: 200 OK : : : 7: 200 OK : 6: 200 OK :
+.....................+ +...............................+ +.....................+ +...............................+
====> SIP request ====> SIP request
----> non-SIP protocols ----> non-SIP protocols
Figure 1: Example of SIP proxy server Figure 1: Example of SIP proxy server
server may use other protocols, such as finger [14], rwho, A callee may move between a number of different end systems over
multicast-based protocols or operating-system dependent mechanism to time. These locations can be dynamically registered with the SIP
actively determine the end system where a user is reachable. The server (Section 4.2.5) or a location server, typically for a single
location services yield a list of a zero or more possible locations, administrative domain, or a location server may use other protocols,
possibly even sorted in order of likelihood of success. 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 The location server can be part of the SIP server or the SIP server
may use a different protocol (e.g., finger [14] or LDAP [15]) to map may use a different protocol (e.g., finger [16] or LDAP [17]) to map
addresses. A single user may be registered at different locations, addresses. A single user may be registered at different locations,
either because she is logged in at several hosts simultaneously or either because she is logged in at several hosts simultaneously or
because the location server has (temporarily) inaccurate information. because the location server has (temporarily) inaccurate information.
The action taken on receiving a list of locations varies with the 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 type of SIP server. A SIP redirect server simply returns the list to
the client sending the request as Location headers (Section 6.17). A the client sending the request as Location headers (Section 6.18). A
SIP proxy server can sequentially try the addresses until the call is SIP proxy server can sequentially or in parallel try the addresses
successful (2xx response) or the callee has declined the call (40x
response). Alternatively, the server may issue several requests in
parallel. A proxy server can only issue more than one sequential or
parallel connection request if it is the first in the chain of hosts
+....... cs.columbia.edu .......+ +....... cs.columbia.edu .......+
: : : :
: (~~~~~~~~~~) : : (~~~~~~~~~~) :
: ( location ) : : ( location ) :
: ( service ) : : ( service ) :
: (~~~~~~~~~~) : : (~~~~~~~~~~) :
: ^ | : : ^ | :
: | hgs@play : : | hgs@play :
: 2| 3| : : 2| 3| :
: | | : : | | :
skipping to change at page 11, line 33 skipping to change at page 12, line 33
. ==================================================> play : . ==================================================> play :
..................................................... : ..................................................... :
7: 200 OK : : 7: 200 OK : :
+...............................+ +...............................+
====> SIP request ====> SIP request
----> non-SIP protocols ----> non-SIP protocols
Figure 2: Example of SIP redirect server Figure 2: Example of SIP redirect server
noted in the Via header to do so. If it is not the first, it must until the call is successful (2xx response) or the callee has
issue a redirect response. 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 SHOULD add itself to the 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 6.31) end of the list of forwarders noted in the Via (Section 6.33)
headers. A proxy server also notes whether it is attempting to reach headers. The Via trace ensures that replies can take the same path
several possible locations at once ("connection forking"). The Via back, thus ensuring correct operation through compliant firewalls and
trace ensures that replies can take the same path back, thus ensuring loop-free requests. On the reply path, each host most remove its Via,
correct operation through compliant firewalls and loop-free requests. so that routing internal information is hidden from the callee and
On the reply path, each host most remove its Via, so that routing outside networks. When a multicast request is made, first the host
internal information is hidden from the callee and outside networks. making the request, then the multicast address itself are added to
When a multicast request is made, first the host making the request, the path. A proxy server MUST check that it does not generate a
then the multicast address itself are added to the path. request to a host listed in the Via list. (Note: If a host has
several names or network addresses, this may not always work. Thus,
each host also checks if it is part of the Via list.)
As discussed in Section 1.6, a SIP address may designate a group 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 rather than an individual. A client indicates using the Reach
request header whether it wants to reach the first available request header whether it wants to reach the first available
individual or treat the address as a group, to be invited as a whole. 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 default is to attempt to reach the first available callee. If
the address is designated as a group address, a proxy server MUST the address is designated as a group address, a proxy server MUST
return the list of individuals instead of attempting to connect to return the list of individuals instead of attempting to connect to
these. 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.)
Otherwise, the proxy cannot report errors and call status 1.4.6 Changing an Existing Session
appropriately.
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.
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 2 SIP Uniform Resource Locators
SIP URLs are used within SIP messages to indicate the originator and SIP URLs are used within SIP messages to indicate the originator and
recipient of a SIP request, and to specify redirection addresses. A recipient of a SIP request, and to specify redirection addresses. A
SIP URL may be embedded in web pages or other hyperlinks to indicate SIP URL may also be embedded in web pages or other hyperlinks to
that a user or service may be called. Within SIP messages, an email indicate that a user or service may be called.
address could have been used, but this would have made it more
difficult to gateway between SIP and other protocols with other
addressing schemes.
For greater functionality, because interaction with some resources Because interaction with some resources may require message headers
may require message headers or message bodies to be specified as well or message bodies to be specified as well as the SIP address, the sip
as the SIP address, the sip URL scheme is extended to allow setting URL scheme is defined to allow setting SIP request-header fields and
SIP request-header fields and the SIP message-body. the SIP message-body. (This is similar to the mailto: URL.)
A SIP URL follows the guidelines of RFC 1630 [16,17] and takes the A SIP URL follows the guidelines of RFC 1630 [18,19] and takes the
following form: following form:
SIP-URL = short-sip-url | full-sip-url SIP-URL = short-sip-url | full-sip-url
full-sip-url = "sip://" user [ ":" password ] "@" host full-sip-url = "sip://" ( user | phone ) [ ":" password ]
"@" [ host | nhost ]
url-parameters [ headers ] url-parameters [ headers ]
short-sip-url = user [ ":" password ] "@" host : port short-sip-url = ( user | phone) [ ":" password ]
user = ; defined in RFC 1738 [18] "@" [ host | nhost ] : port
user = ; defined in RFC 1738 [20]
phone = "+" DIGIT *( DIGIT | "-" | "." )
host = ; defined in RFC 1738 host = ; defined in RFC 1738
nhost = "[" hostnumber "]" | hostnumber
hostnumber = digits "." digits "." digits "." digits
port = *digit port = *digit
url-parameters = *( ";" url-parameter) url-parameters = *( ";" url-parameter)
url-parameter = transport-param | url-parameter = transport-param |
ttl-param | maddr-param ttl-param | maddr-param
transport-param = "transport=" ( "udp" | "tcp" ) transport-param = "transport=" ( "udp" | "tcp" )
ttl-param = "ttl=" ttl ttl-param = "ttl=" ttl
ttl = 1*3DIGIT ; 0 to 255 ttl = 1*3DIGIT ; 0 to 255
maddr-param = "maddr=" maddr maddr-param = "maddr=" maddr
maddr = ; dotted decimal multicast address maddr = ; dotted decimal multicast address
headers = "?" header *( " " header ) headers = "?" header *( " " header )
header = hname "=" hvalue header = hname "=" hvalue
hname = *urlc hname = *urlc
hvalue = *urlc hvalue = *urlc
urlc = ; defined in [17] urlc = ; defined in [19]
digits = 1*digit
Thus a SIP URL can take either a short form or a full form. The short Thus, a SIP URL can take either a short form or a full form. The
form MAY only be used within SIP messages where the scheme (SIP) can short form MAY only be used within SIP messages where the scheme
be assumed. In all other cases, and when parameters are required to (SIP) can be assumed. In all other cases, and when parameters are
be specified, the full form MUST be used. required to be specified, the full form MUST be used.
Note that all URL reserved characters must be encoded. The special Note that all URL reserved characters must be encoded. The special
hname "body" indicates that the associated hvalue is the message- hname "body" indicates that the associated hvalue is the message-
body of the SIP INVITE request. Within sip URLs, the characters body of the SIP INVITE request. Within sip URLs, the characters
"?", "=", "&" are reserved. "?", "=", "&" are reserved.
Examples of short and full form SIP URLs with identical address are: 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
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 j.doe@big.com
sip://j.doe@big.com sip://j.doe@big.com
sip://j.doe:secret@big.com;transport=tcp sip://j.doe:secret@big.com;transport=tcp
sip://j.doe@big.com?subject=project sip://j.doe@big.com?subject=project
sip://+1-212-555-1212:1234@gateway.com
The password parameter allows to easily specify a call-back address sip://alice@[10.1.2.3]
on a secure web page, but carries the same security risks as all sip://alice@10.1.2.3
URL-based passwords and should only be used under special
circumstances where security requirements are low or all transport
paths are secured.
Within a SIP message, URLs are used to indicate the source and Within a SIP message, URLs are used to indicate the source and
intended destination of a request, redirection addresses and the intended destination of a request, redirection addresses and the
current destination of a request. Normally all these fields will current destination of a request. Normally all these fields will
contain SIP URLs. When additional parameters are not required, the contain SIP URLs. When additional parameters are not required, the
short form SIP URL can be used unambiguously. short form SIP URL can be used unambiguously.
In some circumstances a non-SIP URL may be used in a SIP message. An 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 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 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 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. 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 Thus where SIP specifies user addresses it allows these addresses to
be URLs. be URLs.
Clearly not all URLs are appropriate to be used in a SIP message as a Clearly not all URLs are appropriate to be used in a SIP message as a
user address. It is unlikely, for example, that HTTP or FTP URLs are user address. The correct behavior when an unknown scheme is
useful in this context. The correct behavior when an unknown scheme encountered by a SIP server is defined in the context of each of the
is encountered by a SIP server is defined in the context of each of header fields that use a SIP URL.
the header fields that use a SIP URL.
SIP URLs can define specific parameters of the request, including the SIP URLs can define specific parameters of the request, including the
transport mechanism (UDP or TCP) and the use of multicast to make a transport mechanism (UDP or TCP) and the use of multicast to make a
request. These parameters are added after the host and are separated 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 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 multicast to 239.255.255.1 with a ttl of 15, the following URL would
be used: be used:
sip://j.doe@big.com;maddr=239.255.255.1;ttl=15 sip://j.doe@big.com;maddr=239.255.255.1;ttl=15
The transport protocol UDP is to be assumed when a multicast address The transport protocol UDP is to be assumed when a multicast address
is given. is given.
3 SIP Message Overview 3 SIP Message Overview
Since much of the message syntax is identical to HTTP/1.1, rather 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 than repeating it here we use [HX.Y] to refer to Section X.Y of the
current HTTP/1.1 specification [9]. In addition, we describe SIP in current HTTP/1.1 specification [11]. In addition, we describe SIP in
both prose and an augmented Backus-Naur form (BNF) [H2.1] described both prose and an augmented Backus-Naur form (BNF) [H2.1] described
in detail in [19]. in detail in [21].
All SIP messages are text-based and use HTTP/1.1 conventions [H4.1], 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 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 TCP or UDP, multiple SIP transactions can be carried in a single TCP
connection or UDP datagram. UDP datagrams, including all headers, connection or UDP datagram. UDP datagrams, including all headers,
should not normally be larger than the path maximum transmission unit 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. (MTU) if the MTU is known, or 1500 bytes if the MTU is unknown.
The 1400 bytes accommodates lower-layer packet headers The 1400 bytes accommodates lower-layer packet headers
within the "typical" MTU of around 1500 bytes. There are within the "typical" MTU of around 1500 bytes. There are
few MTU values around 1 kB; the next value is 1006 bytes few MTU values around 1 kB; the next value is 1006 bytes
for SLIP and 296 for low-delay PPP [20]. Recent studies for SLIP and 296 for low-delay PPP [22]. Recent studies
[21] indicate that an MTU of 1500 bytes is a reasonable [23] indicate that an MTU of 1500 bytes is a reasonable
assumption. Thus, another reasonable value would be a assumption. Thus, another reasonable value would be a
message size of 950 bytes, to accommodate packet headers message size of 950 bytes, to accommodate packet headers
within the SLIP MTU without fragmentation. within the SLIP MTU without fragmentation.
A SIP message is either a request from a client to a server, or a A SIP message is either a request from a client to a server, or a
response from a server to a client. response from a server to a client.
SIP-message = Request | Response ; SIP messages SIP-message ___ Request | Response ; SIP messages
Both Request (section 4) and Response (section 5) messages use the Both Request (section 4) and Response (section 5) messages use the
generic message format of RFC 822 [22] for transferring entities (the generic message format of RFC 822 [24] for transferring entities (the
payload of the message). Both types of message consist of a start- 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, one or more header fields (also known as "headers"), an empty
line (i.e., a line with nothing preceding the carriage-return line- line (i.e., a line with nothing preceding the carriage-return line-
feed ( CRLF)) indicating the end of the header fields, and an feed ( CRLF)) indicating the end of the header fields, and an
optional message-body. To avoid confusion with similar-named headers optional message-body. To avoid confusion with similar-named headers
in HTTP, we refer to the header describing the message body as entity in HTTP, we refer to the header describing the message body as entity
headers. These components are described in detail in the upcoming headers. These components are described in detail in the upcoming
sections. sections.
generic-message = start-line generic-message = start-line
*message-header *message-header
CRLF CRLF
[ message-body ] [ message-body ]
start-line = Request-Line | Status-Line start-line = Request-Line | Section 4.1
Status-Line Section 5.1
Request = Request-Line ; Section 4.1 message-header = *( general-header
*( general-header
| request-header | request-header
| entity-header ) | entity-header )
CRLF
[ message-body ]
Response = Status-Line ; Section 5.1
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ]
In the interest of robustness, any leading empty line(s) MUST be In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins ignored. In other words, if the Request or Response message begins
with a CRLF, the CRLF should be ignored. with a CRLF, the CRLF should be ignored.
4 Request 4 Request
The Request message format is shown below: The Request message format is shown below:
general-header = Call-ID ; Section 6.11
| Date ; Section 6.14
| Expires ; Section 6.15
| From ; Section 6.16
| Sequence ; Section 6.26
| Via ; Section 6.31
entity-header = Content-Length ; Section 6.12
| Content-Type ; Section 6.13
request-header = Accept ; Section 6.6
| Accept-Language ; Section 6.7
| Authorization ; Section 6.9
| Organization ; Section 6.18
| Priority ; Section 6.20
| Proxy-Authorization ; Section 6.22
| Reach ; Section 6.24
| Subject ; Section 6.28
| To ; Section 6.29
| User-Agent ; Section 6.30
response-header = Location ; Section 6.17
| Proxy-Authenticate ; Section 6.21
| Public ; Section 6.23
| Retry-After ; Section 6.25
| Server ; Section 6.27
| Warning ; Section 6.32
| WWW-Authenticate ; Section 6.33
Table 1: SIP headers
Request = Request-Line ; Section 4.1 Request = Request-Line ; Section 4.1
*( general-header *( general-header
| request-header | request-header
| entity-header ) | entity-header )
CRLF CRLF
[ message-body ] ; Section 8 [ message-body ] ; Section 8
4.1 Request-Line 4.1 Request-Line
The Request-Line begins with a method token, followed by the The Request-Line begins with a method token, followed by the
Request-URI and the protocol version, and ending with CRLF. The Request-URI and the protocol version, and ending with CRLF. The
elements are separated by SP characters. No CR or LF are allowed elements are separated by SP characters. No CR or LF are allowed
except in the final CRLF sequence. except in the final CRLF sequence.
Request-Line = Method SP Request-URI SP SIP-Version CRLF general-header = Call-ID ; Section 6.12
| CSeq ; Section 6.26
| Date ; Section 6.15
| Expires ; Section 6.16
| From ; Section 6.17
| Via ; Section 6.33
entity-header = Content-Length ; Section 6.13
| Content-Type ; Section 6.14
request-header = Accept ; Section 6.6
| Accept-Language ; Section 6.7
| Authorization ; Section 6.10
| Call-Disposition ; Section 6.11
| Organization ; Section 6.19
| Priority ; Section 6.20
| Proxy-Authorization ; Section 6.22
| Require ; Section 6.24
| Subject ; Section 6.28
| To ; Section 6.31
| User-Agent ; Section 6.32
response-header = Location ; Section 6.18
| Proxy-Authenticate ; Section 6.21
| Public ; Section 6.23
| Retry-After ; Section 6.25
| Server ; Section 6.27
| Unsupported ; Section 6.29
| Warning ; Section 6.34
| WWW-Authenticate ; Section 6.35
4.1.1 Methods Table 2: SIP headers
The following methods are defined: Request-Line ___ Method SP Request-URI SP SIP-Version CRLF
method = "INVITE" | "CONNECTED" | "OPTIONS" | "BYE" 4.2 Methods
| "REGISTER" | "UNREGISTER"
INVITE: The user or service is being invited to participate in the The methods are defined below. Methods that are not supported by a
session. This method MUST be supported by a SIP server. proxy or redirect server SHOULD be treated by that server as if they
were an INVITE method and forwarded accordingly.
CONNECTED: A CONNECTED request confirms that the client has received Methods that are not supported by a user agent server should cause a
a successful response to an INVITE request. See Section 11 for "501 Not Implemented" response to be returned (Section 7).
details. This method MUST be supported by a SIP server.
OPTIONS: The client is being queried as to its capabilities. A server method = "INVITE" | "ACK" | "OPTIONS"
that believes it can contact the user, such as a user agent | "BYE" | "REGISTER" | "UNREGISTER"
where the user is logged in and has been recently active, MAY
respond to this request with a capability set. Support of this
method is OPTIONAL.
BYE: The client indicates to the server that it wishes to abort the 4.2.1 INVITE
call attempt. The leaving party can use a Location header field
to indicate that the recipient of request should contact the The INVITE method indicates that the user or service is being
named address. This implements the "call transfer" telephony invited to participate in a session. The message body contains a
functionality. A client SHOULD also use this method to indicate description of the session the callee is being invited to. For two-
to the callee that it wishes to abort an on-going call attempt. 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
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 With UDP, the caller has no other way to signal her intent
to drop the call attempt and the callee side will keep to drop the call attempt and the callee side will keep
"ringing". When using TCP, a client MAY also close the "ringing". When using TCP, a client MAY also close the
connection to abort a call attempt. Support of this method connection to abort a call attempt. Support of this method
is OPTIONAL. is OPTIONAL.
REGISTER: A client uses the REGISTER method to register the address Support of this method is OPTIONAL.
listed in the request line to a SIP server. In the future, the
server MAY use the source address and port to forward SIP 4.2.5 REGISTER
requests to. A server SHOULD silently drop the registration
after one hour, unless refreshed by the client. A server may set A client uses the REGISTER method to register the address listed in
or lower or higher refresh interval and indicate the interval the request line to a SIP server. The host part of the request-URI
through the Expires header (Section 6.15). A single address (if SHOULD correspond to (one of the aliases of) name of the server or to
host-independent) may be registered from several different the domain that it represents, if location-independent. After
clients. Support of this method is OPTIONAL. 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 Beyond its use as a simple location service, this method is
needed if there are several SIP servers on a single host, needed if there are several SIP servers on a single host,
so that some cannot use the default port number. Each such so that some cannot use the default port number. Each such
server would register with a server for the administrative server would register with a server for the administrative
domain. domain.
UNREGISTER: A client cancels an existing registration established for 4.2.6 UNREGISTER
the Request-URI with REGISTER with the UNREGISTER method. If
it unregisters a Request-URI unknown to the servers, the server
returns a 200 (OK) response. Support of this method is OPTIONAL.
BYE and REGISTER are experimental and need to be discussed.
Methods that are not supported by a proxy server SHOULD be treated by
that proxy as if they were an INVITE method, and relayed through
unchanged or cause a redirection as appropriate.
Methods that are not supported by a server should cause a "501 Not A client cancels an existing registration established for the
Implemented" response to be returned (Section 7). 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.
4.1.2 Request-URI 4.3 Request-URI
The Request-URI field is a SIP URL as described in Section 2 or a The Request-URI field is a SIP URL as described in Section 2 or a
general URI. It indicates the user or service that this request is general URI. It indicates the user or service that this request is
being addressed to. Unlike the To field, the Request-URI field may 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 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 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 alias, and then use this username as part of the Request-URI field
of requests it generates. of requests it generates.
If a SIP server receives a request contain a URI indicating a scheme 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 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 return a "400 Bad Request" response. It MUST do this even if the To
field contains a scheme it does understand. field contains a scheme it does understand.
4.1.3 SIP Version 4.3.1 SIP Version
Both request and response messages include the version of SIP in use, Both request and response messages include the version of SIP in use,
and basically follow [H3.1], with HTTP replaced by SIP. To be and basically follow [H3.1], with HTTP replaced by SIP. To be
conditionally compliant with this specification, applications sending conditionally compliant with this specification, applications sending
SIP messages MUST include a SIP-Version of "SIP/2.0". 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
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 5 Response
After receiving and interpreting a request message, the recipient After receiving and interpreting a request message, the recipient
responds with a SIP response message. The response message format is responds with a SIP response message. The response message format is
shown below: shown below:
Response = Status-Line ; Section 5.1 Response = Status-Line ; Section 5.1
*( general-header *( general-header
| response-header | response-header
| entity-header ) | entity-header )
CRLF CRLF
[ message-body ] ; Section 8 [ message-body ] ; Section 8
[H6] applies except that HTTP-Version is replaced by SIP-Version. [H6] applies except that HTTP-Version is replaced by SIP-Version.
Also, SIP defines additional response codes and does not use some Also, SIP defines additional response codes and does not use some
HTTP codes. HTTP codes.
5.1 Status-Line 5.1 Status-Line
The first line of a Response message is the Status-Line, consisting The first line of a Response message is the Status-Line, consisting
of the protocol version ((Section 4.1.3) followed by a numeric of the protocol version ((Section 4.3.1) followed by a numeric
Status-Code and its associated textual phrase, with each element Status-Code and its associated textual phrase, with each element
separated by SP characters. No CR or LF is allowed except in the separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence. final CRLF sequence.
Status-Line = SIP-version SP Status-Code SP Reason-Phrase Status-Line ___ SIP-version SP Status-Code SP Reason-Phrase CRLF
CRLF
5.1.1 Status Codes and Reason Phrases 5.1.1 Status Codes and Reason Phrases
The Status-Code is a 3-digit integer result code that indicates the The Status-Code is a 3-digit integer result code that indicates the
outcome of the attempt to understand and satisfy the request. The outcome of the attempt to understand and satisfy the request. The
Reason-Phrase is intended to give a short textual description of the Reason-Phrase is intended to give a short textual description of the
Status-Code. The Status-Code is intended for use by automata, Status-Code. The Status-Code is intended for use by automata,
whereas the Reason-Phrase is intended for the human user. The client whereas the Reason-Phrase is intended for the human user. The client
is not required to examine or display the Reason-Phrase. is not required to examine or display the Reason-Phrase.
We provide an overview of the Status-Code below, and provide full We provide an overview of the Status-Code below, and provide full
definitions in section 7. The first digit of the Status-Code defines definitions in section 7. The first digit of the Status-Code defines
the class of response. The last two digits do not have any the class of response. The last two digits do not have any
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Presented below are the individual values of the numeric response Presented below are the individual values of the numeric response
codes, and an example set of corresponding reason phrases for codes, and an example set of corresponding reason phrases for
SIP/2.0. These reason phrases are only recommended; they may be SIP/2.0. These reason phrases are only recommended; they may be
replaced by local equivalents without affecting the protocol. Note replaced by local equivalents without affecting the protocol. Note
that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response 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 codes in the range starting at x80 to avoid conflicts with newly
defined HTTP response codes, and extends these response codes in the defined HTTP response codes, and extends these response codes in the
6xx range. 6xx range.
Status-Code = Informational Fig. 3
| Success Fig. 3
| Redirection Fig. 4
| Client-Error Fig. 5
| Server-Error Fig. 6
| Global-Failure Fig. 7
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
Informational = "100" ; Trying
| "180" ; Ringing
| "181" ; Queued
Success = "200" ; OK
Figure 3: Informational and success status codes
Redirection = "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "305" ; Use Proxy
| "380" ; Alternative Service
Figure 4: Redirection status codes
SIP response codes are extensible. SIP applications are not required SIP response codes are extensible. SIP applications are not required
to understand the meaning of all registered response codes, though to understand the meaning of all registered response codes, though
such understanding is obviously desirable. However, applications MUST such understanding is obviously desirable. However, applications MUST
understand the class of any response code, as indicated by the first understand the class of any response code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the digit, and treat any unrecognized response as being equivalent to the
x00 response code of that class, with the exception that an x00 response code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if a client unrecognized response MUST NOT be cached. For example, if a client
receives an unrecognized response code of 431, it can safely assume receives an unrecognized response code of 431, it can safely assume
that there was something wrong with its request and treat the that there was something wrong with its request and treat the
response as if it had received a 400 response code. In such cases, 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 user agents SHOULD present to the user the message body returned with
the response, since that message body is likely to include human- the response, since that message body is likely to include human-
readable information which will explain the unusual status. readable information which will explain the unusual status.
6 Header Field Definitions 6 Header Field Definitions
Client-Error = "400" ; Bad Request
SIP header fields are similar to HTTP header fields in both syntax
and semantics [H4.2], [H14]. In general the ordering of the header
fields is not of importance (with the exception of Via fields, see
below), but proxies MUST NOT reorder or otherwise modify header
fields other than by adding a new Via field. This allows an
authentication field to be added after the Via fields that will not
be invalidated by proxies.
To, From, and Call-ID header MUST be present in each request with
method INVITE. The Content-Type and Content-Length headers are
required when there is a valid message body (of non-zero length)
associated with the message (Section 8).
A server MUST understand the PEP-Require header.
Other headers may be added as required; a server MAY ignore headers
that it does not understand. A compact form of these header fields is
Status-Code = "100" ; Trying
| "180" ; Ringing
| "200" ; OK
| "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "305" ; Use Proxy
| "380" ; Alternative Service
| "400" ; Bad Request
| "401" ; Unauthorized | "401" ; Unauthorized
| "402" ; Payment Required | "402" ; Payment Required
| "403" ; Forbidden | "403" ; Forbidden
| "404" ; Not Found | "404" ; Not Found
| "405" ; Method Not Allowed | "405" ; Method Not Allowed
| "407" ; Proxy Authentication Required | "407" ; Proxy Authentication Required
| "408" ; Request Timeout | "408" ; Request Timeout
| "409" ; Conflict | "409" ; Conflict
| "410" ; Gone | "410" ; Gone
| "411" ; Length Required | "411" ; Length Required
| "412" ; Precondition Failed | "412" ; Precondition Failed
| "413" ; Request Message Body Too Large | "413" ; Request Message Body Too Large
| "414" ; Request-URI Too Large | "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type | "415" ; Unsupported Media Type
| "420" ; Bad Extension | "420" ; Bad Extension
| "480" ; Temporarily not available | "480" ; Temporarily not available
| "500" ; Internal Server Error | "481" ; Invalid Call-ID
| "482" ; Loop Detected
Figure 5: Client error status codes
Server-Error = "500" ; Internal Server Error
| "501" ; Not Implemented | "501" ; Not Implemented
| "502" ; Bad Gateway | "502" ; Bad Gateway
| "503" ; Service Unavailable | "503" ; Service Unavailable
| "504" ; Gateway Timeout | "504" ; Gateway Timeout
| "505" ; SIP Version not supported | "505" ; SIP Version not supported
| "600" ; Busy
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
Global-Failure | "600" ; Busy
| "603" ; Decline | "603" ; Decline
| "604" ; Does not exist anywhere | "604" ; Does not exist anywhere
| "606" ; Not Acceptable | "606" ; Not Acceptable
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
Figure 3: Status Codes 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 also defined in Section 10 for use over UDP when the request has to
fit into a single packet and size is an issue. fit into a single packet and size is an issue.
6.1 General Header Fields 6.1 General Header Fields
There are a few header fields that have general applicability for There are a few header fields that have general applicability for
both request and response messages. These header fields apply only to both request and response messages. These header fields apply only to
the message being transmitted. the message being transmitted.
General-header field names can be extended reliably only in General-header field names can be extended reliably only in
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body may contain a transformed version of the message body. The body may contain a transformed version of the message body. The
original message body is referred to as the "entity". We retain the original message body is referred to as the "entity". We retain the
same terminology for header fields but usually refer to the "message same terminology for header fields but usually refer to the "message
body" rather then the entity as the two are the same in SIP. body" rather then the entity as the two are the same in SIP.
6.3 Request Header Fields 6.3 Request Header Fields
The request-header fields allow the client to pass additional The request-header fields allow the client to pass additional
information about the request, and about the client itself, to the information about the request, and about the client itself, to the
server. These fields act as request modifiers, with semantics server. These fields act as request modifiers, with semantics
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 equivalent to the parameters on a programming language method
invocation. invocation.
Request-header field names can be extended reliably only in Request-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of request- experimental header fields MAY be given the semantics of request-
header fields if all parties in the communication recognize them to header fields if all parties in the communication recognize them to
be request-header fields. Unrecognized header fields are treated as be request-header fields. Unrecognized header fields are treated as
entity-header fields. entity-header fields.
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combination with a change in the protocol version. However, new or combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response- experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as be response-header fields. Unrecognized header fields are treated as
entity-header fields. entity-header fields.
6.5 Header Field Format 6.5 Header Field Format
Header fields ( general-header, request-header, response-header, and Header fields ( general-header, request-header, response-header, and
entity-header) follow the same generic header format as that given in entity-header) follow the same generic header format as that given in
Section 3.1 of RFC 822 [22]. Section 3.1 of RFC 822 [24].
Each header field consists of a name followed by a colon (":") and Each header field consists of a name followed by a colon (":") and
the field value. Field names are case-insensitive. The field value the field value. Field names are case-insensitive. The field value
may be preceded by any amount of leading white space (LWS), though a may be preceded by any amount of leading white space (LWS), though a
single space (SP) is preferred. Header fields can be extended over single space (SP) is preferred. Header fields can be extended over
multiple lines by preceding each extra line with at least one SP or multiple lines by preceding each extra line with at least one SP or
horizontal tab (HT). Applications SHOULD follow HTTP "common form" horizontal tab (HT). Applications SHOULD follow HTTP "common form"
when generating these constructs, since there might exist some when generating these constructs, since there might exist some
implementations that fail to accept anything beyond the common forms. implementations that fail to accept anything beyond the common forms.
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subsequent field-value to the first, each separated by a comma. The 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 order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field therefore significant to the interpretation of the combined field
value, and thus a proxy MUST NOT change the order of these field value, and thus a proxy MUST NOT change the order of these field
values when a message is forwarded. values when a message is forwarded.
Field names are not case-sensitive, although their values may be. Field names are not case-sensitive, although their values may be.
6.6 Accept 6.6 Accept
See [H14.1]. This request header field is used only with the OPTIONS See [H14.1] for syntax. This request header field is used only with
request to indicate what description formats are acceptable. the OPTIONS and INVITE request methods to indicate what description
formats are acceptable in the response.
Example: Example:
Accept: application/sdp;level=1, application/x-private Accept: application/sdp;level=1, application/x-private
6.7 Accept-Language 6.7 Accept-Language
See [H14.4]. The Accept-Language request header can be used to allow See [H14.4] for syntax. The Accept-Language request header can be
the client to indicate to the server in which language it would used to allow the client to indicate to the server in which language
prefer to receive reason phrases. This may also be used as a hint by it would prefer to receive reason phrases. This may also be used as a
the proxy as to which destination to connect the call to (e.g., for hint by the proxy as to which destination to connect the call to
selecting a human operator). (e.g., for selecting a human operator).
Example: Example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7 Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.8 Allow 6.8 Allow
See [H14.7]. See [H14.7].
6.9 Authorization 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).
Also ___ "Also" ":" 1#( SIP-URL ) [ comment ]
Example:
Also: sip://jones@foo.com, sip://mueller@bar.edu
6.10 Authorization
See [H14.8]. See [H14.8].
6.10 Authentication 6.11 Call-Disposition
Authentication fields provide a digital signature of the remaining The Call-Disposition request header field allows the client to
fields for authentication purposes. They are not yet defined The use indicate how the server is to handle the call. The following options
of authentication headers is optional. If used, authentication can be used singly or in combination:
headers MUST be added to the header after the Via fields and before
the rest of the fields.
HS: Should probably re-use S/MIME here rather than invent all: If the user part of the SIP request address identifies a group
our own. Maybe better to fold into Authorization field. 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.
6.11 Call-ID 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.)
The Call-ID uniquely identifies a particular invitation. Note that a queue: If the called party is temporarily unreachable, e.g., because
single multimedia conference may give rise to several calls, e.g., if it is in another call, the caller can indicate that it wants to
a user invites several different people. Calls to different callee have its call queued rather than rejected immediately. If the
MUST always use different Call-IDs unless they are the result of a call is queued, the server returns "181 Queued" (see Section
proxy server "forking" a single request. 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
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 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 be globally unique for the duration of the request. Using the
initiator's IP-address, process id, and instance (if more than one initiator's IP-address, process id, and instance (if more than one
request is being made simultaneously) satisfies this requirement. request is being made simultaneously) satisfies this requirement.
The form local-id@host is recommended, where host is either the The form local-id@host is recommended, where host is either the
fully qualified domain name or a globally routable IP address, and fully qualified domain name or a globally routable IP address, and
local-id depends on the application and operating system of the host, 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 but is an ID that can be guaranteed to be unique during this session
initiation request. initiation request.
Call-ID = ( "Call-ID" | "i" ) ":" atom "@" host Call-ID ___ ( "Call-ID" | "i" ) ":" atom "@" host
Example: Example:
Call-ID: 9707211351.AA08181@foo.bar.com Call-ID: 9707211351.AA08181@foo.bar.com
6.12 Content-Length 6.13 Content-Length
The Content-Length entity-header field indicates the size of the The Content-Length entity-header field indicates the size of the
message-body, in decimal number of octets, sent to the recipient. message-body, in decimal number of octets, sent to the recipient.
Content-Length = "Content-Length" ":" 1*DIGIT Content-Length = "Content-Length" ":" 1*DIGIT
An example is An example is
Content-Length: 3495 Content-Length: 3495
Applications SHOULD use this field to indicate the size of the Applications SHOULD use this field to indicate the size of the
message-body to be transferred, regardless of the media type 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 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 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 header MAY be omitted or set to zero. Section 8 describes how to
determine the length of the message body. determine the length of the message body.
6.13 Content-Type 6.14 Content-Type
The Content-Type entity-header field indicates the media type of the The Content-Type entity-header field indicates the media type of the
message-body sent to the recipient. message-body sent to the recipient.
Content-Type = "Content-Type" ":" media-type Content-Type ___ "Content-Type" ":" media-type
An example of the field is An example of the field is
Content-Type: application/sdp Content-Type: application/sdp
6.14 Date 6.15 Date
See [H14.19]. General header field. See [H14.19].
The Date header field is useful for simple devices without The Date header field is useful for simple devices without
their own clock. their own clock.
6.15 Expires 6.16 Expires
The Expires header field gives the date/time after which the The Expires entity-header field gives the date and time after which
registration expires. the message content expires.
This header field is currently defined only for the REGISTER and This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a response-header field and INVITE methods. For REGISTER, it is a request and response-header
allows the server to indicate when the client has to re-register. For field and allows the client to indicate how long the registration
INVITE, it is a request-header with which the callee can limit the should be valid; the server uses it to indicate when the client has
validity of an invitation. (For example, if a client wants to limit to re-register. The server's choice overrides that of the client. The
how long a search should take at most or when a conference being server MAY choose a shorter time interval than that requested by the
invited to is time-limited. A user interface may take this is as a client, but SHOULD not choose a longer one.
hint to leave the invitation window on the screen even if the user is
not currently at the workstation.) 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 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 number of seconds (in decimal), measured from the receipt of the
request. request.
Expires = "Expires" ":" ( HTTP-date | delta-seconds ) Expires ___ "Expires" ":" ( HTTP-date | delta-seconds )
Two example of its use are Two example of its use are
Expires: Thu, 01 Dec 1994 16:00:00 GMT Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5 Expires: 5
6.16 From 6.17 From
Requests MUST and responses SHOULD contain a From header field, Requests MUST and responses SHOULD contain a From header field,
indicating the invitation initiator. The field MUST be a SIP URL as indicating the invitation initiator. The field MUST be a SIP URL as
defined in Section 2. Only a single initiator and a single invited 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 user are allowed to be specified in a single SIP request. The sense
of To and From header fields is maintained from request to 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 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 request then it is MUST also be sip://bob@example.edu in the response
to that request. to that request.
The From field is a URL and not a simple SIP address (Section 1.6 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 address to allow a gateway to relay a call into a SIP request and
still produce an appropriate From field. An example might be a still produce an appropriate From field.
telephone call relayed into a SIP request where the from field might
contain a phone:// URL. Normally however this field will contain a
sip:// URL in either the long or short form.
If a SIP agent or proxy receives a request sourced From a URL
indicating a scheme other that SIP that is unknown to it, this MUST
NOT be treated as an error.
From = ( "From" | "f" ) ":" *1( ( SIP-URL | URL ) [ comment From ___ ( "From" | "f" ) ":" *1( ( SIP-URL | URL ) [ comment ] )
] )
Example: Examples:
From: mjh@isi.edu (Mark Handley) From: agb@bell-telephone.com (A. G. Bell)
From: +12125551212@server.phone2net.com
6.17 Location 6.18 Location
The Location response header can be used with a 2xx or 3xx response The Location response header can be used with a 2xx or 3xx response
codes to indicate a new location to try. It contains a SIP URL giving codes to indicate a new location to try. It contains a URL giving the
the new location or username to try, or may simply specify addition new location or username to try, or may simply specify additional
transport parameters. For example, a "301 Moved Permanently" response transport parameters. A "301 Moved Permanently" or "302 Moved
SHOULD contain a Location field containing the SIP URL giving the Temporarily" response SHOULD contain a Location field containing the
new location and username to try. However, a "302 Moved Temporarily" URL giving a new address to try. A 301 or 302 response may also give
MAY give simply the same location and username that was being tried the same location and username that was being tried but specify
but specify additional transport parameters such as a multicast additional transport parameters such as a multicast address to try or
address to try or a change of transport from UDP to TCP or vice a change of SIP transport from UDP to TCP or vice versa.
versa.
A user agent or redirect server sending a definitive, positive A user agent or redirect server sending a definitive, positive
response (2xx), SHOULD insert a Location response header indicating response (2xx), SHOULD insert a Location response header indicating
the SIP address under which it is reachable most directly for future 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 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 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.
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.
Location = ( "Location" | "m" ) ( SIP-URL | URL ) Location = ( "Location" | "m" ) ( SIP-URL | URL )
*( ";" location-params ) *( ";" location-params )
extension-name = token extension-name = token
extension-value = *( token | quoted-string | LWS | extension-specials) extension-value = *( token | quoted-string | LWS | extension-specials)
extension-specials = < any element of tspecials except <"> > extension-specials = < any element of tspecials except <"> >
language-tag = < see [H3.10] > language-tag = < see [H3.10] >
service-tag = "fax" | "IP" | "PSTN" | "ISDN" | "pager" | "voice-mail priority-tag = "urgent" | "normal" | "non-urgent"
| "attendant" service-tag = "fax" | "IP" | "PSTN" | "ISDN" | "pager"
media-tag = < see SDP: "audio" | "video" | ... media-tag = < see SDP: "audio" | "video" | "email" ...
feature-list = to be determined feature-list = "voice-mail" | "attendant"
location-params = "q" "=" qvalue location-params = "q" "=" qvalue
| "mobility" "=" ( "fixed" | "mobile" )
| "class" "=" ( "personal" | "business" ) | "class" "=" ( "personal" | "business" )
| "description" "=" quoted-string
| "duplex" "=" ( "full" | "half" |
"receive-only" | "send-only" )
| "features" "=" 1# feature-list
| "language" "=" 1# language-tag | "language" "=" 1# language-tag
| "service" "=" 1# service-tag
| "media" "=" 1# media-tag | "media" "=" 1# media-tag
| "features" "=" 1# feature-list | "mobility" "=" ( "fixed" | "mobile" )
| "description" "=" quoted-string | "priority" "=" 1# priority-tag
| "duplex" "=" ( "full" | "half" | "receive-only" | | "service" "=" 1# service-tag
"send-only" )
| extension-attributes | extension-attributes
extension-attribute = extension-name "=" extension-value extension-attribute = extension-name "=" extension-value
Examples: Examples:
Location: sip://hgs@erlang.cs.columbia.edu ;service=IP,voice-mail Location: sip://watson@worcester.bell-telephone.com ;service=IP,voice-mail
;media=audio ;duplex=full ;q=0.7 ;media=audio ;duplex=full ;q=0.7;
Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed; Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
language=en,es,iw ;q=0.5 language=en,es,iw ;q=0.5
Location: phone://1-800-555-1212 ; service=pager;mobility=mobile; Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
duplex=send-only;media=text; q=0.1 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- Attributes which are unknown should be omitted. New tags for class-
tag and service-tag can be registered with IANA. The media tag uses 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 Internet media types, e.g., audio, video, application/x-wb, etc. This
is meant for indicating general communication capability, sufficient is meant for indicating general communication capability, sufficient
for the caller to choose an appropriate address. for the caller to choose an appropriate address.
6.18 Organization 6.19 Organization
The Organization request-header fields conveys the name of the The Organization request-header fields conveys the name of the
organization to which the callee belongs. It may be inserted by organization to which the callee belongs. It may be inserted by
proxies at the boundary of an organization and may be used by client proxies at the boundary of an organization and may be used by client
software to filter calls. software to filter calls.
6.19 PEP
This corresponds to the PEP header in the "Protocol Extension
Protocol" defined in RFC XXXX. The Protocol Extension Protocol (PEP)
is an extension mechanism designed to accommodate dynamic extension
of applications such as SIP clients and servers by software
components. The PEP general header declares new headers and whether
an application must or may understand them. Servers MUST parse this
field and MUST return "420 Bad Extension" when there is a PEP
extension of strength "must" (see RFC XXXX) that they do not
understand.
6.20 Priority 6.20 Priority
The priority request header signals the urgency of the call to the The priority request header signals the urgency of the call to the
callee. callee.
Priority = "Priority" ":" priority-value Priority = "Priority" ":" priority-value
priority-value = "urgent" | "normal" | "non-urgent" priority-value = "urgent" | "normal" | "non-urgent"
Example: Example:
skipping to change at page 30, line 9 skipping to change at page 38, line 20
See [H14.33]. See [H14.33].
6.22 Proxy-Authorization 6.22 Proxy-Authorization
See [H14.34]. See [H14.34].
6.23 Public 6.23 Public
See [H14.35]. See [H14.35].
6.24 Reach 6.24 Require
The Reach request header field allows the client to indicate whether The Require header is used by clients to query the server about
it wants to reach the group identified by the user part of the options that it may or may not support. The server MUST respond to
address (value "all") or the first available individual (value this header by returning status code "420 Bad Extension" and list
"first"). If not present, a value of "first" is implied. The "do- those options it does not understand in the Unsupported header.
not-forward" request prohibits proxies from forwarding the call to
another individual (e.g., the call is personal or the caller does not
want to be shunted to a secretary if the line is busy.) Section 1.6
describes the behavior of proxy servers when resolving group aliases.
Reach = "Reach" ":" 1#( "first" | "all" ) ( "do-not- Require ___ "Require" ":" 1#option-tag
forward" )
Example: Example:
Reach: first, do-not-forward C->S: INVITE sip:watson@bell-telephone.com SIP/2.0
Require: com.example.billing
Payment: sheep_skins, conch_shells
HS: This header is experimental. 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.
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 6.25 Retry-After
The Retry-After response header field can be used with a "503 The Retry-After response header field can be used with a "503
Service Unavailable" response to indicate how long the service is Service Unavailable" response to indicate how long the service is
expected to be unavailable to the requesting client and with a "404 expected to be unavailable to the requesting client and with a "404
Not Found" or "451 Busy" response to indicate when the called party Not Found", "600 Busy", "603 Decline" response to indicate when the
may be available again. The value of this field can be either an called party may be available again. The value of this field can be
HTTP-date or an integer number of seconds (in decimal) after the time either an HTTP-date or an integer number of seconds (in decimal)
of the response. 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 Retry-After ___ "Retry-After" ":" ( HTTP-date | delta-seconds )
) [ comment ] [ ";duration" "=" delta-seconds
Two examples of its use are Examples of its use are
Retry-After: Mon, 21 Jul 1997 18:48:34 GMT 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 Retry-After: 120
In the latter example, the delay is 2 minutes.
6.26 Sequence 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.
The Sequence header field MAY be added by a SIP client making a 6.26 CSeq
request if it needs to distinguish responses to several consecutive
requests sent with the same Call-ID. A Sequence field contains a
single decimal sequence number chosen by the requesting client.
Consecutive different requests made with the same Call-ID MUST
contain strictly monotonically increasing sequence numbers although
the sequence space MAY NOT be contiguous. A server responding to a
request containing a sequence number MUST echo the sequence number
back in the response.
Sequence = "Sequence" ":" 1*DIGIT 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
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.
Sequence header fields are NOT needed for SIP requests using the CSeq = "CSeq" ":" 1*DIGIT
INVITE or OPTIONS methods but may be needed for future methods.
CSeq header fields are NOT needed for SIP requests using the INVITE
or OPTIONS methods but may be needed for future methods.
Example: Example:
Sequence: 4711 CSeq: 4711
6.27 Server 6.27 Server
See [H14.39]. See [H14.39].
6.28 Subject 6.28 Subject
This is intended to provide a summary, or indicate the nature, of the This is intended to provide a summary, or indicate the nature, of the
call, allowing call filtering without having to parse the session call, allowing call filtering without having to parse the session
description. (Also, the session description may not necessarily use description. (Also, the session description may not necessarily use
the same subject indication as the invitation.) the same subject indication as the invitation.)
Subject = ( "Subject" | "s" ) ":" *text Subject ___ ( "Subject" | "s" ) ":" *text
Example: Example:
Subject: Tune in - they are talking about your work! Subject: Tune in - they are talking about your work!
6.29 To 6.29 Unsupported
The Unsupported response header lists the features not supported by
the server.
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 The To request header field specifies the invited user, with the
same SIP URL syntax as the From field. same SIP URL syntax as the From field.
To = ( "To" | "t" ) ":" ( SIP-URL | URL ) [ comment ] To = ( "To" | "t" ) ":" ( SIP-URL | URL ) [ comment ]
If a SIP server receives a request destined To a URL indicating a 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 scheme other than SIP and that is unknown to it, the server returns a
"400 bad request" response. "400 bad request" response.
Example: Example:
To: sip://operator@cs.columbia.edu (The Operator) To: sip://operator@cs.columbia.edu (The Operator)
6.30 User-Agent 6.32 User-Agent
See [H14.42]. See [H14.42].
6.31 Via 6.33 Via
The Via field indicates the path taken by the request so far. This The Via field indicates the path taken by the request so far. This
prevents request looping and ensures replies take the same path as prevents request looping and ensures replies take the same path as
the requests, which assists in firewall traversal and other unusual the requests, which assists in firewall traversal and other unusual
routing situations. routing situations.
In the request path, an initiator MUST add its own Via field to each The client originating the request MUST insert a Via field
request. This Via field MUST be the first field in the request. Each containing its address to the request. Each subsequent proxy server
subsequent client or proxy that sends the message onwards MUST add that sends the request onwards MUST add its own additional Via
its own additional Via field, which MUST be added before any field, which MUST be added before any existing Via fields.
existing Via fields. Additionally, if the message goes to a Additionally, if the message goes to a multicast address, an extra
multicast address, an extra Via field is added before all the others Via field is added before all the others giving the multicast address
giving the multicast address and TTL. 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 In the return path, Via fields are processed by a proxy or client
according to the following rules: according to the following rules:
o If the first Via field in the reply received is the client's o If the first Via field in the reply received is the client's
or server's local address, remove the Via field and process or server's local address, remove the Via field and process
the reply. the reply.
o If the first Via field in a reply you are going to send is a oIf the first Via field in a reply is a multicast address,
multicast address, remove that Via field before sending to the remove that Via field before sending to the multicast address.
multicast address.
These rules ensure that a client or proxy server only has to check
the first Via field in a reply to see if it needs processing.
When a reply passes through a proxy on the reverse path, that proxies These rules ensure that a proxy server only has to check the first
Via field MUST be removed from the reply. Via field in a reply to see if it needs processing.
The format for a Via header is: The format for a Via header is:
Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] ) *( ";" via-params ) [ comment ] )
via-params = "ttl" "=" ttl via-params = "ttl" "=" ttl
| "fanout"
sent-protocol = [ protocol-name "/" ] protocol-version sent-protocol = [ protocol-name "/" ] protocol-version
[ "/" transport ] [ "/" transport ]
protocol-name = "SIP" | token protocol-name = "SIP" | token
protocol-version = token protocol-version = token
transport = "UDP" | "TCP" transport = "UDP" | "TCP"
sent-by = host [ ":" port ] sent-by = host [ ":" port ]
ttl = 1*3DIGIT ; 0 to 255 ttl = 1*3DIGIT ; 0 to 255
The "ttl" parameter is included only if the address is a multicast The "ttl" parameter is included only if the address is a multicast
address. The "fanout" parameter indicates that this proxy has address.
initiated several connection attempts and that subsequent proxies
should not do the same.
Example: Example:
Via: SIP/2.0/UDP first.example.com:4000 ;fanout Via: SIP/2.0/UDP first.example.com:4000
6.32 Warning 6.34 Warning
The Warning response-header field is used to carry additional The Warning response-header field is used to carry additional
information about the status of a response. Warning headers are sent information about the status of a response. Warning headers are sent
with responses using: with responses and have the following format:
Warning = "Warning" ":" 1#warning-value Warning = "Warning" ":" 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text warning-value = warn-code SP warn-agent SP warn-text
warn-code = 2DIGIT warn-code = 2DIGIT
warn-agent = ( host [ ":" port ] ) | pseudonym warn-agent = ( host [ ":" port ] ) | pseudonym
; the name or pseudonym of the server adding ; the name or pseudonym of the server adding
; the Warning header, for use in debugging ; the Warning header, for use in debugging
warn-text = quoted-string warn-text = quoted-string
A response may carry more than one Warning header. A response may carry more than one Warning header.
The warn-text should be in a natural language and character set that 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 is most likely to be intelligible to the human user receiving the
response. This decision may be based on any available knowledge, such 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 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 request, the Content-Language field in a response, etc. The default
language is English and the default character set is ISO- 8859-1. language is English.
Any server may add Warning headers to a response. New Warning Any server may add Warning headers to a response. New Warning
headers should be added after any existing Warning headers. A proxy headers should be added after any existing Warning headers. A proxy
server MUST NOT delete any Warning header that it received with a server MUST NOT delete any Warning header that it received with a
response. response.
When multiple Warning headers are attached to a response, the user When multiple Warning headers are attached to a response, the user
agent SHOULD display as many of them as possible, in the order that 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 they appear in the response. If it is not possible to display all of
the warnings, the user agent should follow these heuristics: the warnings, the user agent should follow these heuristics:
skipping to change at page 34, line 40 skipping to change at page 44, line 14
warn-codes and warn-agents. warn-codes and warn-agents.
Systems that generate multiple Warning headers should order them Systems that generate multiple Warning headers should order them
with this user agent behavior in mind. with this user agent behavior in mind.
Example: Example:
Warning: 606.4 isi.edu Multicast not available Warning: 606.4 isi.edu Multicast not available
Warning: 606.2 isi.edu Incompatible protocol (RTP/XXP) Warning: 606.2 isi.edu Incompatible protocol (RTP/XXP)
6.33 WWW-Authenticate 6.35 WWW-Authenticate
See [H14.46]. See [H14.46].
7 Status Code Definitions 7 Status Code Definitions
The response codes are consistent with, and extend, HTTP/1.1 response The response codes are consistent with, and extend, HTTP/1.1 response
codes. Not all HTTP/1.1 response codes are appropriate, and only codes. Not all HTTP/1.1 response codes are appropriate, and only
those that are appropriate are given here. Response codes not defined 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 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 response codes. Also, SIP defines a new class, 6xx. The default
skipping to change at page 35, line 31 skipping to change at page 45, line 5
7.1.1 100 Trying 7.1.1 100 Trying
Some further action is being taken (e.g., the request is being Some further action is being taken (e.g., the request is being
forwarded) but the user has not yet been located. forwarded) but the user has not yet been located.
7.1.2 180 Ringing 7.1.2 180 Ringing
The user agent or conference server has located a possible location The user agent or conference server has located a possible location
where the user has been recently and is trying to alert them. where the user has been recently and is trying to alert them.
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 7.2 Successful 2xx
The request was successful and MUST terminate a search. The request was successful and MUST terminate a search.
7.2.1 200 OK 7.2.1 200 OK
The request was successful in contacting the user, and the user has The request was successful in contacting the user, and the user has
agreed to participate. agreed to participate.
7.3 Redirection 3xx 7.3 Redirection 3xx
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The response SHOULD include an entity containing a list of resource The response SHOULD include an entity containing a list of resource
characteristics and location(s) from which the user or user agent can characteristics and location(s) from which the user or user agent can
choose the one most appropriate. The entity format is specified by choose the one most appropriate. The entity format is specified by
the media type given in the Content-Type header field. Depending the media type given in the Content-Type header field. Depending
upon the format and the capabilities of the user agent, selection of upon the format and the capabilities of the user agent, selection of
the most appropriate choice may be performed automatically. However, the most appropriate choice may be performed automatically. However,
this specification does not define any standard for such automatic this specification does not define any standard for such automatic
selection. selection.
The choices SHOULD also be listed as Location fields (Section 6.17). The choices SHOULD also be listed as Location fields (Section 6.18).
Unlike HTTP, the SIP response may contain several Location fields. Unlike HTTP, the SIP response may contain several Location fields.
User agents MAY use the Location field value for automatic User agents MAY use the Location field value for automatic
redirection or MAY ask the user to confirm a choice. redirection or MAY ask the user to confirm a choice.
7.3.2 301 Moved Permanently 7.3.2 301 Moved Permanently
The requesting client should retry on the new address given by the The requesting client should retry on the new address given by the
Location field because the user has permanently moved and the address 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 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 user. A 301 response MUST NOT suggest any of the hosts in the Via
path of the request as the user's new location. (Section 6.33) path of the request as the user's new location.
7.3.3 302 Moved Temporarily 7.3.3 302 Moved Temporarily
The requesting client should retry on the new address(es) given by 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 the Location header. A 302 response MUST NOT suggest any of the hosts
in the Via path of the request as the user's new location. 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 7.3.4 380 Alternative Service
The call was not successful, but alternative services are possible. The call was not successful, but alternative services are possible.
The alternative services are described in the message body of the The alternative services are described in the message body of the
response. response.
7.4 Request Failure 4xx 7.4 Request Failure 4xx
4xx responses are definite failure responses from a particular 4xx responses are definite failure responses from a particular
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The server did not understand the protocol extension specified with The server did not understand the protocol extension specified with
strength "must". strength "must".
7.4.10 480 Temporarily Unavailable 7.4.10 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is 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 currently unavailable (e.g., not logged in or logged in in such a
manner as to preclude communication with the callee). The response manner as to preclude communication with the callee). The response
may indicate a better time to call in the Retry-After header. The may indicate a better time to call in the Retry-After header. The
user may also be available elsewhere (unbeknownst to this host), user may also be available elsewhere (unbeknownst to this host),
thus, this response does terminate any searches. 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
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 7.5 Server Failure 5xx
5xx responses are failure responses given when a server itself has 5xx responses are failure responses given when a server itself has
erred. They are not definitive failures, and SHOULD NOT terminate a erred. They are not definitive failures, and SHOULD NOT terminate a
search if other possible locations remain untried. search if other possible locations remain untried.
7.5.1 500 Server Internal Error 7.5.1 500 Server Internal Error
The server encountered an unexpected condition that prevented it from The server encountered an unexpected condition that prevented it from
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Note: The existence of the 503 status code does not imply that a 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 server must use it when becoming overloaded. Some servers may wish to
simply refuse the connection. simply refuse the connection.
7.5.5 504 Gateway Timeout 7.5.5 504 Gateway Timeout
The server, while acting as a gateway, did not receive a timely The server, while acting as a gateway, did not receive a timely
response from the upstream server (e.g., a location server) it response from the upstream server (e.g., a location server) it
accessed in attempting to complete the request. accessed in attempting to complete the request.
7.6 Global Failures 7.6 Global Failures 6xx
6xx responses indicate that a server has definitive information about 6xx responses indicate that a server has definitive information about
a particular user, not just the particular instance indicated in the a particular user, not just the particular instance indicated in the
Request-URI. All further searches for this user are doomed to failure Request-URI. All further searches for this user are doomed to failure
and pending searches SHOULD be terminated. and pending searches SHOULD be terminated.
7.6.1 600 Busy 7.6.1 600 Busy
The callee's end system was contacted successfully but the callee is 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 busy and does not wish to take the call at this time. The response
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elsewhere will not yield any results. elsewhere will not yield any results.
7.6.4 606 Not Acceptable 7.6.4 606 Not Acceptable
The user's agent was contacted successfully but some aspects of the The user's agent was contacted successfully but some aspects of the
session profile (the requested media, bandwidth, or addressing style) session profile (the requested media, bandwidth, or addressing style)
were not acceptable. were not acceptable.
A "606 Not Acceptable" reply means that the user wishes to A "606 Not Acceptable" reply means that the user wishes to
communicate, but cannot adequately support the session described. The communicate, but cannot adequately support the session described. The
"604 Not Acceptable" reply MAY contain a list of reasons in a Warning "606 Not Acceptable" reply MAY contain a list of reasons in a Warning
header describing why the session described cannot be supported. header describing why the session described cannot be supported.
These reasons can be one or more of: These reasons can be one or more of:
606.1 Insufficient Bandwidth: The bandwidth specified in the session 606.1 Insufficient Bandwidth: The bandwidth specified in the session
description or defined by the media exceeds that known to be description or defined by the media exceeds that known to be
available. available.
606.2 Incompatible Protocol: One or more protocols described in the 606.2 Incompatible Protocol: One or more protocols described in the
request are not available. request are not available.
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8 SIP Message Body 8 SIP Message Body
The session description body gives details of the session the user is 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 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 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 by the Content-Length header field. If the body has undergone any
encoding (such as compression) then this MUST be indicated by the encoding (such as compression) then this MUST be indicated by the
Content-encoding header field, otherwise Content-encoding MUST be Content-encoding header field, otherwise Content-encoding MUST be
omitted. omitted.
If required, the session description can be encrypted using public
key cryptography, and then can also carry private session keys for
the session. If this is the case, four random bytes are added to the
beginning of the session description before encryption and are
removed after decryption but before parsing.
8.1 Body Inclusion 8.1 Body Inclusion
For a request message, the presence of a body is signaled by the 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 inclusion of a Content-Length header. A body may be included in a
request only when the request method allows one. request only when the request method allows one.
For response messages, whether or not a body is included is dependent For response messages, whether or not a body is included is dependent
on both the request method and the response message's response code. on both the request method and the response message's response code.
All 1xx informational responses MUST NOT include a body. All other All 1xx informational responses MUST NOT include a body. All other
responses MAY include a payload, although it may be of zero length. responses MAY include a payload, although it may be of zero length.
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1. Any response message which MUST NOT include a body (such as 1. Any response message which MUST NOT include a body (such as
the 1xx responses) is always terminated by the first empty the 1xx responses) is always terminated by the first empty
line after the header fields, regardless if any entity- line after the header fields, regardless if any entity-
header fields are present. header fields are present.
2. Otherwise, a Content-Length header MUST be present (this 2. Otherwise, a Content-Length header MUST be present (this
requirement differs from HTTP/1.1). Its value in bytes requirement differs from HTTP/1.1). Its value in bytes
represents the length of the message body. represents the length of the message body.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP. 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 Examples
9.1 Invitation 9.1 Invitation to Multimedia Conference
9.1.1 Request The first example invites schooler@vlsi.cs.caltech.edu to a multicast
session.
The example below is a request message en route from initiator to 9.1.1 Request
invitee:
C->S: INVITE schooler@vlsi.cs.caltech.edu SIP/2.0 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 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131 Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu (Mark Handley) From: mjh@isi.edu (Mark Handley)
Subject: SIP will be discussed, too Subject: SIP will be discussed, too
To: schooler@cs.caltech.edu (Eve Schooler) To: schooler@cs.caltech.edu (Eve Schooler)
Call-ID: 62729-27@oregon.isi.edu Call-ID: 62729-27@oregon.isi.edu
Content-type: application/sdp Content-type: application/sdp
CSeq: 4711
Content-Length: 187 Content-Length: 187
v=0 v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5 o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio s=Mbone Audio
i=Discussion of Mbone Engineering Issues i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127 c=IN IP4 224.2.0.1/127
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
The first line above states that this is a SIP version 2.0 request.
The Via fields give the hosts along the path from invitation The Via fields list the hosts along the path from invitation
initiator (the first element of the list) towards the invitee. In the initiator (the first element of the list) towards the invitee. In the
example above, the message was last multicast to the administratively example above, the message was last multicast to the administratively
scoped group 239.128.16.254 with a ttl of 16 from the host scoped group 239.128.16.254 with a ttl of 16 from the host
131.215.131.131 131.215.131.131
The request header above states that the request was initiated by 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 mjh@isi.edu the host 128.16.64.19 schooler@cs.caltech.edu is being
invited; the message is currently being routed to invited; the message is currently being routed to
schooler@vlsi.cs.caltech.edu schooler@vlsi.cs.caltech.edu
In this case, the session description is using the Session In this case, the session description is using the Session
Description Protocol (SDP), as stated in the Content-type header. Description Protocol (SDP), as stated in the Content-type header.
The header is terminated by an empty line and is followed by a The header is terminated by an empty line and is followed by a
message body containing the session description. message body containing the session description.
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The called user agent, directly or indirectly through proxy servers, The called user agent, directly or indirectly through proxy servers,
indicates that it is alerting ("ringing") the called party: indicates that it is alerting ("ringing") the called party:
S->C: SIP/2.0 180 Ringing S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 239.128.16.254 16 Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131 Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 1 Via: SIP/2.0/UDP 128.16.64.19 1
From: mjh@isi.edu From: mjh@isi.edu
Call-ID: 62729-27@128.16.64.19 Call-ID: 62729-27@128.16.64.19
Location: sip://es@jove.cs.caltech.edu Location: sip://es@jove.cs.caltech.edu
CSeq: 4711
A sample reply to the invitation is given below. The first line of 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, 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 which means the request was successful. The Via headers are taken
from the request, and entries are removed hop by hop as the reply 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 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 added by the invited user's agent if required. The Call-ID is taken
directly from the original request, along with the remaining fields directly from the original request, along with the remaining fields
of the request message. The original sense of From field is of the request message. The original sense of From field is
preserved (i.e., it is the session initiator). preserved (i.e., it is the session initiator).
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which should be reachable from the caller's host. which should be reachable from the caller's host.
S->C: SIP/2.0 200 OK S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP 239.128.16.254 16 Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131 Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 1 Via: SIP/2.0/UDP 128.16.64.19 1
From: mjh@isi.edu From: mjh@isi.edu
To: schooler@cs.caltech.edu To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19 Call-ID: 62729-27@128.16.64.19
Location: sip://es@jove.cs.caltech.edu Location: sip://es@jove.cs.caltech.edu
CSeq: 4711
For two-party Internet phone calls, the response must contain a The caller confirms the invitation by sending a request to the
description of where to send data to, for example the reply from location named in the Location header:
schooler to mjh :
S->C: SIP/2.0 200 OK C->S: ACK schooler@jove.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 1
From: mjh@isi.edu From: mjh@isi.edu
To: schooler@cs.caltech.edu To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19 Call-ID: 62729-27@128.16.64.19
Location: sip://es@jove.cs.caltech.edu CSeq: 4711
Content-Length: 102
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 v=0
o=schooler 4858949 4858949 IN IP4 192.1.2.3 o=bell 53655765 2353687637 IN IP4 128.3.4.5
t=0 0 c=IN IP4 135.180.144.94
m=audio 5004 RTP/AVP 0 m=audio 3456 RTP/AVP 0 3 4 5
c=IN IP4 131.215.131.147
The caller confirms the invitation by sending a request to the S->C: SIP/2.0 200 OK
location named in the Location header: 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
Content-Length: ...
C->S: CONNECTED schooler@jove.cs.caltech.edu SIP/2.0 v=0
From: mjh@isi.edu o=watson 4858949 4858949 IN IP4 192.1.2.3
To: schooler@cs.caltech.edu c=IN IP4 135.180.161.25
Call-ID: 62729-27@128.16.64.19 m=audio 5004 RTP/AVP 0 3
9.1.3 Aborting a Call 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. If the caller wants to abort a pending call, it sends a BYE request.
C->S: BYE schooler@jove.cs.caltech.edu C->S: BYE schooler@jove.cs.caltech.edu
From: mjh@isi.edu From: mjh@isi.edu
To: schooler@cs.caltech.edu To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19 Call-ID: 62729-27@128.16.64.19
9.1.4 Redirects 9.3.1 Redirects
Replies with response codes "301 Moved Permanently" or "302 Moved Replies with status codes "301 Moved Permanently" or "302 Moved
Temporarily" SHOULD specify another location using the Location Temporarily" SHOULD specify another location using the Location
field. field.
S->C: SIP/2.0 302 Moved temporarily S->C: SIP/2.0 302 Moved temporarily
Via: SIP/2.0/UDP 131.215.131.131 Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu From: mjh@isi.edu
To: schooler@cs.caltech.edu To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19 Call-ID: 62729-27@128.16.64.19
Location: sip://239.128.16.254;ttl=16;transport=udp Location: sip://239.128.16.254;ttl=16;transport=udp
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advised to contact the multicast group 239.128.16.254 with a ttl of 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 16 and UDP transport. In normal situations, a server would not
suggest a redirect to a local multicast group unless, as in the above 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 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 scope of the local group. If the request is redirected to a multicast
group, a proxy server SHOULD query the multicast address itself group, a proxy server SHOULD query the multicast address itself
rather than sending the redirect back towards the client as multicast rather than sending the redirect back towards the client as multicast
may be scoped; this allows a proxy within the appropriate scope may be scoped; this allows a proxy within the appropriate scope
regions to make the query. regions to make the query.
9.1.5 Alternative Services 9.3.2 Alternative Services
An example of a "350 Alternative Service" reply is: An example of a "350 Alternative Service" reply is:
S->C: SIP/2.0 350 Alternative Service S->C: SIP/2.0 350 Alternative Service
Via: SIP/2.0/UDP 131.215.131.131 Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19 Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu From: mjh@isi.edu
To: schooler@cs.caltech.edu To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19 Call-ID: 62729-27@128.16.64.19
Location: recorder@131.215.131.131 Location: recorder@131.215.131.131
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o=mm-server 2523535 0 IN IP4 131.215.131.131 o=mm-server 2523535 0 IN IP4 131.215.131.131
s=Answering Machine s=Answering Machine
i=Leave an audio message i=Leave an audio message
c=IN IP4 128.16.64.19 c=IN IP4 128.16.64.19
t=0 0 t=0 0
m=audio 26472 RTP/AVP 0 m=audio 26472 RTP/AVP 0
Invitation initiators MAY choose to treat a "350 Alternative Service" Invitation initiators MAY choose to treat a "350 Alternative Service"
reply as a failure if they wish to do so. reply as a failure if they wish to do so.
9.1.6 Negotiation 9.3.3 Negotiation
An example of a "606 Not Acceptable" reply is: An example of a "606 Not Acceptable" reply is:
S->C: SIP/2.0 606 Not Acceptable S->C: SIP/2.0 606 Not Acceptable
From: mjh@isi.edu From: mjh@isi.edu
To: schooler@cs.caltech.edu To: schooler@cs.caltech.edu
Call-ID:62729-27@128.16.64.19 Call-ID:62729-27@128.16.64.19
Location: mjh@131.215.131.131 Location: mjh@131.215.131.131
Warning: 606.1 Insufficient bandwidth (only have ISDN), Warning: 606.1 Insufficient bandwidth (only have ISDN),
606.3 Incompatible format, 606.3 Incompatible format,
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In this example, the original request specified 256 kb/s total In this example, the original request specified 256 kb/s total
bandwidth, and the reply states that only 128 kb/s is available. The bandwidth, and the reply states that only 128 kb/s is available. The
original request specified GSM audio, H.261 video, and WB whiteboard. original request specified GSM audio, H.261 video, and WB whiteboard.
The audio coding and whiteboard are not available, but the reply The audio coding and whiteboard are not available, but the reply
states that DVI, PCM or LPC audio could be supported in order of states that DVI, PCM or LPC audio could be supported in order of
preference. The reply also states that multicast is not available. preference. The reply also states that multicast is not available.
In such a case, it might be appropriate to set up a transcoding In such a case, it might be appropriate to set up a transcoding
gateway and re-invite the user. gateway and re-invite the user.
9.2 OPTIONS Request 9.4 OPTIONS Request
A caller Alice can use an OPTIONS request to find out the A caller Alice can use an OPTIONS request to find out the
capabilities of a potential callee Bob, without "ringing" the capabilities of a potential callee Bob, without "ringing" the
designated address. In this case, Bob indicates that he can be designated address. In this case, Bob indicates that he can be
reached at three different addresses, ranging from voice-over-IP to a reached at three different addresses, ranging from voice-over-IP to a
PSTN phone to a pager. PSTN phone to a pager.
C->S: OPTIONS bob@example.com SIP/2.0 C->S: OPTIONS bob@example.com SIP/2.0
From: alice@anywhere.org (Alice) From: alice@anywhere.org (Alice)
To: bob@example.com (Bob) To: bob@example.com (Bob)
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SIP assumes no additional reliability from IP. Requests or replies SIP assumes no additional reliability from IP. Requests or replies
may be lost. A SIP client SHOULD simply retransmit a SIP request may be lost. A SIP client SHOULD simply retransmit a SIP request
periodically with timer T1 (default value of T1: once a second) until 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 it receives a response, or until it has reached a set limit on the
number of retransmissions. The default limit is 20. number of retransmissions. The default limit is 20.
SIP requests and replies are matched up by the client using the SIP requests and replies are matched up by the client using the
Call-ID header field; thus, a server can only have one outstanding Call-ID header field; thus, a server can only have one outstanding
request per call at any given time. request per call at any given time.
HS: A transaction or request ID would remove this
limitation.
If the reply is a provisional response, the initiating client SHOULD If the reply is a provisional response, the initiating client SHOULD
continue retransmitting the request, albeit less frequently, using continue retransmitting the request, albeit less frequently, using
timer T2. The default retransmission interval T2 is 5 seconds. timer T2. The default retransmission interval T2 is 5 seconds.
After the server sends a final response, it cannot be sure the client 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 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 least 30 seconds to avoid having to, for example, contact the user or
user location server again upon receiving a retransmission. user location server again upon receiving a retransmission.
11.1.2 INVITE 11.1.2 INVITE
Special considerations apply for the INVITE method. Special considerations apply for the INVITE method.
1. After receiving an invitation, considerable time may elapse 1. After receiving an invitation, considerable time may elapse
before the server can determine the outcome. For example, before the server can determine the outcome. For example,
the called party may be "rung" or extensive searches may be the called party may be "rung" or extensive searches may be
performed, so delays can reach several tens of seconds. 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 2. It is possible that the invitation request reaches the
callee and the callee is willing to take the call, but that callee and the callee is willing to take the call, but that
the final response (200 OK, in this case) is lost on the the final response (200 OK, in this case) is lost on the
way to the caller. If the session still exists but the way to the caller. If the session still exists but the
initiator gives up on including the user, the contacted initiator gives up on including the user, the contacted
user has sufficient information to be able to join the user has sufficient information to be able to join the
session. However, if the session no longer exists because session. However, if the session no longer exists because
the invitation initiator "hung up" before the reply arrived the invitation initiator "hung up" before the reply arrived
and the session was to be two-way, the conferencing system and the session was to be two-way, the conferencing system
skipping to change at page 50, line 6 skipping to change at page 60, line 33
connection or search to succeed. One cannot rely on the connection or search to succeed. One cannot rely on the
absence of request retransmission, since the server would absence of request retransmission, since the server would
have to continue honoring the request for several request have to continue honoring the request for several request
retransmission periods, that is, possible tens of seconds retransmission periods, that is, possible tens of seconds
if only one or two packets can be lost. if only one or two packets can be lost.
The first problem is solved by indicating progress to the caller: the The first problem is solved by indicating progress to the caller: the
server returns a provisional response indicating it is searching or server returns a provisional response indicating it is searching or
ringing the user. ringing the user.
The server retransmits the final response at intervals of T3 (default The second and third problems are solved by having the server
value of T3 = 2 seconds) until it receives a CONNECTED request for retransmit the final response at intervals of T3 (default value of T3
the same Call-ID or until it has retransmitted the final response 10 = 2 seconds) until it receives an ACK request for the same Call-ID
times. The CONNECTED request is acknowledged only once. If the and CSeq or until it has retransmitted the final response 10 times.
request is syntactically valid and the Request-URI matches that in The ACK request is acknowledged only once. If the request is
the INVITED request with the same Call-ID, the server answers with syntactically valid and the Request-URI matches that in the INVITED
status code 200, otherwise with status code 400. request with the same Call-ID, the server answers with status code
200, otherwise with status code 400.
Fig. 4 and 5 show the client and server state diagram for Fig. 8 and 9 show the client and server state diagram for
invitations. invitations.
11.2 Connection Management for TCP 11.2 Connection Management for TCP
A single TCP connection can serve one or more SIP transactions. A A single TCP connection can serve one or more SIP transactions. A
transaction contains zero or more provisional responses followed by transaction contains zero or more provisional responses followed by
exactly one final response. exactly one final response.
The client MAY close the connection at any time. Closing the
connection before receiving a final response signals that the client
wishes to abort the request.
The server SHOULD NOT close the TCP connection until it has sent its
final response, at which point it MAY close the TCP connection if it
wishes to. However, normally it is the client's responsibility to
close the connection.
If the server leaves the connection open, and if the client so
desires it may re-use the connection for further SIP requests or for
requests from the same family of protocols (such as HTTP or stream
control commands).
12 Behavior of SIP Servers
This section describes behavior of a SIP server in detail. Servers
can operate in proxy or redirect mode. Proxy servers can "fork"
connections, i.e., a single incoming request spawns several outgoing
(client) requests.
A proxy server always inserts a Via header field containing their
own address into requests it issues that are caused by an incoming
request.
We define an "A--B proxy" as a proxy that receives SIP requests over
transport protocol A and issues requests, acting as a SIP client,
+===========+ +===========+
| Initial | | Initial |
+===========+ +===========+
| |
| |
| - | -
| ------ | ------
| INVITE | INVITE
+------v v +------v v
T1 +-----------+ T1 +-----------+
------ | Calling |-------------------+ ------ | Calling |--------+
INVITE +-----------+ | INVITE +-----------+ |
+------| | | | +------| | | |
+----------------+ | | +----------------+ | |
| | | | | 1xx | >= 200
| | | | | --- | ------
| | | | | - | ACK
| | | | | |
| +------v v v-----| | | +------v v v-----| |
| T2 +-----------+ 1xx | | T2 +-----------+ 1xx |
| ------ | Ringing | --- | | ------ | Ringing | --- |
| INVITE +-----------+ - | | INVITE +-----------+ - |
| +------| | | |-----+ | | +------| | |-----+ |
| | +--------------+ | | | |
| 2xx | | >=300 | | 2xx | |
| --------- | 2xx | ----- | | --- | 2xx |
| CONNECTED | --------- | - | | ACK | --- |
| | CONNECTED | | | | ACK |
+----------------+ | | | +----------------+ | |
+------v | v v v +------v | v |
2xx +-----------+ +-----------+ xxx +-----------+ |
--------- | Connected | | Failure | --- | Completed |<-------+
CONNECTED +-----------+ +-----------+ ACK +-----------+
+------| +------|
event event
------- -------
message message
Figure 4: State transition diagram of client for INVITE method Figure 8: State transition diagram of client for INVITE method
using transport protocol B. If not stated explicitly, rules apply to The client MAY close the connection at any time. Closing the
any combination of transport protocols. For conciseness, we only connection before receiving a final response signals that the client
describe behavior with UDP and TCP, but the same rules apply for any wishes to abort the request.
unreliable datagram or reliable protocol, respectively.
+===========+ +===========+
+------------>| Initial |<-------------+ | Initial |<-------------+
| +===========+ | +===========+ |
| |
| |
| INVITE |
| ------ |
| 1xx |
+------v v |
INVITE +-----------+ |
------ | Searching | |
1xx +-----------+ |
+------| | | +---------------->+
| | | | | |
| failure | | failure | | callee picks up |
| ----------- | INVITE | ------- | | --------------- |
| 3xx,4xx,5xx | ------ | >= 300 | | 200 |
| | 1xx | | | | BYE
| +------v v | +------v v v v-----| | ---
| INVITE +-----------+ |
| ------ | Searching | |
| 1xx +-----------+ |
| +------| | | +---------------->+
| | | |
| | | callee picks up |
+----------------+ | --------------- |
| 200 |
| | BYE
+------v v v-----| | ---
INVITE +-----------+ T3 | 200 INVITE +-----------+ T3 | 200
------ | Answered | --- | ------ | Answered | ------ |
1xx +-----------+ 200 | status +-----------+ status |
+------| | | |-----+ | +------| | | |-----+ |
| +---------------->+ | +---------------->+
| | | |
| CONNECTED | | ACK |
| --------- | | --- |
| 200 | | 200 |
| | | |
+------v v | +------v v |
CONNECTED +-----------+ | ACK +-----------+ |
--------- | Connected | | --- | Connected | |
200 +-----------+ | 200 +-----------+ |
+------| | | +------| | |
+-----------------+ +-----------------+
event event
------- -------
message message
Figure 5: State transition diagram of server for INVITE method 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
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 The detailed connection behavior for UDP and TCP is described in
Section 11. Section 11.
12.1 Redirect Server 12.1 Redirect Server
A redirect server does not issue any SIP requests of its own. It can A redirect server does not issue any SIP requests of its own. It can
return a response that accepts, refuses or redirects the request. return a response that refuses or redirects the request. After
After receiving a request, a redirect server proceeds through the receiving an INVITE request, a redirect server proceeds through the
following steps: following steps:
1. If the request cannot be answered immediately (e.g., 1. If the INVITE request cannot be answered immediately
because a location server needs to be contacted), it (e.g., because a location server needs to be contacted), it
returns one or more provisional responses. returns one or more provisional responses.
2. Once the server has gathered the list of alternative 2. Once the server has gathered the list of alternative
locations or has decided to accept or refuse the call, it locations or has decided to refuse the call, it returns the
returns the final response. This ends the SIP transaction. final response. This ends the SIP transaction.
The redirect server maintains transaction state for the whole SIP The redirect server maintains transaction state for the whole SIP
transaction. Servers in user agents are redirect servers. transaction.
12.2 Proxies Issuing Single Unicast Requests 12.2 User Agent Server
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 Proxies in this category issue at most a single unicast request for
each incoming SIP request, that is, they do not "fork" requests. each incoming SIP request, that is, they do not "fork" requests.
Servers may choose to always operate in the mode described in Section Servers may choose to always operate in the mode described in Section
12.3. 12.4.
12.2.1 UDP--UDP Proxy Server
The UDP--UDP server can forward the request and any responses. It
does not have to maintain any state for the SIP transaction. UDP
reliability is assured by the next redirect server in the server
chain.
12.2.2 UDP--TCP Proxy Server
A proxy server issuing a single request over TCP maintains state for
the whole SIP transaction indexed by the Call-ID.
If it receives a UDP retransmission of the same request for an The server can forward the request and any responses. It does not
existing session, it retransmits the last response received from the have to maintain any state for the SIP transaction. Reliability is
TCP side. Any changes in the message body compared to the last assured by the next redirect server in the server chain.
request for the Call-ID are silently ignored. (Otherwise, the proxy
would have to remember and compare the message body; this also
violates the notion of a SIP transaction. TBD) The server SHOULD
cache the final response for a particular Call-ID after the SIP
transaction on the TCP side has completed.
After the cache entry has been expired, the server cannot tell A proxy server SHOULD cache the result of any address translations
whether an incoming request is actually a retransmission of an older and the response to speed forwarding. After the cache entry has been
request, where the TCP side has terminated. It will treat it as a new expired, the server cannot tell whether an incoming request is
request. actually a retransmission of an older request, where the TCP side has
terminated. It will treat it as a new request.
12.3 Proxy Server Issuing Several Requests 12.4 Proxy Server Issuing Several Requests
All requests carry the same Call-ID. For unicast, each of the All requests carry the same Call-ID. For unicast, each of the
requests has a different (host-dependent) Request-URI. For requests has a different (host-dependent) Request-URI. For
multicast, a single request is issued, likely with a host-independent multicast, a single request is issued, likely with a host-independent
Request-URI. A client receiving a multicast query does not have to 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 check whether the host part of the Request-URI matches its own host
or domain name. To avoid response implosion, servers SHOULD NOT or domain name. To avoid response implosion, servers SHOULD NOT
answer multicast requests with a 404 (Not Found) status code. answer multicast requests with a 404 (Not Found) status code.
Servers MAY decide not to answer multicast requests if their response Servers MAY decide not to answer multicast requests if their response
would be 5xx. would be 5xx.
The server MAY respond to the request immediately with a "100 Trying" The server MAY respond to the request immediately with a "100 Trying"
response; otherwise it MAY wait until either the first response to or "180 Ringing" response; otherwise it MAY wait until either the
its requests or the UDP retransmission interval. first response to its requests or the UDP retransmission interval.
The following pseudo-code describes the behavior of a proxy server The following pseudo-code describes the behavior of a proxy server
issuing several requests in response to an incoming request. The issuing several requests in response to an incoming request. The
function request(a) sends a SIP request to address a. function request(r, a) sends a SIP request r to address a.
await_response() waits until a response is received and returns the await_response() waits until a response is received and returns the
response. request_close(a) closes the TCP connection to client with response. close(a) closes the TCP connection to client with address
address a; this is optional. response(s, l, L) sends a response to a. response(s, l, L) sends a response to the client with status s and
the client with status s and list of locations L, with l entries. list of locations L, with l entries. ismulticast() returns 1 if the
ismulticast() returns 1 if the location is a multicast address and location is a multicast address and zero otherwise. The variable
zero otherwise. The variable timeleft indicates the amount of time timeleft indicates the amount of time left until the maximum response
left until the maximum response time has expired. The variable time has expired. The variable recurse indicates whether the server
recurse indicates whether the server will recursively try addresses will recursively try addresses returned through a 3xx response. A
returned through a 3xx response. A server MAY decide to recursively server MAY decide to recursively try only certain addresses, e.g.,
try only certain addresses, e.g., those which are within the same those which are within the same domain as the proxy server. Thus, an
domain as the proxy server. Thus, an initial multicast request may initial multicast request may trigger additional unicast requests.
trigger additional unicast requests.
enum {INVITE, /* request type */
ACK, OPTIONS, BYE, REGISTER, UNREGISTER} R;
int N = 0; /* number of connection attempts */ int N = 0; /* number of connection attempts */
address_t address[]; /* list of addresses */ address_t address[]; /* list of addresses */
int done[]; /* address has responded */
location[]; /* list of locations */ location[]; /* list of locations */
int heard = 0; /* number of sites heard from */ int heard = 0; /* number of sites heard from */
int class; /* class of status code */ int class; /* class of status code */
int best = 1000; /* best response so far */ int best = 1000; /* best response so far */
int timeleft = 120; /* sample timeout value */ int timeleft = 120; /* sample timeout value */
int loc = 0; /* number of locations */ int loc = 0; /* number of locations */
struct { /* response */ struct { /* response */
int status; /* response status */ int status; /* response status */
char *location; /* redirect locations */ char *location; /* redirect locations */
address_t a; /* address of respondent */ address_t a; /* address of respondent */
} r; } r;
int i; int i;
if (multicast) { if (multicast) {
request(address[0]); request(R, address[0]);
} else { } else {
N = /* number of addresses to try */ N = /* number of addresses to try */
for (i = 0; i < N; i++) { for (i = 0; i < N; i++) {
request(address[i]); request(R, address[i]);
done[i] = 0;
} }
} }
while (timeleft > 0 && (heard < N || multicast)) { while (timeleft > 0 && (heard < N || multicast)) {
r = await_response(); r = await_response();
class = r.status / 100; class = r.status / 100;
if (class >= 2) { if (class >= 2) {
heard++; heard++;
if (tcp) request_close(a); for (i = 0; i < N; i++) {
if (address[i] == r.a) {
done[i] = 1;
break;
}
}
} }
if (class == 2) { if (class == 2) {
best = r.status; best = r.status;
break; break;
} }
else if (class == 3) { else if (class == 3) {
/* A server may optionally recurse. The server MUST check whether /* A server may optionally recurse. The server MUST check whether
* it has tried this location before and whether the location is * it has tried this location before and whether the location is
* part of the Via path of the incoming request. This check is * part of the Via path of the incoming request. This check is
* omitted here for brevity. Multicast locations MUST NOT be * omitted here for brevity. Multicast locations MUST NOT be
* returned to the client if the server is not recursing. * returned to the client if the server is not recursing.
*/ */
if (recurse) { if (recurse) {
multicast = 0; multicast = 0;
N++; N++;
request(r.location); request(R, r.location);
} else if (!ismulticast(r.location)) { } else if (!ismulticast(r.location)) {
locations[loc++] = r.location; locations[loc++] = r.location;
best = r.status; best = r.status;
} }
} }
else if (class == 4) { else if (class == 4) {
if (best >= 400) best = r.status; if (best >= 400) best = r.status;
} }
else if (class == 5) { else if (class == 5) {
if (best >= 500) best = r.status; if (best >= 500) best = r.status;
} }
else if (class == 6) { else if (class == 6) {
best = r.status; best = r.status;
break; break;
} }
} }
/* We haven't heard anything useful from anybody. */ /* We haven't heard anything useful from anybody. */
if (best == 1000) { if (best == 1000) {
best = 404; best = 404;
} }
if (best/100 != 3) locs = 0; if (best/100 != 3) loc = 0;
response(best, locs, locations); 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);
}
}
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 When operating in this mode, a proxy server MUST ignore any responses
received for Call-IDs that it does not have a pending transaction received for Call-IDs that it does not have a pending transaction
for. (If server were to forward responses not belonging to a current for. (If server were to forward responses not belonging to a current
transaction using the Via field, the requesting client would get transaction using the Via field, the requesting client would get
confused if it has just issued another request using the same Call- confused if it has just issued another request using the same Call-
ID.) ID.)
13 Security Considerations 13 Third-Party Call Initiation
13.1 Confidentiality 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?
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.
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.
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
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
(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 Unless SIP transactions are protected by lower-layer security
mechanisms such as SSL , an attacker may be able to eavesdrop on call mechanisms such as SSL , an attacker may be able to eavesdrop on call
establishment and invitations and, through the Subject header field establishment and invitations and, through the Subject header field
or the session description, gain insights into the topic of or the session description, gain insights into the topic of
conversation. conversation.
13.2 Integrity 15.2 Integrity
Unless SIP transactions are protected by lower-layer security Unless SIP transactions are protected by lower-layer security
mechanisms such as SSL , an active attacker may be able to modify SIP mechanisms such as SSL , an active attacker may be able to modify SIP
requests. requests.
13.3 Access Control 15.3 Access Control
SIP requests are not authenticated unless the SIP Authorization and SIP requests are not authenticated unless the SIP Authorization and
WWW-Authenticate headers are being used. The strengths and weaknesses WWW-Authenticate headers are being used. The strengths and weaknesses
of these authentication mechanisms are the same as for HTTP. of these authentication mechanisms are the same as for HTTP.
13.4 Privacy 15.4 Privacy
User location and SIP-initiated calls may violate a callee's 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, An implementation SHOULD be able to restrict, on a per-user basis,
what kind of location and availability information is given out to what kind of location and availability information is given out to
certain classes of callers. certain classes of callers.
A Summary of Augmented BNF 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.
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 In this specification we use the Augmented Backus-Naur Form notation
described in [19]. For quick reference, the following is a brief described in [21]. For quick reference, the following is a brief
summary of the main features of this ABNF. summary of the main features of this ABNF.
"abc" "abc"
The case-insensitive string of characters "abc" (or "Abc", The case-insensitive string of characters "abc" (or "Abc",
"aBC", etc.); "aBC", etc.);
%d32 %d32
The character with ASCII code decimal 32 (space); The character with ASCII code decimal 32 (space);
*term *term
skipping to change at page 58, line 13 skipping to change at page 75, line 19
here for completeness. here for completeness.
CR = %d13 ; carriage return character CR = %d13 ; carriage return character
LF = %d10 ; line feed character LF = %d10 ; line feed character
CRLF = CR LF ; typically the end of a line CRLF = CR LF ; typically the end of a line
SP = %d32 ; space character SP = %d32 ; space character
TAB = %d09 ; tab character TAB = %d09 ; tab character
LWS = *( SP | TAB) ; linear whitespace LWS = *( SP | TAB) ; linear whitespace
DIGIT = "0" .. "9" ; a single decimal digit DIGIT = "0" .. "9" ; a single decimal digit
Changes Changes in Version -04
Since version -01, the following things have changed:
o Added personal note to "Searching" section indicating that 6xx
codes may not be necessary. Added figures.
o Initial author's note removed; dated.
o Introduction rewritten to give quick, concise overview as to
what SIP does.
o Conference control (tight vs. loose) seems less and less
appropriate. All share some state such as notions of
membership; some (ITU versions) tend to keep it in a central
server, others distribute it. Some state is synchronized at
larger timescales than other. (After all, even a server won't
know if a participant disconnects from the network until TCP
keep-alive, if any, kicks in.)
o Added list of related protocols to emphasize that this is part
of a whole architecture.
o Terminology: user always reminds me of controlled substances;
thus, this term is avoided where better terminology exists.
Since this protocol sits at the boundary between traditional
Internet and telephony services, some of the terminology
familiar in that realm is introduced.
o Terminology: user location server replaced by redirect server,
since a proxy server may also invoke user location. Also, the
actual user location server (e.g., an LDAP, ULS or similar
directory) may be invoked using protocols other than SIP.
o Rearranged ordering of address resolution to correspond to
host requirements for MX and suggestions in DNS SRV RFC. Adding
note about caching and socket implementation. Added note about
using SMTP EXPN to get an alternate address.
o Defined SIP transaction, provisional and final responses.
o Assigned values to timeout parameters; otherwise, there will
be unnecessary retransmissions between different
implementations.
o Retransmission was greatly simplified; there does not seem to
be a need for all the rules governing transitions between TCP
and UDP domains. A proxy should look just like a server to one
side and like a client to the other. Proxies need to maintain
transaction state in any event since they need to remember
where they forwarded the last SIP request to ( Confirm wouldn't
work otherwise, for example.). Invoking a location service may
yield inconsistent results, introduces additional failure modes
(what if the location server is temporarily unavailable?),
increases delay and processing overhead. UDP--UDP proxies can
still be built without state; they just forward packets and
responses. Proxies with TCP on one and UDP on the other side
will have to act like a normal UDP server and issue 100
responses.
o Removed redundancies and contradictions from request and
response definitions (space vs. SP, duplicate CRLF definition,
recursive request-header, ...).
o Added the experimental methods CONNECTED, REGISTER,
UNREGISTER and BYE.
o Re-engineered the invitation reliability mechanism to use a
separate confirmation message.
o Tentative increase of MTU to 1500 bytes, as per discussion in
Stevens.
o Added Reach, Organization, Subject, Priority,
Authorization, WWW-Authentication headers for improved call
handling. WWW "basic" authentication isn't great, but it is
widely deployed and probably sufficient for giving out
"private" telephone numbers, particularly those where the
callee incurs a charge. (I want to be able to give somebody a
password to call my 800 number via an Internet gateway;
authenticating who that person is requires that I modify a
script on my server to add another distinguished name to the
list of allowable callees.)
o Renamed Reason to Warning (to align with HTTP) header since Since version -03, the following changes have been made.
the response line already offers a failure reason.
Unfortunately, listing several failures is not all that helpful
since the calling party cannot determine which of the media
within the description causes the difficulty or whether it was
the set of media as a whole, but it may give the user agent
some indication as to what's going on.
o SEP and CRLF in headers removed, since this is always implied oThe introduction has been reorganized and large parts
between items. Missing ":" added. CRLF was already in the rewritten.
message definition. Also, unlike RFC 822 and HTTP, the
definition did not allow spaces between the field name and the
colon.
o Added (reluctantly) password to URL. It's no worse than ftp oCONNECTED changed to ACK, as it applies to all responses, not
and needed to easily call from a secure web page, without just 200.
having to type in a password manually.
o Added port to SIP URL to specify non-standard port. oStatus code 181 (Queued) and Call-Disposition: Queue added.
o CAPABILITIES to OPTIONS for closer alignment with HTTP and oStatus code 481 (Invalid Call-ID) added.
RTSP;
o Path to Via for closer alignment with HTTP and RTSP; oStatus code 482 (Loop Detected) added. Via description contains
motivation.
o Content type meta changed to application, since "meta" doesn't oAllow phone numbers in SIP URL for easy connection to Internet
exist as a top-level Internet media type. telephony gateways.
o Formatting closer to HTTP and RTSP. oAdded Also header for third-party connectivity.
o Explain relationship to H.323. 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.
B Open Issues oAdded duration parameter to Retry-After to indicate how long
the callee is likely to be reachable at the address given.
RELIABLE: How to provide reliability? oChanged Sequence to CSeq for consistency with RTSP.
BYE: Use of BYE method? C Open Issues
REGISTER: Use of REGISTER method? Full meshes: How about just transferring an SDP description with
multiple addresses?
H.323: Interaction with H.323 and H.245. H.323: Detailed interaction with H.323 and H.245.
TRANSACTION: Should we have a transaction id in addition to a call 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 ID? Call-IDs are for the end system, but a transaction ID is for
a single SIP exchange. This is useful for Internet telephony, a single SIP exchange. This is useful for Internet telephony,
where a single call may trigger several transactions. 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.
C Acknowledgments D Acknowledgments
We wish to thank the members of the IETF MMUSIC WG for their comments We wish to thank the members of the IETF MMUSIC WG for their comments
and suggestions. This work is based, inter alia, on [23,24]. and suggestions. Detailed comments were provided by Jonathan
Rosenberg. This work is based, inter alia, on [29,30]. Parameters of
Parameters of the terminal negotiation mechanism were influenced by the terminal negotiation mechanism were influenced by Scott Petrack's
Scott Petrack's CMA design. CMA design.
D Authors' Addresses E Authors' Addresses
Mark Handley Mark Handley
USC Information Sciences Institute USC Information Sciences Institute
c/o MIT Laboratory for Computer Science c/o MIT Laboratory for Computer Science
545 Technology Square 545 Technology Square
Cambridge, MA 02139 Cambridge, MA 02139
USA USA
electronic mail: mjh@isi.edu electronic mail: mjh@isi.edu
Henning Schulzrinne Henning Schulzrinne
skipping to change at page 61, line 33 skipping to change at page 77, line 7
USA USA
electronic mail: schulzrinne@cs.columbia.edu electronic mail: schulzrinne@cs.columbia.edu
Eve Schooler Eve Schooler
Computer Science Department 256-80 Computer Science Department 256-80
California Institute of Technology California Institute of Technology
Pasadena, CA 91125 Pasadena, CA 91125
USA USA
electronic mail: schooler@cs.caltech.edu electronic mail: schooler@cs.caltech.edu
E Bibliography F Bibliography
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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. IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.
[2] R. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin, [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 "Resource reservation protocol (RSVP) -- version 1 functional
specification," Internet Draft, Internet Engineering Task Force, June specification," Internet Draft, Internet Engineering Task Force, June
1997. Work in progress. 1997. Work in progress.
[3] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a [7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
transport protocol for real-time applications," RFC 1889, Internet transport protocol for real-time applications," Tech. Rep. RFC 1889,
Engineering Task Force, Jan. 1996. Internet Engineering Task Force, Jan. 1996.
[4] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming [8] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Internet Draft, Internet Engineering Task Force, protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
Mar. 1997. Work in progress. July 1997. Work in progress.
[5] M. Handley, "SAP: Session announcement protocol," Internet Draft,
Internet Engineering Task Force, Nov. 1996. Work in progress.
[6] M. Handley and V. Jacobson, "SDP: Session description protocol," [9] M. Handley and V. Jacobson, "SDP: Session description protocol,"
Internet Draft, Internet Engineering Task Force, Mar. 1997. Work in Internet Draft, Internet Engineering Task Force, Mar. 1997. Work in
progress. progress.
[7] P. Lantz, "Usage of H.323 on the Internet," Internet Draft, [10] S. Bradner, "Key words for use in RFCs to indicate requirement
Internet Engineering Task Force, Feb. 1997. Work in progress. level," Tech. Rep. RFC 2119, Internet Engineering Task Force, Mar.
1997.
[8] S. Bradner, "Key words for use in RFCs to indicate requirement [11] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997. Lee, "Hypertext transfer protocol -- HTTP/1.1," Tech. Rep. RFC 2068,
Internet Engineering Task Force, Jan. 1997.
[9] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-Lee, [12] C. Partridge, "Mail routing and the domain system," Tech. Rep.
"Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
Engineering Task Force, Jan. 1997.
[10] C. Partridge, "Mail routing and the domain system," STD 14, RFC RFC 974, Internet Engineering Task Force, Jan. 1986.
974, Internet Engineering Task Force, Jan. 1986.
[11] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the [13] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the
location of services (DNS SRV)," RFC 2052, Internet Engineering Task location of services (DNS SRV)," Tech. Rep. RFC 2052, Internet
Force, Oct. 1996. Engineering Task Force, Oct. 1996.
[12] P. Mockapetris, "Domain names - implementation and [14] P. V. Mockapetris, "Domain names - implementation and
specification," STD 13, RFC 1035, Internet Engineering Task Force, specification," Tech. Rep. RFC 1035, Internet Engineering Task
Nov. 1987. Force, Nov. 1987.
[13] R. Braden, "Requirements for internet hosts - application and [15] R. T. Braden, "Requirements for internet hosts - application and
support," STD 3, RFC 1123, Internet Engineering Task Force, Oct. support," Tech. Rep. RFC 1123, Internet Engineering Task Force, Oct.
1989. 1989.
[14] D. Zimmerman, "The finger user information protocol," RFC 1288, [16] D. Zimmerman, "The finger user information protocol," Tech. Rep.
Internet Engineering Task Force, Dec. 1991. RFC 1288, Internet Engineering Task Force, Dec. 1991.
[15] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access [17] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
protocol," RFC 1777, Internet Engineering Task Force, Mar. 1995. protocol," Tech. Rep. RFC 1777, Internet Engineering Task Force, Mar.
1995.
[16] T. Berners-Lee, "Universal resource identifiers in WWW: a [18] T. Berners-Lee, "Universal resource identifiers in WWW: a
unifying syntax for the expression of names and addresses of objects unifying syntax for the expression of names and addresses of objects
on the network as used in the world-wide web," RFC 1630, Internet on the network as used in the world-wide web," Tech. Rep. RFC 1630,
Engineering Task Force, June 1994. Internet Engineering Task Force, June 1994.
[17] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource [19] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
locators (URL): Generic syntax and semantics," Internet Draft, locators (URL): Generic syntax and semantics," Internet Draft,
Internet Engineering Task Force, May 1997. Work in progress. Internet Engineering Task Force, May 1997. Work in progress.
[18] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource [20] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
locators (URL)," RFC 1738, Internet Engineering Task Force, Dec. locators (URL)," Tech. Rep. RFC 1738, Internet Engineering Task
1994. Force, Dec. 1994.
[19] D. Crocker, "Augmented BNF for syntax specifications: ABNF," [21] D. Crocker, "Augmented BNF for syntax specifications: ABNF,"
Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in
progress. progress.
[20] J. Mogul and S. Deering, "Path MTU discovery," RFC 1191, [22] J. C. Mogul and S. E. Deering, "Path MTU discovery," Tech. Rep.
Internet Engineering Task Force, Nov. 1990. RFC 1191, Internet Engineering Task Force, Nov. 1990.
[21] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1. [23] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1.
Reading, Massachusetts: Addison-Wesley, 1994. Reading, Massachusetts: Addison-Wesley, 1994.
[22] D. Crocker, "Standard for the format of ARPA internet text [24] D. Crocker, "Standard for the format of ARPA internet text
messages," STD 11, RFC 822, Internet Engineering Task Force, Aug. messages," Tech. Rep. Also STD0011, RFC 822, Internet Engineering
1982. Task Force, Aug. 1982.
[23] E. M. Schooler, "Case study: multimedia conference control in a [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: packet-switched teleconferencing system," Journal of Internetworking:
Research and Experience , vol. 4, pp. 99--120, June 1993. ISI Research and Experience , vol. 4, pp. 99--120, June 1993. ISI
reprint series ISI/RS-93-359. reprint series ISI/RS-93-359.
[24] H. Schulzrinne, "Personal mobility for multimedia services in [30] H. Schulzrinne, "Personal mobility for multimedia services in
the Internet," in European Workshop on Interactive Distributed the Internet," in European Workshop on Interactive Distributed
Multimedia Systems and Services , (Berlin, Germany), Mar. 1996. Multimedia Systems and Services , (Berlin, Germany), Mar. 1996.
Full Copyright Statement
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Table of Contents Table of Contents
1 Introduction ........................................ 2 1 Introduction ........................................ 2
1.1 Overview of SIP Functionality ....................... 2 1.1 Overview of SIP Functionality ....................... 2
1.2 Finding Multimedia Sessions ......................... 3 1.2 Terminology ......................................... 3
1.3 Terminology ......................................... 4 1.3 Definitions ......................................... 4
1.4 Definitions ......................................... 4 1.4 Summary of SIP Operation ............................ 6
1.5 Protocol Properties ................................. 6 1.4.1 SIP Addressing ...................................... 6
1.5.1 Minimal State ....................................... 6 1.4.2 Locating a SIP Server ............................... 7
1.5.2 Transport-Protocol Neutral .......................... 6 1.4.3 SIP Transaction ..................................... 9
1.5.3 Text-Based .......................................... 6 1.4.4 SIP Invitation ...................................... 9
1.6 SIP Addressing ...................................... 6 1.4.5 Locating a User ..................................... 10
1.7 Locating a SIP Server ............................... 8 1.4.6 Changing an Existing Session ........................ 13
1.8 SIP Transactions .................................... 9 1.4.7 Registration Services ............................... 13
1.9 Locating a User ..................................... 9 1.5 Protocol Properties ................................. 13
2 SIP Uniform Resource Locators ....................... 12 1.5.1 Minimal State ....................................... 13
3 SIP Message Overview ................................ 14 1.5.2 Transport-Protocol Neutral .......................... 14
4 Request ............................................. 15 1.5.3 Text-Based .......................................... 14
4.1 Request-Line ........................................ 16 2 SIP Uniform Resource Locators ....................... 14
4.1.1 Methods ............................................. 17 3 SIP Message Overview ................................ 17
4.1.2 Request-URI ......................................... 18 4 Request ............................................. 18
4.1.3 SIP Version ......................................... 18 4.1 Request-Line ........................................ 18
5 Response ............................................ 18 4.2 Methods ............................................. 19
5.1 Status-Line ......................................... 19 4.2.1 INVITE ............................................. 20
5.1.1 Status Codes and Reason Phrases ..................... 19 4.2.2 ACK ................................................ 20
6 Header Field Definitions ............................ 20 4.2.3 OPTIONS ............................................ 20
6.1 General Header Fields ............................... 22 4.2.4 BYE ................................................ 20
6.2 Entity Header Fields ................................ 22 4.2.5 REGISTER ........................................... 21
6.3 Request Header Fields ............................... 22 4.2.6 UNREGISTER ......................................... 21
6.4 Response Header Fields .............................. 22 4.3 Request-URI ......................................... 21
6.5 Header Field Format ................................. 23 4.3.1 SIP Version ......................................... 22
6.6 Accept .............................................. 23 4.4 Option Tags ......................................... 22
6.7 Accept-Language ..................................... 24 4.4.1 Registering New Option Tags with IANA ............... 22
6.8 Allow ............................................... 24 5 Response ............................................ 23
6.9 Authorization ....................................... 24 5.1 Status-Line ......................................... 23
6.10 Authentication ...................................... 24 5.1.1 Status Codes and Reason Phrases ..................... 23
6.11 Call-ID ............................................. 24 6 Header Field Definitions ............................ 25
6.12 Content-Length ...................................... 25 6.1 General Header Fields ............................... 27
6.13 Content-Type ........................................ 25 6.2 Entity Header Fields ................................ 27
6.14 Date ................................................ 26 6.3 Request Header Fields ............................... 27
6.15 Expires ............................................. 26 6.4 Response Header Fields .............................. 29
6.16 From ................................................ 27 6.5 Header Field Format ................................. 29
6.17 Location ............................................ 27 6.6 Accept .............................................. 30
6.18 Organization ........................................ 29 6.7 Accept-Language ..................................... 30
6.19 PEP ................................................. 29 6.8 Allow ............................................... 30
6.20 Priority ............................................ 29 6.9 Also ................................................ 30
6.21 Proxy-Authenticate .................................. 29 6.10 Authorization ....................................... 31
6.22 Proxy-Authorization ................................. 29 6.11 Call-Disposition .................................... 31
6.23 Public .............................................. 30 6.12 Call-ID ............................................. 32
6.24 Reach ............................................... 30 6.13 Content-Length ...................................... 32
6.25 Retry-After ......................................... 30 6.14 Content-Type ........................................ 33
6.26 Sequence ............................................ 31 6.15 Date ................................................ 33
6.27 Server .............................................. 31 6.16 Expires ............................................. 33
6.28 Subject ............................................. 31 6.17 From ................................................ 34
6.29 To .................................................. 32 6.18 Location ............................................ 35
6.30 User-Agent .......................................... 32 6.19 Organization ........................................ 37
6.31 Via ................................................. 32 6.20 Priority ............................................ 37
6.32 Warning ............................................. 33 6.21 Proxy-Authenticate .................................. 38
6.33 WWW-Authenticate .................................... 34 6.22 Proxy-Authorization ................................. 38
7 Status Code Definitions ............................. 34 6.23 Public .............................................. 38
7.1 Informational 1xx ................................... 35 6.24 Require ............................................. 38
7.1.1 100 Trying .......................................... 35 6.25 Retry-After ......................................... 39
7.1.2 180 Ringing ......................................... 35 6.26 CSeq ................................................ 39
7.2 Successful 2xx ...................................... 35 6.27 Server .............................................. 40
7.2.1 200 OK .............................................. 35 6.28 Subject ............................................. 40
7.3 Redirection 3xx ..................................... 35 6.29 Unsupported ......................................... 40
7.3.1 300 Multiple Choices ................................ 35 6.30 Timestamp ........................................... 41
7.3.2 301 Moved Permanently ............................... 36 6.31 To .................................................. 41
7.3.3 302 Moved Temporarily ............................... 36 6.32 User-Agent .......................................... 41
7.3.4 380 Alternative Service ............................. 36 6.33 Via ................................................. 41
7.4 Request Failure 4xx ................................. 36 6.34 Warning ............................................. 43
7.4.1 400 Bad Request ..................................... 36 6.35 WWW-Authenticate .................................... 44
7.4.2 401 Unauthorized .................................... 37 7 Status Code Definitions ............................. 44
7.4.3 402 Payment Required ................................ 37 7.1 Informational 1xx ................................... 44
7.4.4 403 Forbidden ....................................... 37 7.1.1 100 Trying .......................................... 44
7.4.5 404 Not Found ....................................... 37 7.1.2 180 Ringing ......................................... 44
7.4.6 405 Method Not Allowed .............................. 37 7.1.3 181 Queued .......................................... 45
7.4.7 407 Proxy Authentication Required ................... 37 7.2 Successful 2xx ...................................... 45
7.4.8 408 Request Timeout ................................. 37 7.2.1 200 OK .............................................. 45
7.4.9 420 Bad Extension ................................... 37 7.3 Redirection 3xx ..................................... 45
7.4.10 480 Temporarily Unavailable ......................... 38 7.3.1 300 Multiple Choices ................................ 45
7.5 Server Failure 5xx .................................. 38 7.3.2 301 Moved Permanently ............................... 46
7.5.1 500 Server Internal Error ........................... 38 7.3.3 302 Moved Temporarily ............................... 46
7.5.2 501 Not implemented ................................. 38 7.3.4 380 Alternative Service ............................. 46
7.5.3 502 Bad Gateway ..................................... 38 7.4 Request Failure 4xx ................................. 46
7.5.4 503 Service Unavailable ............................. 38 7.4.1 400 Bad Request ..................................... 46
7.5.5 504 Gateway Timeout ................................. 39 7.4.2 401 Unauthorized .................................... 46
7.6 Global Failures ..................................... 39 7.4.3 402 Payment Required ................................ 46
7.6.1 600 Busy ............................................ 39 7.4.4 403 Forbidden ....................................... 46
7.6.2 603 Decline ......................................... 39 7.4.5 404 Not Found ....................................... 46
7.6.3 604 Does not exist anywhere ......................... 39 7.4.6 405 Method Not Allowed .............................. 47
7.6.4 606 Not Acceptable .................................. 39 7.4.7 407 Proxy Authentication Required ................... 47
8 SIP Message Body .................................... 40 7.4.8 408 Request Timeout ................................. 47
8.1 Body Inclusion ...................................... 40 7.4.9 420 Bad Extension ................................... 47
8.2 Message Body Length ................................. 40 7.4.10 480 Temporarily Unavailable ......................... 47
9 Examples ............................................ 41 7.4.11 481 Invalid Call-ID ................................. 47
9.1 Invitation .......................................... 41 7.4.12 482 Loop Detected ................................... 48
9.1.1 Request ............................................. 41 7.5 Server Failure 5xx .................................. 48
9.1.2 Reply ............................................... 42 7.5.1 500 Server Internal Error ........................... 48
9.1.3 Aborting a Call ..................................... 43 7.5.2 501 Not implemented ................................. 48
9.1.4 Redirects ........................................... 44 7.5.3 502 Bad Gateway ..................................... 48
9.1.5 Alternative Services ................................ 44 7.5.4 503 Service Unavailable ............................. 48
9.1.6 Negotiation ......................................... 45 7.5.5 504 Gateway Timeout ................................. 48
9.2 OPTIONS Request ..................................... 46 7.6 Global Failures 6xx ................................. 49
10 Compact Form ........................................ 47 7.6.1 600 Busy ............................................ 49
11 SIP Transport ....................................... 48 7.6.2 603 Decline ......................................... 49
11.1 Achieving Reliability For UDP Transport ............. 48 7.6.3 604 Does not exist anywhere ......................... 49
11.1.1 General Operation ................................... 48 7.6.4 606 Not Acceptable .................................. 49
11.1.2 INVITE .............................................. 49 8 SIP Message Body .................................... 50
11.2 Connection Management for TCP ....................... 50 8.1 Body Inclusion ...................................... 50
12 Behavior of SIP Servers ............................. 50 8.2 Message Body Length ................................. 50
12.1 Redirect Server ..................................... 53 9 Examples ............................................ 51
12.2 Proxies Issuing Single Unicast Requests ............. 53 9.1 Invitation to Multimedia Conference ................. 51
12.2.1 UDP--UDP Proxy Server ............................... 53 9.1.1 Request ............................................. 51
12.2.2 UDP--TCP Proxy Server ............................... 53 9.1.2 Reply ............................................... 52
12.3 Proxy Server Issuing Several Requests ............... 54 9.2 Two-party Call ...................................... 53
13 Security Considerations ............................. 56 9.3 Aborting a Call ..................................... 54
13.1 Confidentiality ..................................... 56 9.3.1 Redirects ........................................... 54
13.2 Integrity ........................................... 56 9.3.2 Alternative Services ................................ 55
13.3 Access Control ...................................... 56 9.3.3 Negotiation ......................................... 56
13.4 Privacy ............................................. 56 9.4 OPTIONS Request ..................................... 57
A Summary of Augmented BNF ............................ 57 10 Compact Form ........................................ 57
B Open Issues ......................................... 60 11 SIP Transport ....................................... 58
C Acknowledgments ..................................... 60 11.1 Achieving Reliability For UDP Transport ............. 59
D Authors' Addresses .................................. 61 11.1.1 General Operation ................................... 59
E Bibliography ........................................ 61 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
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
 End of changes. 

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