draft-ietf-mmusic-sip-02.txt   draft-ietf-mmusic-sip-03.txt 
Internet Engineering Task Force MMUSIC WG Internet Engineering Task Force MMUSIC WG
Internet Draft M. Handley, H. Schulzrinne, E. Schooler Internet Draft Handley/Schulzrinne/Schooler
ietf-mmusic-sip-02.txt ISI/Columbia U./Caltech draft-ietf-mmusic-sip-03.txt ISI/Columbia U./Caltech
March 27, 1997 July 31, 1997
Expires: September 25, 1997 Expires: January 20, 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 1, line 39 skipping to change at page 1, line 39
Distribution of this document is unlimited. Distribution of this document is unlimited.
ABSTRACT ABSTRACT
Many styles of multimedia conferencing are likely to co- Many styles of multimedia conferencing are likely to co-
exist on the Internet, and many of them share the need to exist on the Internet, and many of them share the need to
invite users to participate. The Session Initiation invite users to participate. The Session Initiation
Protocol (SIP) is a simple protocol designed to enable Protocol (SIP) is a simple protocol designed to enable
the invitation of users to participate in such multimedia the invitation of users to participate in such multimedia
sessions. It is not tied to any specific conference sessions. It is not tied to any specific conference
control scheme, providing support for either loosely or control scheme. In particular, it aims to enable user
tightly controlled sessions. In particular, it aims to mobility by relaying and redirecting invitations to a
enable user mobility by relaying and redirecting user's current location.
invitations to a user's current location.
This document is a product of the Multiparty Multimedia This document is a product of the Multi-party Multimedia
Session Control (MMUSIC) working group of the Internet Session Control (MMUSIC) working group of the Internet
Engineering Task Force. Comments are solicited and Engineering Task Force. Comments are solicited and
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.
Authors' Note 1 Introduction
This document is the result of a merger of the Session Invitation 1.1 Overview of SIP Functionality
Protocol (draft-ietf-mmusic-sip-00.txt) and the Simple Conference
Invitation Protocol (draft-ietf-mmusic-scip-00.txt), and of an
attempt to make SIP more generic and to fit into a more flexible
infrastructure that includes companion protocols including SDP, HTTP
and RTSP.
Changes The Session Initiation Protocol (SIP) is an application-layer
protocol that can establish and control multimedia sessions or calls.
These multimedia sessions include multimedia conferences, distance
learning, Internet telephony and similar applications. SIP can
initiate both unicast and multicast sessions; the initiator does not
necessarily have to be a member of the session. Media and
participants can be added to an existing session. SIP can be used to
"call" both persons and "robots", for example, to invite a media
storage device to record an ongoing conference or to invite a video-
on-demand server to play a video into a conference. (SIP does not
directly control these services, however; see RTSP [4].)
Since version -01, the following things have changed: SIP transparently supports name mapping and redirection services,
allowing the implementation of telephony services such as selective
call forwarding, selective call rejection, conditional and
unconditional call forwarding, call forwarding busy, call forwarding
no response. SIP may use multicast to try several possible callee
locations at the same time.
o CAPABILITIES to OPTIONS for closer alignment with HTTP and SIP supports personal mobility telecommunications intelligent network
RTSP; services, this is defined as: "Personal mobility is the ability of
end users to originate and receive calls and access subscribed
telecommunication services on any terminal in any location, and the
ability of the network to identify end users as they move. Personal
mobility is based on the use of a unique personal identity (i.e.,
'personal number')." [1]. Personal mobility complements terminal
mobility, i.e., the ability to maintain communications when moving a
single end system from one network to another.
o Path to Via for closer alignment with HTTP and RTSP; SIP supports some or all of four facets of establishing multimedia
communications:
o Content type meta changed to application, since "meta" doesn't 1. user location: determination of the end system to be used
exist as a top-level Internet media type. for communication;
o Formatting closer to HTTP and RTSP. 2. user capabilities: determination of the media and media
parameters to be used;
o Explain relationship to H.323. 3. user availability: determination of the willingness of the
called party to engage in communications;
1 Introduction 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
introduce conference control protocols to a session.
SIP is designed as part of the overall IETF multimedia data and
control architecture currently incorporating protocols such as RSVP
[2] for reserving network resources, RTP [3] for transporting real-
time data and providing QOS feedback, RTSP [4] for controlling
delivery of streaming media, SAP [5] for advertising multimedia
sessions via multicast and SDP [6] for describing multimedia
sessions, but SIP does not depend for its operation on any of these
protocols.
1.2 Finding Multimedia Sessions
There are two basic ways to locate and to participate in a multimedia There are two basic ways to locate and to participate in a multimedia
session: session:
o The session is advertised, users see the advertisement, then Advertisement: The session is advertised, potential participants see
join the session address to participate. the advertisement, then join the session address to participate.
o Users are invited to participate in a session, which may or
may not already be advertised.
The Session Description Protocol (SDP) [1] together with the Session Invitation: Users are invited by others to participate in a session,
Announcement Protocol (SAP) [2], provide a mechanism for the former. which may or may not be advertised.
This document presents the Session Initiation Protocol (SIP) to
perform the latter. SIP MAY also use SDP to describe a session.
Figure omitted in ASCII version Sessions may be advertised using multicast protocols such as SAP [5],
electronic mail, news groups, web pages or directories (LDAP), among
others. SIP serves the role of the invitation protocol.
Figure 1: Session Lifecycle SIP does not prescribe how a conference is to be managed, e.g.,
We make the design decision that how a user discovers that a session whether it uses a central server to manage conference and participant
exists is orthogonal to a session's conference control model. Figure state or distributes state via multicast.
1 shows a potential place for SIP in the lifecycle of both
lightweight sessions and in more tightly-coupled conferencing. Note
that the Session Initiation Protocol and the Session Announcement
Protocol may be invoked or re-invoked at later stages in a session's
lifecycle.
The Session Initiation Protocol is also intended to be used to invite SIP does not allocate multicast addresses, leaving this functionality
servers into sessions. Examples might be where a recording server can to protocols such as SAP [5].
be invited to participate in a live multimedia session to record that
session, or a video-on-demand server can be invited to play a video
stream into a live multimedia conference. In such cases we would like
SIP to lead the server gracefully into the control protocol that
controls the actual recording and playback.
We also make the design decision that inviting a user to participate SIP can invite users to conferences with and without resource
in a session is independent of quality of service (QoS) guarantees reservation. SIP does not reserve resources, but may convey to the
for that session. Such QoS guarantees (if they are required) may be invited system the information necessary to do this. Quality-of-
dependent on the full membership of the session, and this may or may service guarantees, if required, may depend on knowing the full
not be known to the agent performing session invitation. membership of the session; this information may or may not be known
to the agent performing session invitation.
SIP offers some of the same functionality as H.323, but can also be SIP offers some of the same functionality as H.323, but may also be
used in conjunction with it. In this mode, SIP is used to locate the used in conjunction with it. In this mode, H.323 is used to locate
appropriate terminal, where the terminal is identified by its H.245 the appropriate terminal identified by a H.245 address [TBD: what
address [TBD: what does this look like?]. An H.323-capable terminal does this look like?]. An H.323-capable terminal then proceeds with a
then proceeds with a normal H.323/H.245 invitation [3]. normal H.323/H.245 invitation [7].
1.1 Requirements 1.3 Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
document are to be interpreted as described in RFC xxxx [4]. and "OPTIONAL" are to be interpreted as described in RFC 2119 [8] and
indicate requirement levels for compliant SIP implementations.
1.2 Terminology 1.4 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 HTTP. client, server and proxy are similar to those used by the Hypertext
Transport Protocol (HTTP) [9].
Client: An application program that establishes connections for the 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.
Initiator: The party initiating a conference invitation. Note that Final response: A response that terminates a -> SIP transaction, as
the calling party does not have to be the same as the one opposed to a -> provisional response 3xx, 4xx, and 5xx
creating a conference. responses are final.
Invitation: A request sent to attempt to contact a user (or service) Initiator, calling party: The party initiating a conference
to request that they participate in a session. invitation. Note that the calling party does not have to be the
same as the one creating a conference.
Invitee, Invited User: The person or service that the calling party Invitation: A request sent to a user (or service) requesting
is trying to invite to a conference. participation in a session.
Location server: A program that is contacted by a client and that Invitee, invited user, called party: The person or service that the
returns one or more possible locations for the user or service calling party is trying to invite to a conference.
without contacting that user or service directly.
Location service: A service used by a location server to obtain Location server: A program that is contacted by a -> client and
information about a user's possible location. 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.
Proxy, Proxy server: An intermediary program that acts as both a Location service: A service used by a -> redirect or -> proxy
server to obtain information about a callee's possible location.
Provisional response: A response used by the server to indicate
progress, but that does not terminate a -> SIP transaction.
All 1xx and 6xx responses are provisional. Other responses are
considered -> final.
Proxy, proxy server: An intermediary program that acts as both a
server and a client for the purpose of making requests on behalf 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.
Server: An application program that accepts connections in order to Redirect server: A server that accepts a SIP request, maps the
service requests by sending back responses. A server may be the address into zero or more new addresses and returns these
called user agent, a proxy server, or a location server. addresses to the client. Unlike a -> proxy server, it does not
initiate its own SIP request.
User Agent, Called User Agent: The server application which contacts Server: An application program that accepts requests in order to
the invitee to inform them of the invitation, and to return a service requests and sends back responses to those requests.
reply. Servers are either proxy, redirect or user agent servers. An
application program may act as both server and client.
Any given program may be capable of acting both as a client and a Session: "A multimedia session is a set of multimedia senders and
server. A typical multimedia conference controller would act as a receivers and the data streams flowing from senders to
client to initiate calls or to invite others to conferences and as a receivers. A multimedia conference is an example of a multimedia
server to accept invitations. session." [6] For SIP, a session is equivalent to a "call".
(Note: a session as defined here may comprise one or more RTP
sessions.)
1.3 General Requirements (SIP) transaction: A SIP transaction occurs between a -> client and
a -> server and comprises all messages from the first request
sent from the client to the server up to a -> final (non-1xx)
response sent from the server to the client. A transaction is
for a single call (identified by a Call-ID, Section 6.11).
There can only be one pending transaction between a server and
client for each call id.
SIP is a Session Initiation Protocol. It is not a conference control User agent server, called user agent: The server application that
protocol. SIP can be used to perform a search for a user or service contacts the user when a session request is received and that
and to request that that user or service participate in a session. returns a reply on behalf of the user. The reply may accept,
reject or redirect the call. (Note: in SIP, user agents can be
both clients and servers.)
Once SIP has been used to initiate a multimedia session SIP's task is An application program may be capable of acting both as a client and
finished. There is no concept of a SIP session (as opposed to a SIP a server. For example, a typical multimedia conference control
search for a user or service). If whatever conference control application would act as a client to initiate calls or to invite
mechanism is used in the session needs to add or remove a media others to conferences and as a user agent server to accept
stream, SIP may be used to perform this task, but again, once the invitations.
information has been successfully conveyed to the participants, SIP
is then no longer involved.
SIP must be able to utilize both UDP and TCP as transport protocols. 1.5 Protocol Properties
From a performance point of view, UDP is preferable as it allows the 1.5.1 Minimal State
application to more carefully control the timing of messages, it
allows parallel searches without requiring connection state for each
outstanding request, and allows the use of multicast.
From a pragmatic point of view, TCP allows easier passage through There is no concept of an ongoing SIP session that lasts for the
existing firewalls, and with appropriate protocol design, allows duration of the conference or call. Rather, a single conference
common SIP, HTTP and RTSP servers. 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.
When TCP is used, SIP can use either one or more than one connection At most, a server has to maintain state for a single SIP transaction.
to attempt to contact a user or to modify parameters of an existing In some cases, it can process each message without regard to previous
session. The concept of a session is not implicitly bound to a TCP messages ( stateless server ), as described in Section 12.
connection, so the initial SIP request and a subsequent SIP request
may use different TCP connections or a single persistent connection 1.5.2 Transport-Protocol Neutral
as appropriate.
SIP is able to utilize both UDP and TCP as transport protocols. UDP
allows the application to more carefully control the timing of
messages and their retransmission, to perform parallel searches
without requiring connection state for each outstanding request, and
to use multicast. TCP allows easier passage through existing
firewalls, and given the similar protocol design, allows common
servers for SIP, HTTP and the Real Time Streaming Protocol (RTSP)
[4].
When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing session. The
concept of a session is not implicitly bound to a TCP connection, so
the initial SIP request and a subsequent SIP request may use
different TCP connections or a single persistent connection as
appropriate.
Clients SHOULD implement both UDP and TCP transport, servers MUST.
1.5.3 Text-Based
SIP is text based. This allows easy implementation in languages such SIP is text based. This allows easy implementation in languages such
as TCL and Perl, allows easy debugging, and most importantly, makes as Tcl and Perl, allows easy debugging, and most importantly, makes
SIP flexible and extensible. As SIP is only used for session SIP flexible and extensible. As SIP is primarily used for session
initiation, it is believed that the additional overhead of using a initiation, it is believed that the additional overhead of using a
text-based protocol is not significant. text-based protocol is not significant.
Unlike control protocols, there is minimal shared-state in SIP -- in 1.6 SIP Addressing
a minimal implementation the initiator maintains all the state about
the current attempt to locate and contact a user or service - servers
or proxies can be stateless (although they don't have to be). All the
state needed to get a response back from a server to the initiator is
carried in the SIP request itself - this is also necessary for loop
prevention.
Whilst redesigning SIP, we have attempted to ensure that it SIP uses two kinds of address identifiers, host-specific addresses
has a clear interaction with the currently evolving Real- and host-independent addresses form user@host , where user is any
Time Stream Control Protocol. 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.
1.4 Addressing For a host-specific address, the user part is an operating-system
user name. The host part is either a domain name having a DNS A
(address) record, or a numeric network address. Examples include:
SIP is a protocol that exchanges messages between peer user agents or mjh@metro.isi.edu
proxies for user agents. We assume the user agent is an application hgs@erlang.cs.columbia.edu
that acts on behalf of the user it represents (thus it is sometimes root@193.175.132.42
described as a client of the user) and that is co-resident with that
user. A proxy for a user agent serves as a forwarding mechanism or
bridge to the actual location of the user agent. We also refer to
such proxies as location server
In the computer realm, the equivalent of a personal telephone number A user's host-specific address can be obtained out-of-band, can be
combines the user's login id ( mjh ) with a machine host name ( learned via existing media agents, can be included in some mailers'
metro.isi.edu ) or numeric network address ( 128.16.64.78 ). A user's message headers, or can be recorded during previous invitation
location-specific address can be obtained out-of-band, can be learned interactions.
via existing media agents, can be included in some mailers' message
headers, or can be recorded during previous invitation interactions.
However, users also publish several well-known addresses that are Host-independent addresses contain a moniker (such as a civil name)
relatively location-independent, such as email or web home-page or user name and domain name that may not map into a single host.
addresses. Rather than require that users provide their specific [1]
network locales, we can take advantage of email and web addresses as
being (relatively) memorable, and also leverage off the Domain Name
Service (DNS) to provide a first stage location mechanism. Note that
an email address ( M.Handley@cs.ucl.ac.uk ) is usually different from
the combination of a specific machine name and login name (
mjh@mercury.lcs.mit.edu ). SIP should allow both forms of addressing
to be used, with the former requiring a location server to locate the
user.
One perceived problem of email addressing is that it is possible to Host-independent addresses may use any unambiguous user name,
guess peoples' addresses and thus the system of unlisted (in the including aliases, identifying the called party as the user part of
telephone directory) numbers is more of a problem. However, this the address. They may use a domain name having an MX [10], SRV [11]
really only provides security through obscurity, and real security is or A [12] record for the host part. These addresses may have
better provided through authentication and call screening. different degrees of location- and provider-independence and are
often chosen to be mnemonic. In many cases, the host-independent SIP
address can be the same as a user's electronic mail address, but this
is not required. SIP can thus leverage off the domain name system
(DNS) to provide a first-stage location mechanisms. Examples of
host-independent names include
1.5 Call Setup M.Handley@cs.ucl.ac.uk
H.G.Schulzrinne@ieee.org
info@ietf.org
Call setup is a multi-phase procedure. In the first phase, the An address can designate an individual (possibly located at one of
requesting client tries to ascertain the address where it should several end systems), the first available person from a group of
contact the remote user agent or user agent proxy. The local client individuals or a whole group. The form of the address, e.g.,
checks if the user address is location-specific. If so, then that is _________________________
the address used for the remote user agent. If not, the requesting [1] We avoid the term location-independent , since
client looks up the domain part of the user address in the DNS. This the address may indeed refer to a specific location,
provides one or more records giving IP addresses. If a new service e.g., a company department.
(SRV) resource record [5] is returned giving a location server, then
that is the address to contact next. If no relevant resource record
is returned, but an A record is returned, then that is the address to
contact next. If neither a resource record or an A record is
returned, but an MX record is returned, then the mail host is the
address to contact next.
Presuming an address for the invitee is found from the DNS, the sales@example.com , is not sufficient, in general, to determine the
second and subsequent phases basically implement a request-response intent of the caller.
protocol. A session description (typically using SDP format) is sent
to the contact address with an invitation for the user to join the
session.
This request may be sent over a TCP connection or as a single UDP If a user or service chooses to be reachable at an address that is
datagram (the format of both is the same and is described later), and guessable from the person's name and organizational affiliation, the
is sent to a well-known port. traditional method of ensuring privacy by having an unlisted "phone"
number is compromised. However, unlike traditional telephony, SIP
offers authentication and access control mechanisms and can avail
itself of lower-layer security mechanisms, so that client software
can reject unauthorized or undesired call attempts.
If a user agent or conference server is listening on the relevant 1.7 Locating a SIP Server
port, it can send one of the responses below. If no server or agent
is listening, an ICMP port-unreachable response will be triggered
which should cause the TCP connection setup to fail or cause a UDP
send failure on retransmissions.
1.6 Locating a User Call setup may proceed in several phases. A SIP client MUST follow
the following steps to resolve the user part of a callee address. If
a client only supports TCP or UDP, but not both, the respective
address type is omitted.
It is expected that a user is situated at one of several frequented 1. If there is a SRV DNS resource record [11] of type sip.udp
locations. These locations can be dynamically registered with a , contact the listed SIP servers in order of preference
location server for a site (for a local area network or value using UDP as a transport protocol at the port number
organization), and incoming connections can be routed simultaneously listed in the DNS resource record.
to all of these locations if so desired. It is entirely up to the
location server whether the server issues proxy requests for the
requesting user, or if the server instructs the client to redirect
the request.
In general a reply MUST be sent by the same mechanism that the 2. If there is a SRV DNS resource record [11] of type sip.tcp
request was sent by. Hence, if a request was unicast, then the reply , contact the listed SIP servers in order of preference
MUST be unicast back to the requester; if the request was multicast, value using TCP as a transport protocol at the port number
the reply MUST be multicast to the same group to which the request listed in the DNS resource record.
was sent; if the request was sent by TCP, the reply MUST be sent by
TCP.
In all cases where a request is forwarded onwards, each host relaying 3. If there is a DNS MX record [10], contact the hosts listed
the message SHOULD add its own address to the path of the message so in their order of preference at the default port number
that the replies can take the same path back, thus ensuring correct (TBD). For each host listed, first try to contact the
operation through compliant firewalls and loop-free requests. On the server using UDP, then TCP.
reply path, these routing headers MUST be removed as the reply
retraces the path, so that routing internal to sites is hidden. When
a multicast request is made, first the host making the request, then
the multicast address itself are added to the path.
2 Notational Conventions and Generic Grammar 4. Finally, check if there is a DNS CNAME or A record for the
given host and try to contact a SIP server at the one or
more addresses listed, again trying first UDP, then TCP.
Since many of the definitions and syntax are identical to HTTP/1.1, 5. If all of the above methods fail, the caller MAY contact an
this specification only points to the section where they are defined SMTP server at the user's host and use the SMTP EXPN
rather than copying it. For brevity, [HX.Y] is to be taken to refer command to obtain an alternate address and repeat the steps
to Section X.Y of the current HTTP/1.1 specification (RFC 2068). above. As a last resort, a client MAY choose to deliver the
session description to the callee using electronic mail.
All the mechanisms specified in this document are described in both If a server is found using one of the methods below, the other
prose and an augmented Backus-Naur form (BNF) similar to that used in methods are not tried. A client SHOULD rely on ICMP "Port
RFC 2068 [H2.1]. It is described in detail in [6]. Unreachable" messages rather than time-outs to determine that a
server is not reachable at a particular address. A client MAY cache
the result of the reachability steps, but SHOULD start at the
beginning of the sequence when the cached address fails.
In this draft, we use indented and smaller-type paragraphs to provide Implementation note for socket-based programs: For TCP, connect()
background and motivation. returns ECONNREFUSED if there is no server at the designated address;
for UDP, the socket should be bound to the destination address using
connect() rather than sendto() or similar.
3 Protocol Parameters This sequence is modeled after that described for SMTP,
where MX records are to be checked before A records [13].
3.1 SIP Version 1.8 SIP Transactions
applies, with HTTP replaced by SIP. Once the host part has been resolved to a SIP server, the client
sends one or more SIP requests to that server and receives one or
more responses from the server. If the invitation is SIP request is
an invitation, it contains a session description, for example written
in SDP format, that provides the called party with enough information
to join the session.
Applications sending Request or Response messages, as defined by this If TCP is used, request and responses within a single SIP transaction
specification, MUST include an SIP-Version of "SIP/2.0". Use of this are carried over the same TCP connection. Thus, the client SHOULD
version number indicates that the sending application is at least maintain the connection until a final response has been received.
conditionally compliant with this specification. Several SIP requests from the same client to the same server may use
the same TCP connection or may open a new connection for each
request. If the client sent the request sends via unicast UDP, the
response is sent to the source address of the UDP request. If the
request is sent via multicast UDP, the response is directed to the
same multicast address and destination port. For UDP, reliability is
achieved using retransmission (Section 11).
3.2 UCI: Universal Communication Identifier Need motivation why we ALWAYS want to have a multicast
return.
[TBD: describe all legal address formats.] The SIP message format and operation is independent of the transport
protocol.
4 SIP Message The basic message flow is shown in Fig. 1 and Fig. 2, for proxy and
redirect modes, respectively.
All messages are text-based, using the conventions of HTTP/1.1 1.9 Locating a User
[H4.1], except for the additional ability of SIP to use UDP. When
sent over TCP or UDP, multiple requests can be carried in a single
TCP connection or UDP datagram. UDP Datagrams should not normally
exceed the path MTU in size if it is known, or 1,000 bytes if the MTU
is unknown.
4.1 Message Types A callee may move between a number of different end systems over
time. These locations can be dynamically registered with a location
server, typically for a single administrative domain, or a location
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@play :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | | 4: INVITE 5: ring :
: cz@cs.tu-berlin.de ========================> tune =========> play :
: <........................ <......... :
: : 7: 200 OK : 6: 200 OK :
+.....................+ +...............................+
SIP messages consist of requests from client to server and responses ====> SIP request
from server to client. ----> non-SIP protocols
SIP-message = Request | Response ; HTTP/1.1 messages Figure 1: Example of SIP proxy server
Request (section 5) and response (section 6) messages use the generic server may use other protocols, such as finger [14], rwho,
message format of RFC 822 for transferring entities (the payload of multicast-based protocols or operating-system dependent mechanism to
the message). Both types of messages consist of a start-line, one or actively determine the end system where a user is reachable. The
more header fields (also known as "headers"), an empty line (i.e., a location services yield a list of a zero or more possible locations,
line with nothing preceding the CRLF) indicating the end of the possibly even sorted in order of likelihood of success.
header fields, and an optional message-body.
generic-message = start-line The location server can be part of the SIP server or the SIP server
*message-header may use a different protocol (e.g., finger [14] or LDAP [15]) to map
CRLF addresses. A single user may be registered at different locations,
[ message-body ] either because she is logged in at several hosts simultaneously or
because the location server has (temporarily) inaccurate information.
start-line = Request-Line | Status-Line The action taken on receiving a list of locations varies with the
In the interest of robustness, servers SHOULD ignore any empty type of SIP server. A SIP redirect server simply returns the list to
line(s) received where a Request-Line is expected. In other words, if the client sending the request as Location headers (Section 6.17). A
the server is reading the protocol stream at the beginning of a SIP proxy server can sequentially try the addresses until the call is
message and receives a CRLF first, it should ignore the CRLF. 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 .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@play :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | | :
: cz@cs.tu-berlin.de =======================> tune :
: ^ | <....................... :
: . | : 4: 302 Moved : :
+...........|.........+ hgs@play : :
. | : :
. | 5: INVITE hgs@play.cs.columbia.edu 6: ring :
. ==================================================> play :
..................................................... :
7: 200 OK : :
+...............................+
4.2 Message Headers ====> SIP request
----> non-SIP protocols
HTTP header fields, which include general-header (section ), Figure 2: Example of SIP redirect server
request-header (section ), response-header (section ), fields, follow
the same generic format as that given in Section 3.1 of RFC 822. Each
header field consists of a name followed by a colon (":") and the
field value. Field names are case-insensitive. The field value may be
preceded by any amount of LWS, though a single SP is preferred.
Header fields can be extended over multiple lines by preceding each
extra line with at least one SP or HT. Applications SHOULD follow
"common form" when generating HTTP constructs, since there might
exist some implementations that fail to accept anything beyond the
common forms.
message-header = field-name ":" [ field-value ] CRLF noted in the Via header to do so. If it is not the first, it must
issue a redirect response.
field-name = token If a proxy server forwards a SIP request, it SHOULD add itself to the
field-value = *( field-content | LWS ) end of the list of forwarders noted in the Via (Section 6.31)
field-content = <the OCTETs making up the field-value headers. A proxy server also notes whether it is attempting to reach
and consisting of either *TEXT or combinations several possible locations at once ("connection forking"). The Via
of token, tspecials, and quoted-string> trace ensures that replies can take the same path back, thus ensuring
correct operation through compliant firewalls and loop-free requests.
On the reply path, each host most remove its Via, so that routing
internal information is hidden from the callee and outside networks.
When a multicast request is made, first the host making the request,
then the multicast address itself are added to the path.
The order in which header fields with differing field names are As discussed in Section 1.6, a SIP address may designate a group
received is not significant. rather than an individual. A client indicates using the Reach
request header whether it wants to reach the first available
individual or treat the address as a group, to be invited as a whole.
The default is to attempt to reach the first available callee. If
the address is designated as a group address, a proxy server MUST
return the list of individuals instead of attempting to connect to
these.
Multiple message-header fields with the same field-name may be Otherwise, the proxy cannot report errors and call status
present in a message if and only if the entire field-value for that appropriately.
header field is defined as a comma-separated list (i.e., #(values) ).
It MUST be possible to combine the multiple header fields into one
"field-name: field-value" pair, without changing the semantics of
the message, by appending each subsequent field-value to the first,
each separated by a comma. The order in which header fields with the
same field-name are received is therefore significant to the
interpretation of the combined field value, and thus a proxy MUST NOT
change the order of these field values when a message is forwarded.
4.3 Message Body 2 SIP Uniform Resource Locators
The rules for when a message-body is allowed in a message differ for SIP URLs are used within SIP messages to indicate the originator and
requests and responses. recipient of a SIP request, and to specify redirection addresses. A
SIP URL may be embedded in web pages or other hyperlinks to indicate
that a user or service may be called. Within SIP messages, an email
address could have been used, but this would have made it more
difficult to gateway between SIP and other protocols with other
addressing schemes.
The presence of a message-body in a request is signaled by the For greater functionality, because interaction with some resources
inclusion of a Content-Length or Transfer-Encoding header field in may require message headers or message bodies to be specified as well
the request's message-headers. A message-body MAY be included in a as the SIP address, the sip URL scheme is extended to allow setting
request only when the request method allows an entity-body. SIP request-header fields and the SIP message-body.
For response messages, whether or not a message-body is included with A SIP URL follows the guidelines of RFC 1630 [16,17] and takes the
a message is dependent on both the request method and the response following form:
status code (section ). All 1xx (informational) responses MUST NOT
include a message-body. All other responses do include a message-
body, although it may be of zero length.
4.4 Message Length SIP-URL = short-sip-url | full-sip-url
full-sip-url = "sip://" user [ ":" password ] "@" host
url-parameters [ headers ]
short-sip-url = user [ ":" password ] "@" host : port
user = ; defined in RFC 1738 [18]
host = ; defined in RFC 1738
port = *digit
url-parameters = *( ";" url-parameter)
url-parameter = transport-param |
ttl-param | maddr-param
transport-param = "transport=" ( "udp" | "tcp" )
ttl-param = "ttl=" ttl
ttl = 1*3DIGIT ; 0 to 255
maddr-param = "maddr=" maddr
maddr = ; dotted decimal multicast address
headers = "?" header *( " " header )
header = hname "=" hvalue
hname = *urlc
hvalue = *urlc
urlc = ; defined in [17]
When a message-body is included with a message, the length of that Thus a SIP URL can take either a short form or a full form. The short
body is determined by one of the following (in order of precedence): form MAY only be used within SIP messages where the scheme (SIP) can
be assumed. In all other cases, and when parameters are required to
be specified, the full form MUST be used.
1. Any response message which MUST NOT include a message-body Note that all URL reserved characters must be encoded. The special
(such as the 1xx responses) is always terminated by the hname "body" indicates that the associated hvalue is the message-
first empty line after the header fields, regardless of the body of the SIP INVITE request. Within sip URLs, the characters
entity-header fields present in the message. "?", "=", "&" are reserved.
2. Otherwise, a Content-Length header MUST be present. (This Examples of short and full form SIP URLs with identical address are:
requirement differs from HTTP/1.1.) Its value in bytes
represents the length of the message-body.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP. j.doe@big.com
sip://j.doe@big.com
sip://j.doe:secret@big.com;transport=tcp
sip://j.doe@big.com?subject=project
4.5 General Header Fields The password parameter allows to easily specify a call-back address
on a secure web page, but carries the same security risks as all
URL-based passwords and should only be used under special
circumstances where security requirements are low or all transport
paths are secured.
There are a few header fields which have general applicability for Within a SIP message, URLs are used to indicate the source and
both request and response messages. These header fields apply only to intended destination of a request, redirection addresses and the
the message being transmitted. current destination of a request. Normally all these fields will
contain SIP URLs. When additional parameters are not required, the
short form SIP URL can be used unambiguously.
general-header = Date ; Section In some circumstances a non-SIP URL may be used in a SIP message. An
| Transfer-Encoding ; Section example might be making a call from a telephone which is relayed by a
| Via ; Section gateway onto the internet as a SIP request. In such a case, the
source of the call is really the telephone number of the caller, and
so a SIP URL is inappropriate and a phone URL might be used instead.
Thus where SIP specifies user addresses it allows these addresses to
be URLs.
General-header field names can be extended reliably only in Clearly not all URLs are appropriate to be used in a SIP message as a
combination with a change in the protocol version. However, new or user address. It is unlikely, for example, that HTTP or FTP URLs are
experimental header fields may be given the semantics of general useful in this context. The correct behavior when an unknown scheme
header fields if all parties in the communication recognize them to is encountered by a SIP server is defined in the context of each of
be general-header fields. the header fields that use a SIP URL.
SIP URLs can define specific parameters of the request, including the
transport mechanism (UDP or TCP) and the use of multicast to make a
request. These parameters are added after the host and are separated
by semi-colons. For example, to specify to call j.doe@big.com using
multicast to 239.255.255.1 with a ttl of 15, the following URL would
be used:
sip://j.doe@big.com;maddr=239.255.255.1;ttl=15
The transport protocol UDP is to be assumed when a multicast address
is given.
3 SIP Message Overview
Since much of the message syntax is identical to HTTP/1.1, rather
than repeating it here we use [HX.Y] to refer to Section X.Y of the
current HTTP/1.1 specification [9]. In addition, we describe SIP in
both prose and an augmented Backus-Naur form (BNF) [H2.1] described
in detail in [19].
All SIP messages are text-based and use HTTP/1.1 conventions [H4.1],
except for the additional ability of SIP to use UDP. When sent over
TCP or UDP, multiple SIP transactions can be carried in a single TCP
connection or UDP datagram. UDP datagrams, including all headers,
should not normally be larger than the path maximum transmission unit
(MTU) if the MTU is known, or 1500 bytes if the MTU is unknown.
The 1400 bytes accommodates lower-layer packet headers
within the "typical" MTU of around 1500 bytes. There are
few MTU values around 1 kB; the next value is 1006 bytes
for SLIP and 296 for low-delay PPP [20]. Recent studies
[21] indicate that an MTU of 1500 bytes is a reasonable
assumption. Thus, another reasonable value would be a
message size of 950 bytes, to accommodate packet headers
within the SLIP MTU without fragmentation.
A SIP message is either a request from a client to a server, or a
response from a server to a client.
SIP-message = Request | Response ; SIP messages
Both Request (section 4) and Response (section 5) messages use the
generic message format of RFC 822 [22] for transferring entities (the
payload of the message). Both types of message consist of a start-
line, one or more header fields (also known as "headers"), an empty
line (i.e., a line with nothing preceding the carriage-return line-
feed ( CRLF)) indicating the end of the header fields, and an
optional message-body. To avoid confusion with similar-named headers
in HTTP, we refer to the header describing the message body as entity
headers. These components are described in detail in the upcoming
sections.
generic-message = start-line
*message-header
CRLF
[ message-body ]
start-line = Request-Line | Status-Line
Request = Request-Line ; Section 4.1
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ]
Response = Status-Line ; Section 5.1
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ]
In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins
with a CRLF, the CRLF should be ignored.
4 Request
The Request message format is shown below:
general-header = Call-ID ; Section 6.11
| Date ; Section 6.14
| Expires ; Section 6.15
| From ; Section 6.16
| Sequence ; Section 6.26
| Via ; Section 6.31
entity-header = Content-Length ; Section 6.12
| Content-Type ; Section 6.13
request-header = Accept ; Section 6.6
| Accept-Language ; Section 6.7
| Authorization ; Section 6.9
| Organization ; Section 6.18
| Priority ; Section 6.20
| Proxy-Authorization ; Section 6.22
| Reach ; Section 6.24
| Subject ; Section 6.28
| To ; Section 6.29
| User-Agent ; Section 6.30
response-header = Location ; Section 6.17
| Proxy-Authenticate ; Section 6.21
| Public ; Section 6.23
| Retry-After ; Section 6.25
| Server ; Section 6.27
| Warning ; Section 6.32
| WWW-Authenticate ; Section 6.33
Table 1: SIP headers
Request = Request-Line ; Section 4.1
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ] ; Section 8
4.1 Request-Line
5 Request
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 Request-Line = Method SP Request-URI SP SIP-Version CRLF
The method may be either INVITE or CAPABILITY. The request ID may 4.1.1 Methods
be any URL encoded string that can be guaranteed to be globally
unique for the duration of the request. Using the initiator's IP-
address, process id, and instance (if more than one request is being
made simultaneously) satisfies this requirement.
6 Response The following methods are defined:
[H6] applies except that HTTP-Version is replaced by SIP-Version method = "INVITE" | "CONNECTED" | "OPTIONS" | "BYE"
define some HTTP codes. | "REGISTER" | "UNREGISTER"
INVITE: The user or service is being invited to participate in the
session. This method MUST be supported by a SIP server.
CONNECTED: A CONNECTED request confirms that the client has received
a successful response to an INVITE request. See Section 11 for
details. This method MUST be supported by a SIP server.
OPTIONS: The client is being queried as to its capabilities. A server
that believes it can contact the user, such as a user agent
where the user is logged in and has been recently active, MAY
respond to this request with a capability set. Support of this
method is OPTIONAL.
BYE: The client indicates to the server that it wishes to abort the
call attempt. The leaving party can use a Location header field
to indicate that the recipient of request should contact the
named address. This implements the "call transfer" telephony
functionality. A client SHOULD also use this method to indicate
to the callee that it wishes to abort an on-going call attempt.
With UDP, the caller has no other way to signal her intent
to drop the call attempt and the callee side will keep
"ringing". When using TCP, a client MAY also close the
connection to abort a call attempt. Support of this method
is OPTIONAL.
REGISTER: A client uses the REGISTER method to register the address
listed in the request line to a SIP server. In the future, the
server MAY use the source address and port to forward SIP
requests to. A server SHOULD silently drop the registration
after one hour, unless refreshed by the client. A server may set
or lower or higher refresh interval and indicate the interval
through the Expires header (Section 6.15). A single address (if
host-independent) may be registered from several different
clients. Support of this method is OPTIONAL.
Beyond its use as a simple location service, this method is
needed if there are several SIP servers on a single host,
so that some cannot use the default port number. Each such
server would register with a server for the administrative
domain.
UNREGISTER: A client cancels an existing registration established for
the Request-URI with REGISTER with the UNREGISTER method. If
it unregisters a Request-URI unknown to the servers, the server
returns a 200 (OK) response. Support of this method is OPTIONAL.
BYE and REGISTER are experimental and need to be discussed.
Methods that are not supported by a proxy server SHOULD be treated by
that proxy as if they were an INVITE method, and relayed through
unchanged or cause a redirection as appropriate.
Methods that are not supported by a server should cause a "501 Not
Implemented" response to be returned (Section 7).
4.1.2 Request-URI
The Request-URI field is a SIP URL as described in Section 2 or a
general URI. It indicates the user or service that this request is
being addressed to. Unlike the To field, the Request-URI field may
be re-written by proxies. For example, a proxy may perform a lookup
on the contents of the To field to resolve a username from a mail
alias, and then use this username as part of the Request-URI field
of requests it generates.
If a SIP server receives a request contain a URI indicating a scheme
other than SIP which that server does not understand, the server MUST
return a "400 Bad Request" response. It MUST do this even if the To
field contains a scheme it does understand.
4.1.3 SIP Version
Both request and response messages include the version of SIP in use,
and basically follow [H3.1], with HTTP replaced by SIP. To be
conditionally compliant with this specification, applications sending
SIP messages MUST include a SIP-Version of "SIP/2.0".
5 Response
After receiving and interpreting a request message, the recipient After receiving and interpreting a request message, the recipient
responds with an SIP response message. responds with a SIP response message. The response message format is
shown below:
Response = Status-Line ; Section Response = Status-Line ; Section 5.1
*( general-header ; Section *( general-header
| response-header ; Section | response-header
| entity-header ) ; Section | entity-header )
CRLF CRLF
[ message-body ] ; Section [ message-body ] ; Section 8
6.1 Status-Line [H6] applies except that HTTP-Version is replaced by SIP-Version.
Also, SIP defines additional response codes and does not use some
HTTP codes.
5.1 Status-Line
The first line of a Response message is the Status-Line , consisting The first line of a Response message is the Status-Line , consisting
of the protocol version followed by a numeric status code, the of the protocol version ((Section 4.1.3) followed by a numeric
sequence number of the corresponding request and the textual phrase Status-Code and its associated textual phrase, with each element
associated with the status code, with each element separated by SP separated by SP characters. No CR or LF is allowed except in the
characters. No CR or LF is allowed except in the final CRLF sequence. final CRLF sequence.
Note that the addition of a
Status-Line = SIP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF Status-Line = SIP-version SP Status-Code SP Reason-Phrase
CRLF
6.1.1 Status Code and Reason Phrase 5.1.1 Status Codes and Reason Phrases
The Status-Code element is a 3-digit integer result code of the The Status-Code is a 3-digit integer result code that indicates the
attempt to understand and satisfy the request. These codes are fully outcome of the attempt to understand and satisfy the request. The
defined in section10. The Reason-Phrase is intended to give a short Reason-Phrase is intended to give a short textual description of the
textual description of the Status-Code. The Status-Code is intended Status-Code. The Status-Code is intended for use by automata,
for use by automata and the Reason-Phrase is intended for the human whereas the Reason-Phrase is intended for the human user. The client
user. The client is not required to examine or display the Reason- is not required to examine or display the Reason-Phrase.
Phrase
The first digit of the Status-Code defines the class of response. The We provide an overview of the Status-Code below, and provide full
last two digits do not have any categorization role. There are 5 definitions in section 7. The first digit of the Status-Code defines
values for the first digit: the class of response. The last two digits do not have any
categorization role. SIP/2.0 allows 6 values for the first digit:
o 1xx: Informational - Request received, continuing process 1xx: Informational -- request received, continuing process;
o 2xx: Success - The action was successfully received, 2xx: Success -- the action was successfully received, understood, and
understood, and accepted accepted;
o 3xx: Redirection - Further action must be taken in order to 3xx: Redirection -- further action must be taken in order to complete
complete the request the request;
o 4xx: Client Error - The request contains bad syntax or cannot 4xx: Client Error -- the request contains bad syntax or cannot be
be fulfilled fulfilled at this server;
5xx: Server Error -- the server failed to fulfill an apparently valid
request;
o 5xx: Server Error - The server failed to fulfill an apparently 6xx: Global Failure - the request is invalid at any server.
valid request
The individual values of the numeric status codes defined for Presented below are the individual values of the numeric response
SIP/2.0, and an example set of corresponding Reason-Phrase below. The codes, and an example set of corresponding reason phrases for
reason phrases listed here 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 status codes and adds SIP-specific that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
status codes in the starting at 450 to avoid conflicts with newly codes in the range starting at x80 to avoid conflicts with newly
defined HTTP status codes. defined HTTP response codes, and extends these response codes in the
6xx range.
Status-Code = "100" ; Continue SIP response codes are extensible. SIP applications are not required
to understand the meaning of all registered response codes, though
such understanding is obviously desirable. However, applications MUST
understand the class of any response code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 response code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if a client
receives an unrecognized response code of 431, it can safely assume
that there was something wrong with its request and treat the
response as if it had received a 400 response code. In such cases,
user agents SHOULD present to the user the message body returned with
the response, since that message body is likely to include human-
readable information which will explain the unusual status.
6 Header Field Definitions
SIP header fields are similar to HTTP header fields in both syntax
and semantics [H4.2], [H14]. In general the ordering of the header
fields is not of importance (with the exception of Via fields, see
below), but proxies MUST NOT reorder or otherwise modify header
fields other than by adding a new Via field. This allows an
authentication field to be added after the Via fields that will not
be invalidated by proxies.
To, From, and Call-ID header MUST be present in each request with
method INVITE. The Content-Type and Content-Length headers are
required when there is a valid message body (of non-zero length)
associated with the message (Section 8).
A server MUST understand the PEP-Require header.
Other headers may be added as required; a server MAY ignore headers
that it does not understand. A compact form of these header fields is
Status-Code = "100" ; Trying
| "180" ; Ringing
| "200" ; OK | "200" ; OK
| "300" ; Multiple Choices | "300" ; Multiple Choices
| "301" ; Moved Permanently | "301" ; Moved Permanently
| "302" ; Moved Temporarily | "302" ; Moved Temporarily
| "303" ; See Other | "303" ; See Other
| "305" ; Use Proxy | "305" ; Use Proxy
| "380" ; Alternative Service
| "400" ; Bad Request | "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
| "406" ; Not Acceptable
| "407" ; Proxy Authentication Required | "407" ; Proxy Authentication Required
| "408" ; Request Time-out | "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 Entity 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
| "480" ; Temporarily not available
| "500" ; Internal 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 Time-out | "504" ; Gateway Timeout
| "505" ; HTTP Version not supported | "505" ; SIP Version not supported
| "600" ; Busy
| "603" ; Decline
| "604" ; Does not exist anywhere
| "606" ; Not Acceptable
| extension-code | extension-code
extension-code = 3DIGIT extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF> Reason-Phrase = *<TEXT, excluding CR, LF>
SIP status codes are extensible. SIP applications are not required to Figure 3: Status Codes
understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if an
unrecognized status code of 431 is received by the client, it can
safely assume that there was something wrong with its request and
treat the response as if it had received a 400 status code. In such
cases, user agents SHOULD present to the user the entity returned
with the response, since that entity is likely to include human-
readable information which will explain the unusual status.
6.1.2 Response Header Fields also defined in Section 10 for use over UDP when the request has to
fit into a single packet and size is an issue.
The response-header fields allow the request recipient to pass 6.1 General Header Fields
additional information about the response which cannot be placed in
the Status-Line server and about further access to the resource
identified by the Request-URI
response-header = Location ; Section
| Proxy-Authenticate ; Section
| Public ; Section
| Retry-After ; Section
| Server ; Section
| Vary ; Section
| WWW-Authenticate ; Section
Response-header field names can be extended reliably only in There are a few header fields that have general applicability for
both request and response messages. These header fields apply only to
the message being transmitted.
General-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or 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 general
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 general-header fields.
entity-header fields.
7 SIP Message Body
The session description payload gives details of the session the user
is being invited to join. Its Internet media type MUST be given by
the "Content-type:" header field, and the payload length in bytes
MUST be given by the Content-length header field. If the payload has
undergone any encoding (such as compression) then this MUST be
indicated by the Content-encoding: header field, otherwise Content-
encoding: MUST be omitted.
The example below is a request message en route from initiator to
invitee:
INVITE 128.16.64.19/65729 SIP/2.0 6.2 Entity Header Fields
Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Content-type: application/sdp
Content-Length: 187
v=0 Entity-header fields define meta-information about the message-body
o=user1 53655765 2353687637 IN IP4 128.3.4.5 or, if no body is present, about the resource identified by the
s=Mbone Audio request. The term "entity header" is an HTTP 1.1 term where the reply
i=Discussion of Mbone Engineering Issues body may contain a transformed version of the message body. The
e=mbone@somewhere.com original message body is referred to as the "entity". We retain the
c=IN IP4 224.2.0.1/127 same terminology for header fields but usually refer to the "message
t=0 0 body" rather then the entity as the two are the same in SIP.
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 6.3 Request Header Fields
initiator (the first element of the list) towards the invitee. In the
example above, the message was last multicast to the administratively
scoped group 239.128.16.254 with a ttl of 16 from the host
131.215.131.131.
The request header above states that the request was initiated by The request-header fields allow the client to pass additional
mjh@isi.edu (specifically it was initiated from 128.16.64.19, as can information about the request, and about the client itself, to the
be seen from the Via header) and the user being invited is server. These fields act as request modifiers, with semantics
schooler@cs.caltech.edu. equivalent to the parameters on a programming language method
invocation.
In this case, the session description (as stated in the Content-type Request-header field names can be extended reliably only in
header) is a Session Description Protocol (SDP). combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of request-
header fields if all parties in the communication recognize them to
be request-header fields. Unrecognized header fields are treated as
entity-header fields.
The header is terminated by an empty line and is followed by the 6.4 Response Header Fields
session description payload.
If required, the session description can be encrypted using public The response-header fields allow the server to pass additional
key cryptography, and then can also carry private session keys for information about the response which cannot be placed in the Status-
the session. If this is the case, four random bytes are added to the Line. These header fields give information about the server and about
beginning of the session description before encryption and are further access to the resource identified by the Request-URI.
removed after decryption but before parsing.
8 Methods Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as
entity-header fields.
The following methods are defined: 6.5 Header Field Format
INVITE: The user or service is being invited to participate in the Header fields ( general-header, request-header, response-header, and
session. The session description given must be completely entity-header) follow the same generic header format as that given in
acceptable for a "200 OK" response to be given. This method MUST Section 3.1 of RFC 822 [22].
be supported by a SIP server.
OPTIONS: The user or service is being queried as to its capabilities. Each header field consists of a name followed by a colon (":") and
A server that believes it can contact the user (such as a user the field value. Field names are case-insensitive. The field value
agent where the user is logged in and has been recently active) may be preceded by any amount of leading white space (LWS), though a
MAY respond to this request with a capability set. Support of single space (SP) is preferred. Header fields can be extended over
this method is OPTIONAL. multiple lines by preceding each extra line with at least one SP or
horizontal tab (HT). Applications SHOULD follow HTTP "common form"
when generating these constructs, since there might exist some
implementations that fail to accept anything beyond the common forms.
Methods that are not supported by a proxy server SHOULD be treated by message-header = field-name ":" [ field-value ] CRLF
that proxy as if they were an INVITE method, and relayed through field-name = token
unchanged or cause a redirection as appropriate. field-value = *( field-content | LWS )
field-content = < the OCTETs making up the field-value
and consisting of either *TEXT or combinations
of token, tspecials, and quoted-string>
Methods that are not supported by a user agent should cause a "501 The order in which header fields are received is not significant if
Not Implemented" response to be returned. the header fields have different field names. Multiple header fields
with the same field-name may be present in a message if and only if
the entire field-value for that header field is defined as a comma-
separated list (i.e., #(values) ). It MUST be possible to combine the
multiple header fields into one "field-name: field-value" pair,
without changing the semantics of the message, by appending each
subsequent field-value to the first, each separated by a comma. The
order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field
value, and thus a proxy MUST NOT change the order of these field
values when a message is forwarded.
9 Header Field Definitions
SIP header fields are similar to HTTP header fields in both syntax
and semantics. 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.
Field names are not case-sensitive, although their values may be. Field names are not case-sensitive, although their values may be.
Content-Length, Content-Type, To, From header fields are 6.6 Accept
compulsory. Other fields may be added as required. Header fields MUST
be separated by a single linefeed character. The header MUST be
separated from the payload by an empty line (two linefeed
characters).
A compact form of these header fields is also defined in section 10.9 See [H14.1]. This request header field is used only with the OPTIONS
for use over UDP when the request has to fit into a single packet and request to indicate what description formats are acceptable.
size is an issue.
9.1 Accept Example:
See [H14.1]. This header field is used only for the OPTIONS request Accept: application/sdp;level=1, application/x-private
to indicate what description formats are acceptable.
9.2 Accept-Language 6.7 Accept-Language
See [H14.4]. The Accept-Language request header can be used to allow See [H14.4]. The Accept-Language request header can be used to allow
the client to indicate to the server in which language it would the client to indicate to the server in which language it would
prefer to receive reason phrases. This may also be used as a hint by prefer to receive reason phrases. This may also be used as a hint by
the proxy as to which destination to connect the call to (e.g., for the proxy as to which destination to connect the call to (e.g., for
selecting a human operator). selecting a human operator).
9.3 Authentication Example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.8 Allow
See [H14.7].
6.9 Authorization
See [H14.8].
6.10 Authentication
Authentication fields provide a digital signature of the remaining Authentication fields provide a digital signature of the remaining
fields for authentication purposes. They are not yet defined The use fields for authentication purposes. They are not yet defined The use
of authentication headers is optional. If used, authentication of authentication headers is optional. If used, authentication
headers MUST be added to the header after the Via fields and before headers MUST be added to the header after the Via fields and before
the rest of the fields. the rest of the fields.
HS: Ordering and semantics needs work. Maybe we can recycle HS: Should probably re-use S/MIME here rather than invent
the S/MIME work? our own. Maybe better to fold into Authorization field.
9.4 Confirm 6.11 Call-ID
TBD. The Call-ID uniquely identifies a particular invitation. Note that a
single multimedia conference may give rise to several calls, e.g., if
a user invites several different people. Calls to different callee
MUST always use different Call-IDs unless they are the result of a
proxy server "forking" a single request.
9.5 Contact-Host The Call-ID may be any URL-encoded string that can be guaranteed to
TBD. be globally unique for the duration of the request. Using the
initiator's IP-address, process id, and instance (if more than one
request is being made simultaneously) satisfies this requirement.
9.6 From The form local-id@host is recommended, where host is either the
fully qualified domain name or a globally routable IP address, and
local-id depends on the application and operating system of the host,
but is an ID that can be guaranteed to be unique during this session
initiation request.
The request header MUST contain a From request-header field, Call-ID = ( "Call-ID" | "i" ) ":" atom "@" host
indicating the invitation initiator. The field MUST be machine-
usable, as defined my mailbox in RFC 822 (as updated by RFC 1123).
Only a single initiator and a single invited user are allowed to be
specified in a single SIP request.
9.7 Retry-After Example:
The Retry-After response-header field can be used with a 503 Call-ID: 9707211351.AA08181@foo.bar.com
(Service Unavailable) response to indicate how long the service is
expected to be unavailable to the requesting client and with a 404 6.12 Content-Length
(Not Found) or 451* (Busy) response to indicate when the called party
The Content-Length entity-header field indicates the size of the
message-body, in decimal number of octets, sent to the recipient.
Content-Length = "Content-Length" ":" 1*DIGIT
An example is
Content-Length: 3495
Applications SHOULD use this field to indicate the size of the
message-body to be transferred, regardless of the media type of the
entity. Any Content-Length greater than or equal to zero is a valid
value. If no body is present in a message, then the Content-Length
header MAY be omitted or set to zero. Section 8 describes how to
determine the length of the message body.
6.13 Content-Type
The Content-Type entity-header field indicates the media type of the
message-body sent to the recipient.
Content-Type = "Content-Type" ":" media-type
An example of the field is
Content-Type: application/sdp
6.14 Date
See [H14.19].
The Date header field is useful for simple devices without
their own clock.
6.15 Expires
The Expires header field gives the date/time after which the
registration expires.
This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a response-header field and
allows the server to indicate when the client has to re-register. For
INVITE, it is a request-header with which the callee can limit the
validity of an invitation. (For example, if a client wants to limit
how long a search should take at most or when a conference being
invited to is time-limited. A user interface may take this is as a
hint to leave the invitation window on the screen even if the user is
not currently at the workstation.)
The value of this field can be either an HTTP-date or an integer
number of seconds (in decimal), measured from the receipt of the
request.
Expires = "Expires" ":" ( HTTP-date | delta-seconds )
Two example of its use are
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5
6.16 From
Requests MUST and responses SHOULD contain a From header field,
indicating the invitation initiator. The field MUST be a SIP URL as
defined in Section 2. Only a single initiator and a single invited
user are allowed to be specified in a single SIP request. The sense
of To and From header fields is maintained from request to
response, i.e., if the From header is sip://bob@example.edu in the
request then it is MUST also be sip://bob@example.edu in the response
to that request.
The From field is a URL and not a simple SIP address (Section 1.6
address to allow a gateway to relay a call into a SIP request and
still produce an appropriate From field. An example might be a
telephone call relayed into a SIP request where the from field might
contain a phone:// URL. Normally however this field will contain a
sip:// URL in either the long or short form.
If a SIP agent or proxy receives a request sourced From a URL
indicating a scheme other that SIP that is unknown to it, this MUST
NOT be treated as an error.
From = ( "From" | "f" ) ":" *1( ( SIP-URL | URL ) [ comment
] )
Example:
From: mjh@isi.edu (Mark Handley)
6.17 Location
The Location response header can be used with a 2xx or 3xx response
codes to indicate a new location to try. It contains a SIP URL giving
the new location or username to try, or may simply specify addition
transport parameters. For example, a "301 Moved Permanently" response
SHOULD contain a Location field containing the SIP URL giving the
new location and username to try. However, a "302 Moved Temporarily"
MAY give simply the same location and username that was being tried
but specify additional transport parameters such as a multicast
address to try or a change of transport from UDP to TCP or vice
versa.
A user agent or redirect server sending a definitive, positive
response (2xx), SHOULD insert a Location response header indicating
the SIP address under which it is reachable most directly for future
SIP requests. This may be the address of the server itself or that of
a proxy (e.g., if the host is behind a firewall).
Location = ( "Location" | "m" ) ( SIP-URL | URL )
*( ";" location-params )
extension-name = token
extension-value = *( token | quoted-string | LWS | extension-specials)
extension-specials = < any element of tspecials except <"> >
language-tag = < see [H3.10] >
service-tag = "fax" | "IP" | "PSTN" | "ISDN" | "pager" | "voice-mail
| "attendant"
media-tag = < see SDP: "audio" | "video" | ...
feature-list = to be determined
location-params = "q" "=" qvalue
| "mobility" "=" ( "fixed" | "mobile" )
| "class" "=" ( "personal" | "business" )
| "language" "=" 1# language-tag
| "service" "=" 1# service-tag
| "media" "=" 1# media-tag
| "features" "=" 1# feature-list
| "description" "=" quoted-string
| "duplex" "=" ( "full" | "half" | "receive-only" |
"send-only" )
| extension-attributes
extension-attribute = extension-name "=" extension-value
Examples:
Location: sip://hgs@erlang.cs.columbia.edu ;service=IP,voice-mail
;media=audio ;duplex=full ;q=0.7
Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
language=en,es,iw ;q=0.5
Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
duplex=send-only;media=text; q=0.1
Attributes which are unknown should be omitted. New tags for class-
tag and service-tag can be registered with IANA. The media tag uses
Internet media types, e.g., audio, video, application/x-wb, etc. This
is meant for indicating general communication capability, sufficient
for the caller to choose an appropriate address.
6.18 Organization
The Organization request-header fields conveys the name of the
organization to which the callee belongs. It may be inserted by
proxies at the boundary of an organization and may be used by client
software to filter calls.
6.19 PEP
This corresponds to the PEP header in the "Protocol Extension
Protocol" defined in RFC XXXX. The Protocol Extension Protocol (PEP)
is an extension mechanism designed to accommodate dynamic extension
of applications such as SIP clients and servers by software
components. The PEP general header declares new headers and whether
an application must or may understand them. Servers MUST parse this
field and MUST return "420 Bad Extension" when there is a PEP
extension of strength "must" (see RFC XXXX) that they do not
understand.
6.20 Priority
The priority request header signals the urgency of the call to the
callee.
Priority = "Priority" ":" priority-value
priority-value = "urgent" | "normal" | "non-urgent"
Example:
Subject: A tornado is heading our way!
Priority: urgent
6.21 Proxy-Authenticate
See [H14.33].
6.22 Proxy-Authorization
See [H14.34].
6.23 Public
See [H14.35].
6.24 Reach
The Reach request header field allows the client to indicate whether
it wants to reach the group identified by the user part of the
address (value "all") or the first available individual (value
"first"). If not present, a value of "first" is implied. The "do-
not-forward" request prohibits proxies from forwarding the call to
another individual (e.g., the call is personal or the caller does not
want to be shunted to a secretary if the line is busy.) Section 1.6
describes the behavior of proxy servers when resolving group aliases.
Reach = "Reach" ":" 1#( "first" | "all" ) ( "do-not-
forward" )
Example:
Reach: first, do-not-forward
HS: This header is experimental.
6.25 Retry-After
The Retry-After response header field can be used with a "503
Service Unavailable" response to indicate how long the service is
expected to be unavailable to the requesting client and with a "404
Not Found" or "451 Busy" response to indicate when the called party
may be available again. The value of this field can be either an may be available again. The value of this field can be either an
HTTP-date or an integer number of seconds (in decimal) after the time HTTP-date or an integer number of seconds (in decimal) after the time
of the response. of the response.
Retry-After = "Retry-After" ":" ( HTTP-date | delta-seconds ) Retry-After = "Retry-After" ":" ( HTTP-date | delta-seconds
)
Two examples of its use are Two examples of its use are
Retry-After: Fri, 31 Dec 1999 23:59:59 GMT Retry-After: Mon, 21 Jul 1997 18:48:34 GMT
Retry-After: 120 Retry-After: 120
In the latter example, the delay is 2 minutes. In the latter example, the delay is 2 minutes.
9.8 Reason 6.26 Sequence
TBD. The Sequence header field MAY be added by a SIP client making a
request if it needs to distinguish responses to several consecutive
requests sent with the same Call-ID. A Sequence field contains a
single decimal sequence number chosen by the requesting client.
Consecutive different requests made with the same Call-ID MUST
contain strictly monotonically increasing sequence numbers although
the sequence space MAY NOT be contiguous. A server responding to a
request containing a sequence number MUST echo the sequence number
back in the response.
9.9 To Sequence = "Sequence" ":" 1*DIGIT
The To request-header field specifies the invited user, with the Sequence header fields are NOT needed for SIP requests using the
same syntax as the From field. INVITE or OPTIONS methods but may be needed for future methods.
9.10 Via Example:
Sequence: 4711
6.27 Server
See [H14.39].
6.28 Subject
This is intended to provide a summary, or indicate the nature, of the
call, allowing call filtering without having to parse the session
description. (Also, the session description may not necessarily use
the same subject indication as the invitation.)
Subject = ( "Subject" | "s" ) ":" *text
Example:
Subject: Tune in - they are talking about your work!
6.29 To
The To request header field specifies the invited user, with the
same SIP URL syntax as the From field.
To = ( "To" | "t" ) ":" ( SIP-URL | URL ) [ comment ]
If a SIP server receives a request destined To a URL indicating a
scheme other than SIP and that is unknown to it, the server returns a
"400 bad request" response.
Example:
To: sip://operator@cs.columbia.edu (The Operator)
6.30 User-Agent
See [H14.42].
6.31 Via
The Via field indicates the path taken by the request so far. This 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. Initiators MUST add their own Path field to each routing situations.
request. This Path field MUST be the first field in the request.
Subsequent proxies SHOULD each add their own additional Path field In the request path, an initiator MUST add its own Via field to each
which MUST be added before any existing Path fields. When a reply request. This Via field MUST be the first field in the request. Each
passes through a proxy on the reverse path, that proxies Path field subsequent client or proxy that sends the message onwards MUST add
MUST be removed from the reply. its own additional Via field, which MUST be added before any
existing Via fields. Additionally, if the message goes to a
multicast address, an extra Via field is added before all the others
giving the multicast address and TTL.
In the return path, Via fields are processed by a proxy or client
according to the following rules:
o If the first Via field in the reply received is the client's
or server's local address, remove the Via field and process
the reply.
o If the first Via field in a reply you are going to send is a
multicast address, remove that Via field before sending to the
multicast address.
These rules ensure that a client or proxy server only has to check
the first Via field in a reply to see if it needs processing.
When a reply passes through a proxy on the reverse path, that proxies
Via field MUST be removed from the reply.
The format for a Via header is: The format for a Via header is:
Via = "Via" ":" 1#( sent-protocol sent-by [ ttl ] [ comment ] ) Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] )
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 = *DIGIT ttl = 1*3DIGIT ; 0 to 255
TTL is included only if the address is a multicast address. The "ttl" parameter is included only if the address is a multicast
address. The "fanout" parameter indicates that this proxy has
initiated several connection attempts and that subsequent proxies
should not do the same.
10 Status Code Definitions Example:
Via: SIP/2.0/UDP first.example.com:4000 ;fanout
6.32 Warning
The Warning response-header field is used to carry additional
information about the status of a response. Warning headers are sent
with responses using:
Warning = "Warning" ":" 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text
warn-code = 2DIGIT
warn-agent = ( host [ ":" port ] ) | pseudonym
; the name or pseudonym of the server adding
; the Warning header, for use in debugging
warn-text = quoted-string
A response may carry more than one Warning header.
The warn-text should be in a natural language and character set that
is most likely to be intelligible to the human user receiving the
response. This decision may be based on any available knowledge, such
as the location of the cache or user, the Accept-Language field in a
request, the Content-Language field in a response, etc. The default
language is English and the default character set is ISO- 8859-1.
Any server may add Warning headers to a response. New Warning
headers should be added after any existing Warning headers. A proxy
server MUST NOT delete any Warning header that it received with a
response.
When multiple Warning headers are attached to a response, the user
agent SHOULD display as many of them as possible, in the order that
they appear in the response. If it is not possible to display all of
the warnings, the user agent should follow these heuristics:
o Warnings that appear early in the response take priority over
those appearing later in the response.
o Warnings in the user's preferred character set take priority
over warnings in other character sets but with identical
warn-codes and warn-agents.
Systems that generate multiple Warning headers should order them
with this user agent behavior in mind.
Example:
Warning: 606.4 isi.edu Multicast not available
Warning: 606.2 isi.edu Incompatible protocol (RTP/XXP)
6.33 WWW-Authenticate
See [H14.46].
7 Status Code Definitions
The response codes are consistent with, and extend, HTTP/1.1 response 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 are marked with an asterisk, and have codes x50 upwards by HTTP/1.1 have codes x80 upwards to avoid clashes with future HTTP
to avoid clashes with future HTTP response codes, or 6xx which are response codes. Also, SIP defines a new class, 6xx. The default
not used by HTTP. The default behavior for unknown response codes is behavior for unknown response codes is given for each category of
given for each category of codes. codes.
10.1 Informational 1xx 7.1 Informational 1xx
Informational responses indicate that the server or proxy contacted Informational responses indicate that the server or proxy contacted
is performing some further action and does not yet have a definitive is performing some further action and does not yet have a definitive
response. The client SHOULD wait for a further response from the response. The client SHOULD wait for a further response from the
server, and the server SHOULD send such a response without further server, and the server SHOULD send such a response without further
prompting. If UDP transport is being used, the client SHOULD prompting. If UDP transport is being used, the client SHOULD
periodically re-send the request in case the final response is lost. periodically re-send the request in case the final response is lost.
Typically a server should send a "1xx" response if it expects to take Typically a server should send a "1xx" response if it expects to take
more than one second to obtain a final reply. more than one second to obtain a final reply.
10.1.1 100 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.
10.1.2 150 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.
10.2 Successful 2xx 7.2 Successful 2xx
The request was successful and MUST terminate a search. The request was successful and MUST terminate a search.
10.2.1 200 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.
10.3 Redirection 3xx 7.3 Redirection 3xx
3xx responses give information about the user's new location, or 3xx responses give information about the user's new location, or
about alternative services that may be able to satisfy the call. about alternative services that may be able to satisfy the call.
They SHOULD terminate an existing search, and MAY cause the initiator They SHOULD terminate an existing search, and MAY cause the initiator
to begin a new search if appropriate. to begin a new search if appropriate.
10.3.1 300 Multiple Choices 7.3.1 300 Multiple Choices
The requested resource corresponds to any one of a set of The requested resource corresponds to any one of a set of
representations, each with its own specific location, and agent- representations, each with its own specific location, and agent-
driven negotiation information (section 13) is being provided so that driven negotiation (i.e., controlled by the SIP client) is being
the user (or user agent) can select a preferred representation and provided so that the user (or user agent) can select a preferred
redirect its request to that location. communication end point and redirect its request to that location.
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.
If the server has a preferred choice, it SHOULD include the specific The choices SHOULD also be listed as Location fields (Section 6.17).
URL for that representation in the Location field; user agents MAY Unlike HTTP, the SIP response may contain several Location fields.
use the Location field value for automatic redirection. User agents MAY use the Location field value for automatic
redirection or MAY ask the user to confirm a choice.
10.3.2 301 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 Location field because the user has permanently moved and the address
address this response is in reply to is no longer a current address this response is in reply to is no longer a current address for the
for the user. A 301 response MUST NOT suggest any of the hosts in user. A 301 response MUST NOT suggest any of the hosts in the Via
the request's path as the user's new location. path of the request as the user's new location.
10.3.3 302 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 request's path as the user's new location. in the Via path of the request as the user's new location.
10.3.4 350* 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 body of the reply. The alternative services are described in the message body of the
response.
10.4 Request Failure 4xx 7.4 Request Failure 4xx
4xx responses are definite failure responses that MUST terminate the 4xx responses are definite failure responses from a particular
existing search for a user or service. They SHOULD NOT be retried server. The client SHOULD NOT retry the same request without
immediately without modification. modification (e.g., adding appropriate authorization). However, the
same request to a different server may be successful.
10.4.1 400 Bad Request 7.4.1 400 Bad Request
The request could not be understood due to malformed syntax. The request could not be understood due to malformed syntax.
10.4.2 401 Unauthorized 7.4.2 401 Unauthorized
The request requires user authentication. The request requires user authentication.
10.4.3 402 Payment Required 7.4.3 402 Payment Required
Reserved for future use. Reserved for future use.
10.4.4 403 Forbidden 7.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it. The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request should not be repeated. Authorization will not help, and the request should not be repeated.
10.4.5 404 Not Found 7.4.5 404 Not Found
The server has definitive information that the user does not exist at The server has definitive information that the user does not exist at
the domain specified. the domain specified in the Request-URI.
10.4.6 406 Not Acceptable 7.4.6 405 Method Not Allowed
The user's agent was contacted successfully but some aspects of the The method specified in the Request-Line is not allowed for the
session profile (the requested media, bandwidth, or addressing style) address identified by the Request-URI. The response MUST include an
were not acceptable. Allow header containing a list of valid methods for the indicated
address.
10.4.7 450* Decline 7.4.7 407 Proxy Authentication Required
The user's machine was successfully contacted but the user explicitly
does not wish to participate.
10.4.8 451* Busy This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with the proxy. The proxy MUST
return a Proxy-Authenticate header field (section 6.21) containing a
challenge applicable to the proxy for the requested resource. The
client MAY repeat the request with a suitable Proxy-Authorization
header field (section 6.22). SIP access authentication is explained
in section [H11].
The user's machine was successfully contacted but the user is busy, This status code should be used for applications where access to the
or the user does not wish to participate (the ambiguity is communication channel (e.g., a telephony gateway) rather than the
intentional). callee herself requires authentication.
10.5 Server Failure 5xx 7.4.8 408 Request Timeout
The client did not produce a request within the time that the server
was prepared to wait. The client MAY repeat the request without
modifications at any later time.
7.4.9 420 Bad Extension
The server did not understand the protocol extension specified with
strength "must".
7.4.10 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is
currently unavailable (e.g., not logged in or logged in in such a
manner as to preclude communication with the callee). The response
may indicate a better time to call in the Retry-After header. The
user may also be available elsewhere (unbeknownst to this host),
thus, this response does terminate any searches.
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.
10.5.1 500 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
fulfilling the request. fulfilling the request.
10.5.2 501 Not implemented 7.5.2 501 Not implemented
The server does not support the functionality required to fulfill the The server does not support the functionality required to fulfill the
request. This is the appropriate response when the server does not request. This is the appropriate response when the server does not
recognize the request method and is not capable of supporting it for recognize the request method and is not capable of supporting it for
any user. any user.
10.5.3 503 Service Unavailable 7.5.3 502 Bad Gateway
The server, while acting as a gateway or proxy, received an invalid
response from the upstream server it accessed in attempting to
fulfill the request.
7.5.4 503 Service Unavailable
The server is currently unable to handle the request due to a The server is currently unable to handle the request due to a
temporary overloading or maintenance of the server. The implication temporary overloading or maintenance of the server. The implication
is that this is a temporary condition which will be alleviated after is that this is a temporary condition which will be alleviated after
some delay. If known, the length of the delay may be indicated in a some delay. If known, the length of the delay may be indicated in a
Retry-After header. If no Retry-After is given, the client SHOULD Retry-After header. If no Retry-After is given, the client SHOULD
handle the response as it would for a 500 response. handle the response as it would for a 500 response.
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.
10.6 Search Responses 6xx 7.5.5 504 Gateway Timeout
6xx responses are failure responses given whilst trying to locate the The server, while acting as a gateway, did not receive a timely
specified user or service. They are not definitive failures, and response from the upstream server (e.g., a location server) it
SHOULD NOT terminate the search if other possible locations remain accessed in attempting to complete the request.
untried.
10.6.1 600* Search Failure 7.6 Global Failures
The user agent or proxy server understood the user's address, but the
request was unsuccessful in contacting the user. A proxy might return
this error towards the initiator if an attempt to contact a server
failed for an unknown reason.
10.6.2 601* Not known here 6xx responses indicate that a server has definitive information about
a particular user, not just the particular instance indicated in the
Request-URI. All further searches for this user are doomed to failure
and pending searches SHOULD be terminated.
The call was unsuccessful because the user or service was not known 7.6.1 600 Busy
at the address called. This is not a definitive failure; the address
may be valid at another server.
10.6.3 602* Not currently here The callee's end system was contacted successfully but the callee is
busy and does not wish to take the call at this time. The response
may indicate a better time to call in the Retry-After header. If the
callee does not wish to reveal the reason for declining the call, the
callee should use status code 680 instead.
The call was unsuccessful because although the the user or service 7.6.2 603 Decline
was known at the address called, the user or service is not currently
located at this address. This is not a definitive failure; the user
may be contactable at another server.
10.6.4 603* Alternative Address The callee's machine was successfully contacted but the user
explicitly does not wish to participate. The response may indicate a
better time to call in the Retry-After header.
The call was unsuccessful because the user or service is not 7.6.3 604 Does not exist anywhere
available at this location, but one or more alternative non-
definitive locations are suggested to try in addition to any that may
already be being tried. A 603 response MUST NOT suggest any of the
hosts in the request's path as an alternative location.
10.7 Example: Normal Replies The server has authoritative information that the user indicated in
the To request field does not exist anywhere. Searching for the user
elsewhere will not yield any results.
An example reply is given below. The first line of the reply states 7.6.4 606 Not Acceptable
the SIP version number, that it is a "200 OK" reply, which means the
request was successful. The Via header are taken from the request,
and entries are removed hop by hop as the reply retraces the
request's path. A new authentication field is added by the invited
user's agent if required. The session ID is taken directly from the
original request, along with the request header. The original sense
of From field is preserved (i.e, it's the session originator).
In addition, a Contact-host field is added giving details of the The user's agent was contacted successfully but some aspects of the
host the user was located on, or alternatively the relevant proxy session profile (the requested media, bandwidth, or addressing style)
contact point which should be reachable from the invitation were not acceptable.
initiator's host.
SIP/2.0 200 128.16.64.19/65729 A "606 Not Acceptable" reply means that the user wishes to
communicate, but cannot adequately support the session described. The
"604 Not Acceptable" reply MAY contain a list of reasons in a Warning
header describing why the session described cannot be supported.
These reasons can be one or more of:
606.1 Insufficient Bandwidth: The bandwidth specified in the session
description or defined by the media exceeds that known to be
available.
606.2 Incompatible Protocol: One or more protocols described in the
request are not available.
606.3 Incompatible Format: One or more media formats described in the
request is not available.
606.4 Multicast not available: The site where the user is located
does not support multicast.
606.5 Unicast not available: The site where the user is located does
not support unicast communication (usually due to the presence
of a firewall).
Other reasons are likely to be added later. It is hoped that
negotiation will not frequently be needed, and when a new user is
being invited to join a pre-existing lightweight session, negotiation
may not be possible. It is up to the invitation initiator to decide
whether or not to act on a "606 Not Acceptable" reply.
8 SIP Message Body
The session description body gives details of the session the user is
being invited to join. Its Internet media type MUST be given by the
Content-type header field, and the body length in bytes MUST be given
by the Content-Length header field. If the body has undergone any
encoding (such as compression) then this MUST be indicated by the
Content-encoding header field, otherwise Content-encoding MUST be
omitted.
If required, the session description can be encrypted using public
key cryptography, and then can also carry private session keys for
the session. If this is the case, four random bytes are added to the
beginning of the session description before encryption and are
removed after decryption but before parsing.
8.1 Body Inclusion
For a request message, the presence of a body is signaled by the
inclusion of a Content-Length header. A body may be included in a
request only when the request method allows one.
For response messages, whether or not a body is included is dependent
on both the request method and the response message's response code.
All 1xx informational responses MUST NOT include a body. All other
responses MAY include a payload, although it may be of zero length.
8.2 Message Body Length
If no body is present in a message, then the Content-Length header
MAY be omitted or set to zero. When a body is included, its length in
bytes is indicated in the Content-Length header and is determined by
one of the following:
1. Any response message which MUST NOT include a body (such as
the 1xx responses) is always terminated by the first empty
line after the header fields, regardless if any entity-
header fields are present.
2. Otherwise, a Content-Length header MUST be present (this
requirement differs from HTTP/1.1). Its value in bytes
represents the length of the message body.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
9 Examples
9.1 Invitation
9.1.1 Request
The example below is a request message en route from initiator to
invitee:
C->S: INVITE schooler@vlsi.cs.caltech.edu SIP/2.0
Via: SIP/2.0/UDP 239.128.16.254 16
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19
From: mjh@isi.edu (Mark Handley)
Subject: SIP will be discussed, too
To: schooler@cs.caltech.edu (Eve Schooler)
Call-ID: 62729-27@oregon.isi.edu
Content-type: application/sdp
Content-Length: 187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
The first line above states that this is a SIP version 2.0 request.
The Via fields give the hosts along the path from invitation
initiator (the first element of the list) towards the invitee. In the
example above, the message was last multicast to the administratively
scoped group 239.128.16.254 with a ttl of 16 from the host
131.215.131.131
The request header above states that the request was initiated by
mjh@isi.edu the host 128.16.64.19 schooler@cs.caltech.edu is being
invited; the message is currently being routed to
schooler@vlsi.cs.caltech.edu
In this case, the session description is using the Session
Description Protocol (SDP), as stated in the Content-type header.
The header is terminated by an empty line and is followed by a
message body containing the session description.
9.1.2 Reply
The called user agent, directly or indirectly through proxy servers,
indicates that it is alerting ("ringing") the called party:
S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 239.128.16.254 16 Via: SIP/2.0/UDP 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 Call-ID: 62729-27@128.16.64.19
Contact-host: 131.215.131.147 Location: sip://es@jove.cs.caltech.edu
This same format is used for replies for other categories of reply, A sample reply to the invitation is given below. The first line of
except that some of then may require payloads to be carried. the reply states the SIP version number, that it is a "200 OK" reply,
which means the request was successful. The Via headers are taken
from the request, and entries are removed hop by hop as the reply
retraces the path of the request. A new authentication field MAY be
added by the invited user's agent if required. The Call-ID is taken
directly from the original request, along with the remaining fields
of the request message. The original sense of From field is
preserved (i.e., it is the session initiator).
If the invited user's agent requires confirmation of receipt of a In addition, the Location header gives details of the host where the
"200 OK" reply, it may optionally add an additional Confirm: required user was located, or alternatively the relevant proxy contact point
header to the body of the message specifying that an acknowledgment which should be reachable from the caller's host.
is required. This is only permitted with category 2xx replies. An
example is:
SIP/2.0 200 128.16.64.19/65729 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 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
Contact-host: 131.215.131.147 Call-ID: 62729-27@128.16.64.19
Confirm: required Location: sip://es@jove.cs.caltech.edu
In response to such a request, the invitation initiators agent should For two-party Internet phone calls, the response must contain a
retransmit its request with an additional Confirm header, with the description of where to send data to, for example the reply from
value "true" or "false" stating whether the session still exists or schooler to mjh :
no longer exists respectively (see section 7.1 for details). An
example of an confirmation request is:
INVITE 128.16.64.19/65729 SIP/2.0 S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP 239.128.16.254:70 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 1
From: mjh@isi.edu From: mjh@isi.edu
To: schooler@cs.caltech.edu To: schooler@cs.caltech.edu
Confirm: true Call-ID: 62729-27@128.16.64.19
Content-type: application/sdp Location: sip://es@jove.cs.caltech.edu
Content-Length: 187 Content-Length: 102
v=0 v=0
o=user1 2353655765 2353687637 IN IP4 128.3.4.5 o=schooler 4858949 4858949 IN IP4 192.1.2.3
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 5004 RTP/AVP 0
c=IN IP4 131.215.131.147
Such confirmations are still useful when TCP transport is used as The caller confirms the invitation by sending a request to the
they provide application level confirmation rather than transport location named in the Location header:
level confirmation. If they are not used, it is possible that a "200
OK" response may be received after the application making the call
has timed out the call and exited.
10.7.1 Redirects C->S: CONNECTED schooler@jove.cs.caltech.edu SIP/2.0
From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
"603 alternative address" replies and 301 and 302 moved replies 9.1.3 Aborting a Call
should specify another location using the Location field.
An example of a "603 alternative address" reply is: If the caller wants to abort a pending call, it sends a BYE request.
SIP/2.0 603 128.16.64.19/65729 C->S: BYE schooler@jove.cs.caltech.edu
Via: SIP/2.0/UDP 131.215.131.131 1
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
Location: 239.128.16.254 16 Call-ID: 62729-27@128.16.64.19
Content-length:0
In this example, the proxy (131.215.131.131) is being advised to 9.1.4 Redirects
contact the multicast group 239.128.16.254 with a ttl of 16. In
normal situations a server would not suggest a redirect to a local
multicast group unless (as in the above situation) it knows that the
previous proxy or client is within the scope of the local group.
For unicast 603 redirects, a proxy MAY query the suggested location Replies with response codes "301 Moved Permanently" or "302 Moved
itself or send MAY the redirect on back towards the client. For Temporarily" SHOULD specify another location using the Location
multicast 603 redirects, a proxy SHOULD query the multicast address field.
itself rather than sending the redirect back towards the client as
multicast may be scoped and this allows a proxy within the
appropriate scope regions to make the query.
For 301 or 302 redirects, a proxy SHOULD send the redirect on back S->C: SIP/2.0 302 Moved temporarily
towards the client and terminate any other searches it is performing Via: SIP/2.0/UDP 131.215.131.131
for the same request. Multicast 301 or 302 redirects MUST NOT be Via: SIP/2.0/UDP 128.16.64.19
generated. From: mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 62729-27@128.16.64.19
Location: sip://239.128.16.254;ttl=16;transport=udp
Content-length: 0
10.8 Alternative Services In this example, the proxy located at 131.215.131.131 is being
advised to contact the multicast group 239.128.16.254 with a ttl of
16 and UDP transport. In normal situations, a server would not
suggest a redirect to a local multicast group unless, as in the above
situation, it knows that the previous proxy or client is within the
scope of the local group. If the request is redirected to a multicast
group, a proxy server SHOULD query the multicast address itself
rather than sending the redirect back towards the client as multicast
may be scoped; this allows a proxy within the appropriate scope
regions to make the query.
An example of an "350 Alternative Service" reply is: 9.1.5 Alternative Services
SIP/2.0 350 128.16.64.19/32492/2 An example of a "350 Alternative Service" reply is:
S->C: SIP/2.0 350 Alternative Service
Via: SIP/2.0/UDP 131.215.131.131 Via: SIP/2.0/UDP 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
Contact-host: IN IP4 131.215.131.131 Call-ID: 62729-27@128.16.64.19
Location: recorder@131.215.131.131
Content-type: application/sdp Content-type: application/sdp
Content-length: 146 Content-length: 146
v=0 v=0
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 131.215.131.131
t=0 0 t=0 0
m=audio 12345 RTP/AVP 0 m=audio 12345 RTP/AVP 0
In this case, the answering server provides a session description In this case, the answering server provides a session description
that describes an "answering machine". If the invitation initiator that describes an "answering machine". If the invitation initiator
decides to take advantage of this service, it should send an decides to take advantage of this service, it should send an
invitation request to the contact host (131.215.131.131) with the invitation request to the answering machine at 131.215.131.131 with
session description provided. This request should contain a different the session description provided (modified as appropriate for a
session id from the one in the original request. An example would unicast session to contain the appropriate local address and port for
be: the invitation initiator). This request SHOULD contain a different
Call-ID from the one in the original request. An example would be:
INVITE 128.16.64.19/32492/3 SIP/2.0 C->S: INVITE mm-server@131.215.131.131 SIP/2.0
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: 62729-28@128.16.64.19
Content-type: application/sdp Content-type: application/sdp
Content-length: 146 Content-length: 146
v=0 v=0
o=mm-server 2523535 0 IN IP4 128.16.5.31 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 12345 RTP/AVP PCMU m=audio 26472 RTP/AVP 0
Invitation initiators can choose to treat a "350 Alternative Service"
reply as a failure if they wish to do so.
10.8.1 Negotiation
A "406 Not Acceptable" reply means that the user wishes to
communicate, but cannot support the session described adequately. The
"406 Not Acceptable" reply contains a list of reasons why the session
described cannot be supported. These reasons can be one or more of:
406.1 Insufficient Bandwidth: the bandwidth specified in the session
description or defined by the media exceeds that known to be
available.
406.2 Incompatible Protocol: one or more protocols described in the
request is not available.
406.3 Incompatible Format: one or more media formats described in the
request is not available.
406.4 Multicast not available: the site where the user is located
does not support multicast.
406.5 Unicast not available: the site where the user is located does Invitation initiators MAY choose to treat a "350 Alternative Service"
not support unicast communication (usually due to the presence reply as a failure if they wish to do so.
of a firewall).
Other reasons are likely to be added later. It is hoped that 9.1.6 Negotiation
negotiation will not frequently be needed, and when a new user is
being invited to join a pre-existing lightweight session, negotiation
may not be possible. If is up to the invitation initiator to decide
whether or not to act on a "406 Not Acceptable" reply.
A complex example of a "406 Not Acceptable" reply is: An example of a "606 Not Acceptable" reply is:
SIP/2.0 406 128.16.64.19/32492/5 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
Contact-host: 131.215.131.131 Call-ID:62729-27@128.16.64.19
Reason: 406.1, 406.3, 406.4 Location: mjh@131.215.131.131
Content-Type: meta/sdp Warning: 606.1 Insufficient bandwidth (only have ISDN),
606.3 Incompatible format,
606.4 Multicast not available
Content-Type: application/sdp
Content-Length: 50 Content-Length: 50
v=0 v=0
s=Lets talk s=Lets talk
b=CT:128 b=CT:128
c=IN IP4 131.215.131.131 c=IN IP4 131.215.131.131
m=audio 3456 RTP/AVP 7 0 13 m=audio 3456 RTP/AVP 7 0 13
m=video 2232 RTP/AVP 31 m=video 2232 RTP/AVP 31
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.
Invitation initiators MAY choose to treat "406 Not Acceptable" 9.2 OPTIONS Request
replies as a failure if they wish to do so.
10.9 Compact Form A caller Alice can use an OPTIONS request to find out the
capabilities of a potential callee Bob, without "ringing" the
designated address. In this case, Bob indicates that he can be
reached at three different addresses, ranging from voice-over-IP to a
PSTN phone to a pager.
C->S: OPTIONS bob@example.com SIP/2.0
From: alice@anywhere.org (Alice)
To: bob@example.com (Bob)
Accept: application/sdp
S->C: SIP/2.0 200 OK
Location: sip://bob@host.example.com ;service=IP,voice-mail
;media=audio ;duplex=full ;q=0.7
Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
language=en,es,iw ;q=0.5
Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
duplex=send-only;media=text; q=0.1
Alternatively, Bob could have returned a description of
C->S: OPTIONS bob@example.com SIP/2.0
From: alice@anywhere.org (Alice)
To: bob@example.com (Bob)
Accept: application/sdp
S->C: SIP/2.0 200 OK
Content-Length: 81
Content-Type: application/sdp
v=0
m=audio 0 RTP/AVP 0 1 3 99
m=video 0 RTP/AVP 29 30
a:rtpmap:98 SX7300/8000
10 Compact Form
When SIP is carried over UDP with authentication and a complex When SIP is carried over UDP with authentication and a complex
session description, it may be possible that the size of a request or session description, it may be possible that the size of a request or
reply is larger than the MTU (or default 1,000-byte limit if the MTU reply is larger than the MTU. To reduce this problem, a more compact
is not known). To reduce this problem, a more compact form of SIP is form of SIP is also defined by using alternative names for common
also defined by using alternative names for common header fields. header fields. These short forms are NOT abbreviations, they are
These short forms are NOT abbreviations, they are field names. No field names. No other abbreviations are allowed.
other abbreviations are allowed.
short field name long field name note short field name long field name note
a Confirm from "acknowledge"
c Content-Type c Content-Type
e Content-Encoding e Content-Encoding
f From f From
h Contact-Host i Call-ID
l Content-Length l Content-Length
m Location from "moved" m Location from "moved"
r Reason s Subject
t To t To
v Via v Via
Thus the header in section ?? could also be written: Thus the header in section 9.1 could also be written:
INVITE 128.16.64.19/65729 SIP/2.0 INVITE schooler@vlsi.caltech.edu SIP/2.0
p:IN IP4 UDP 239.128.16.254 1 16 v:SIP/2.0/UDP 239.128.16.254 16
p:IN IP4 UDP 131.215.131.131 1 v:SIP/2.0/UDP 131.215.131.131
p:IN IP4 UDP 128.16.64.19 1 v:SIP/2.0/UDP 128.16.64.19
f:mjh@isi.edu f:mjh@isi.edu
t:schooler@cs.caltech.edu t:schooler@cs.caltech.edu
i:62729-27@128.16.64.19
c:application/sdp c:application/sdp
l:187 l: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
skipping to change at page 28, line 27 skipping to change at page 48, line 34
Mixing short field names and long field names is allowed, but not Mixing short field names and long field names is allowed, but not
recommended. Servers MUST accept both short and long field names for recommended. Servers MUST accept both short and long field names for
requests. Proxies MUST NOT translate a request between short and long requests. Proxies MUST NOT translate a request between short and long
forms if authentication fields are present. forms if authentication fields are present.
11 SIP Transport 11 SIP Transport
SIP is defined so it can use either UDP or TCP as a transport SIP is defined so it can use either UDP or TCP as a transport
protocol. protocol.
UDP has advantages over TCP from a performance point of view, as the 11.1 Achieving Reliability For UDP Transport
SIP application can keep control of the precise timing of
retransmissions, and can also make simultaneous call attempts to many
potential locations of many users without needing to keep TCP
connection state for each connection.
TCP has the advantage that clients are simpler to implement because 11.1.1 General Operation
no retransmission timing code needs to be written and also that it is
possible to have a single server serving SIP and HTTP with very
little extra code.
With UDP, all the additional reliability code is in the client. It is SIP assumes no additional reliability from IP. Requests or replies
recommended that servers SHOULD implement both TCP and UDP may be lost. A SIP client SHOULD simply retransmit a SIP request
functionality as the additional server code required is very small. periodically with timer T1 (default value of T1: once a second) until
it receives a response, or until it has reached a set limit on the
number of retransmissions. The default limit is 20.
Clients MAY implement either TCP or UDP transport or both as they see SIP requests and replies are matched up by the client using the
fit. Call-ID header field; thus, a server can only have one outstanding
request per call at any given time.
11.1 Reliability using UDP transport HS: A transaction or request ID would remove this
limitation.
The Session Invitation Protocol is straightforward in operation. Only If the reply is a provisional response, the initiating client SHOULD
the initiating client needs to keep any state regarding the current continue retransmitting the request, albeit less frequently, using
connection attempt. SIP assumes no additional reliability from IP. timer T2. The default retransmission interval T2 is 5 seconds.
Requests or replies may be lost. A SIP client SHOULD simply After the server sends a final response, it cannot be sure the client
retransmit a SIP request until it receives a reply, or until it has has received the response, and thus SHOULD cache the results for at
reached some maximum number of timeouts and retransmissions. If the least 30 seconds to avoid having to, for example, contact the user or
reply is merely a 1xx Informational progress report, the initiating user location server again upon receiving a retransmission.
client SHOULD still continue retransmitting the request, albeit less
frequently.
When the remote user agent or server sends a final 2xx or 4xx 11.1.2 INVITE
response (not a 1xx report), it cannot be sure the client has
received the response, and thus SHOULD cache the results until a
connection setup timeout has occurred to avoid having to contact the
user again. The server MAY also choose to cache 3xx or 6xx responses
if the cost of obtaining the response outweighs the cost of caching
it.
It is possible that a user can be invited successfully, but that the Special considerations apply for the INVITE method.
reply that the user was successfully contacted may not reach the
invitation initiator. If the session still exists but the initiator
gives up on including the user, the contacted user has sufficient
information to be able to join the session. However, if the session
no longer exists because the invitation initiator "hung up" before
the reply arrived and the session was to be two-way, the conferencing
system should be prepared to deal with this circumstance.
One solution is for the initiator to acknowledge the invitee's "200 1. After receiving an invitation, considerable time may elapse
OK" reply. Although not required, in the case of a successful before the server can determine the outcome. For example,
invitation the invited user's agent can make a confirmation request the called party may be "rung" or extensive searches may be
in its "200 OK" reply. In this case the initiator's agent sends a performed, so delays can reach several tens of seconds.
single request with a reply Confirm: true if the request was still
valid or a reply Confirm: false if it was not so that a premature
hang-up can be detected without a long timeout. Such a confirmation
request may be retransmitted by the invited user's agent if it so
desired. Confirmation requests can only be made with "200 OK"
replies, and only the invitation initiator's agent may issue the
actual confirmation.
Only a "200 OK" reply warrants such a confirmation handshake, because 2. It is possible that the invitation request reaches the
it is the only situation where user-relevant state may be callee and the callee is willing to take the call, but that
instantiated anywhere other than at the initiator's client. In all the final response (200 OK, in this case) is lost on the
other cases, it is not necessary that state is maintained. In way to the caller. If the session still exists but the
particular, when a server makes multiple proxy requests, "5xx Server initiator gives up on including the user, the contacted
Error" and "6xx Search Response" replies do not immediately get user has sufficient information to be able to join the
passed back to the invitation initiator, and so no end-to-end session. However, if the session no longer exists because
acknowledgment of a failed request is possible. the invitation initiator "hung up" before the reply arrived
and the session was to be two-way, the conferencing system
should be prepared to deal with this circumstance.
11.2 Reliability using TCP transport 3. If a telephony user interface is modeled or if we need to
interface to the PSTN, the caller will provide "ringback",
a signal that the callee is being alerted. Once the callee
picks up, the caller needs to know so that it can enable
the voice path and stop ringback. The callee's response to
the invitation could get lost. Unless the response is
transmitted reliably, the caller will continue to hear
ringback while the callee assumes that the call exists.
TCP is a reliable transport protocol, and so we do not need to define 4. The client has to be able to terminate an on-going request,
additional reliability mechanisms. However, we must define rules for e.g., because it is no longer willing to wait for the
connection closedown under normal operation. connection or search to succeed. One cannot rely on the
absence of request retransmission, since the server would
have to continue honoring the request for several request
retransmission periods, that is, possible tens of seconds
if only one or two packets can be lost.
The normal mode of operation is for the client (or proxy acting as a The first problem is solved by indicating progress to the caller: the
client) to make a TCP connection to the well-known port of a host server returns a provisional response indicating it is searching or
housing a SIP server. The client then sends the SIP request to the ringing the user.
server over this connection and waits for one or more replies. The
client MAY close the connection at any time.
The server MAY send one or more 1xx Informational responses before The server retransmits the final response at intervals of T3 (default
sending a single 2xx, 3xx, 4xx, 5xx or 6xx reply. The server MUST NOT value of T3 = 2 seconds) until it receives a CONNECTED request for
send more than one reply, with the exception of 1xx responses. The the same Call-ID or until it has retransmitted the final response 10
server SHOULD NOT close the TCP connection until it has sent its times. The CONNECTED request is acknowledged only once. If the
request is syntactically valid and the Request-URI matches that in
the INVITED request with the same Call-ID, the server answers with
status code 200, otherwise with status code 400.
Fig. 4 and 5 show the client and server state diagram for
invitations.
11.2 Connection Management for TCP
A single TCP connection can serve one or more SIP transactions. A
transaction contains zero or more provisional responses followed by
exactly one final response.
The client MAY close the connection at any time. Closing the
connection before receiving a final response signals that the client
wishes to abort the request.
The server SHOULD NOT close the TCP connection until it has sent its
final response, at which point it MAY close the TCP connection if it final response, at which point it MAY close the TCP connection if it
wishes to. However, normally it is the client's responsibility to wishes to. However, normally it is the client's responsibility to
close the connection. close the connection.
If the server leaves the connection open, and if the client so 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 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 requests from the same family of protocols (such as HTTP or stream
control commands). control commands).
The same application-level confirmation rules apply for TCP as for 12 Behavior of SIP Servers
UDP.
12 Searching 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 basic assumption of SIP is that a location server at the user's A proxy server always inserts a Via header field containing their
home site either knows where the user resides, knows how to locate own address into requests it issues that are caused by an incoming
the user, or at the very least knows another location server that request.
possibly might have a better idea. How these servers get this
information is outside the scope of SIP itself, but it is expected
that many different user-location services will exist for some time.
SIP is designed so that it does not care which location service SIP
servers actually employ.
12.1 Proxy servers: Relaying and Redirection 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 |
+===========+
|
|
| -
| ------
| INVITE
+------v v
T1 +-----------+
------ | Calling |-------------------+
INVITE +-----------+ |
+------| | | |
+----------------+ | |
| | |
| | |
| | |
| | |
| +------v v v-----| |
| T2 +-----------+ 1xx |
| ------ | Ringing | --- |
| INVITE +-----------+ - |
| +------| | | |-----+ |
| | +--------------+ |
| 2xx | | >=300 |
| --------- | 2xx | ----- |
| CONNECTED | --------- | - |
| | CONNECTED | |
+----------------+ | | |
+------v | v v v
2xx +-----------+ +-----------+
--------- | Connected | | Failure |
CONNECTED +-----------+ +-----------+
+------|
If a proxy server receives a request for a user whose location it event
does not know, and for whom it has no better idea where the user -------
might be, then the server should return a "601 Not Currently Here" message
reply message.
If the server does have an idea how to contact the user, it can Figure 4: State transition diagram of client for INVITE method
either forward (relay) the request itself, or can redirect the
invitation initiator to another client that is more likely to know by
sending a 603, 301 or 302 response as appropriate. It can also
gateway the request into some other form if some other invitation
protocol is in use in a region containing the invited user, though in
doing so the server is likely to give up being stateless.
Whether to relay the request or to redirect the request is up to the using transport protocol B. If not stated explicitly, rules apply to
server itself. For example, if the server is on a firewall machine, any combination of transport protocols. For conciseness, we only
then it will probably have to relay the request to servers inside the describe behavior with UDP and TCP, but the same rules apply for any
firewall. Additionally, if a local multicast group is to be used for unreliable datagram or reliable protocol, respectively.
user location, then the server is likely to relay the request.
However, if the user is currently away from home, relaying the
request makes little sense, and the server is more likely (though not
compelled) to send a redirect reply. SIP is policy-free on this
issue. In general, local searches are likely to be better performed
by relaying whereas wide-area searches are likely to be better
performed by redirection.
When SIP uses UDP transport, clients and servers can make multiple +===========+
simultaneous requests to locate a particular user at low cost. This +------------>| Initial |<-------------+
greatly speeds up any search for the user, and in most cases will | +===========+ |
only result in one successful response. Although several simultaneous | | |
paths may reach the same host, successful responses arriving from | failure | |
multiple paths will not confuse the client as they should all contain | ----------- | INVITE |
the same successful host address. However, this does imply that paths | 3xx,4xx,5xx | ------ |
with many levels of relaying should be strongly discouraged as if the | | 1xx |
request is fanned out at each hop and relayed many times, request | +------v v |
implosions could result. Thus servers that are not the first hop | INVITE +-----------+ |
servers in a chain of servers SHOULD NOT make multiple parallel | ------ | Searching | |
requests, but should send a redirection response with multiple | 1xx +-----------+ |
alternatives. Thus a firewall host can still perform a parallel | +------| | | +---------------->+
search but can control the fanout of the search. | | | |
| | | callee picks up |
+----------------+ | --------------- |
| 200 |
| | BYE
+------v v v-----| | ---
INVITE +-----------+ T3 | 200
------ | Answered | --- |
1xx +-----------+ 200 |
+------| | | |-----+ |
| +---------------->+
| |
| CONNECTED |
| --------- |
| 200 |
| |
+------v v |
CONNECTED +-----------+ |
--------- | Connected | |
200 +-----------+ |
+------| | |
+-----------------+
12.2 Parallel Searches: Initiator Behavior event
-------
message
The session initiator may make a parallel search for a user. This can Figure 5: State transition diagram of server for INVITE method
occur when DNS resolution results in multiple addresses, or when
contacting a remote server results in a "603 Alternative Address"
response containing multiple addresses to try. All such parallel
searches for the same SIP request MUST contain the same SIP Id,
though the sequence number (given in the Path field) SHOULD be
different for each of the parallel searches.
Whilst performing a parallel search, different responses may result The detailed connection behavior for UDP and TCP is described in
from different servers, and it is important for the initiating client Section 11.
to handle these responses correctly. In general, the following rules
apply:
o If a 2xx response is received, the invitation was successful, 12.1 Redirect Server
the user should be informed and all pending requests should be
terminated and/or ignored.
o If a 4xx response is received the invitation has definitively A redirect server does not issue any SIP requests of its own. It can
failed, the user should be informed, and all pending requests return a response that accepts, refuses or redirects the request.
should be terminated and/or ignored. After receiving a request, a redirect server proceeds through the
following steps:
o If a 3xx response is received, the search should be terminated 1. If the request cannot be answered immediately (e.g.,
and all pending requests should be terminated and/or ignored. because a location server needs to be contacted), it
However, further action MAY be taken depending on the actual returns one or more provisional responses.
reply without informing the user or alternatively the
invitation MAY be regarded as having failed in which case the
user MUST be informed.
o If a 5xx or 6xx response is received, the particular server 2. Once the server has gathered the list of alternative
responding is removed from the parallel search and the search locations or has decided to accept or refuse the call, it
continues. If a "603 Alternative Address" response is returns the final response. This ends the SIP transaction.
received, the search may be expanded to include those servers
listed in the response that have not already responded. The
user SHOULD NOT be informed unless there are no other servers
left to try, in which case the user MUST be informed.
o If a 1xx response is received, the search continues. The user The redirect server maintains transaction state for the whole SIP
MAY be informed as deemed appropriate. transaction. Servers in user agents are redirect servers.
12.3 Parallel Searches: Proxy Behavior 12.2 Proxies Issuing Single Unicast Requests
In the same way that an Initiating Client can discover multiple Proxies in this category issue at most a single unicast request for
addresses to try, a proxy server can also discover multiple addresses each incoming SIP request, that is, they do not "fork" requests.
that it may try. For a proxy server to be stateless, it must not make Servers may choose to always operate in the mode described in Section
multiple SIP requests because it would then be possible to return a 12.3.
5xx or 6xx response to the Initiating Client and afterwards obtain a
definitive answer. To be able to make multiple parallel SIP requests,
it must keep state as to the replies it has already received and MUST
NOT return any reply other than 1xx informational replies until it
has received a definitive reply or has no further addresses to try.
Thus faced with DNS resolution giving multiple addresses, a proxy 12.2.1 UDP--UDP Proxy Server
server that wishes to be stateless should only send a SIP request to
the first address. Similarly a stateless proxy should not attempt to
send SIP request to multiple addresses given in a "603 Alternative
Address" response that is returned it it, but should forward such a
response back towards the initiator.
Proxies that wish to keep state should follow the following rules The UDP--UDP server can forward the request and any responses. It
regarding responses obtained during a parallel search: does not have to maintain any state for the SIP transaction. UDP
reliability is assured by the next redirect server in the server
chain.
o If a 2xx response is received, the invitation was successful, 12.2.2 UDP--TCP Proxy Server
the 2xx response should be forwarded back towards the
initiator, and all pending requests should be terminated and/or
ignored.
o If a 4xx response is received the invitation has definitively A proxy server issuing a single request over TCP maintains state for
failed, the 4xx response should be forwarded back towards the the whole SIP transaction indexed by the Call-ID.
initiator, and all pending requests should be terminated and/or
ignored.
o If a 3xx response is received the invitation is regarded by If it receives a UDP retransmission of the same request for an
the proxy as having failed, the 3xx response should be existing session, it retransmits the last response received from the
forwarded back towards the initiator, the search should be TCP side. Any changes in the message body compared to the last
terminated and all pending requests should be terminated and/or request for the Call-ID are silently ignored. (Otherwise, the proxy
ignored. 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.
o If a 5xx or 6xx response is received, the particular server After the cache entry has been expired, the server cannot tell
responding is removed from the parallel search and the search whether an incoming request is actually a retransmission of an older
continues. If a "603 Alternative Address" response is request, where the TCP side has terminated. It will treat it as a new
received, the search may be expanded to include those servers request.
listed in the response that have not already responded. No
response other than a periodic "100 Trying" response should be
send towards the initiator unless there are no other servers
left to try, in which case a response SHOULD be sent as
described below.
o If a 1xx response is received, the search continues. The 1xx 12.3 Proxy Server Issuing Several Requests
response MAY be forwarded towards the initiator as appropriate.
If a proxy had exhausted its search and still not obtained a All requests carry the same Call-ID. For unicast, each of the
definitive response (it received only 1xx, 5xx, and 6xx responses) requests has a different (host-dependent) Request-URI. For
the proxy should cache these responses and return the first response multicast, a single request is issued, likely with a host-independent
from the following ordered list: Request-URI. A client receiving a multicast query does not have to
check whether the host part of the Request-URI matches its own host
or domain name. To avoid response implosion, servers SHOULD NOT
answer multicast requests with a 404 (Not Found) status code.
Servers MAY decide not to answer multicast requests if their response
would be 5xx.
1. 503 Service Unavailable; The server MAY respond to the request immediately with a "100 Trying"
response; otherwise it MAY wait until either the first response to
its requests or the UDP retransmission interval.
2. 500 Server Internal Error; The following pseudo-code describes the behavior of a proxy server
issuing several requests in response to an incoming request. The
function request(a) sends a SIP request to address a.
await_response() waits until a response is received and returns the
response. request_close(a) closes the TCP connection to client with
address a; this is optional. response(s, l, L) sends a response to
the client with status s and list of locations L, with l entries.
ismulticast() returns 1 if the location is a multicast address and
zero otherwise. The variable timeleft indicates the amount of time
left until the maximum response time has expired. The variable
recurse indicates whether the server will recursively try addresses
returned through a 3xx response. A server MAY decide to recursively
try only certain addresses, e.g., those which are within the same
domain as the proxy server. Thus, an initial multicast request may
trigger additional unicast requests.
3. 501 Not Implemented; int N = 0; /* number of connection attempts */
address_t address[]; /* list of addresses */
location[]; /* list of locations */
int heard = 0; /* number of sites heard from */
int class; /* class of status code */
int best = 1000; /* best response so far */
int timeleft = 120; /* sample timeout value */
int loc = 0; /* number of locations */
struct { /* response */
int status; /* response status */
char *location; /* redirect locations */
address_t a; /* address of respondent */
} r;
int i;
4. any other 5xx error not yet defined; if (multicast) {
request(address[0]);
} else {
N = /* number of addresses to try */
for (i = 0; i < N; i++) {
request(address[i]);
}
}
5. 600 Search Failure; while (timeleft > 0 && (heard < N || multicast)) {
r = await_response();
class = r.status / 100;
6. 602 Not Currently Here; if (class >= 2) {
heard++;
if (tcp) request_close(a);
}
7. 601 Not Known Here; if (class == 2) {
best = r.status;
break;
}
else if (class == 3) {
/* A server may optionally recurse. The server MUST check whether
* it has tried this location before and whether the location is
* part of the Via path of the incoming request. This check is
* omitted here for brevity. Multicast locations MUST NOT be
* returned to the client if the server is not recursing.
*/
if (recurse) {
multicast = 0;
N++;
request(r.location);
} else if (!ismulticast(r.location)) {
locations[loc++] = r.location;
best = r.status;
}
}
else if (class == 4) {
if (best >= 400) best = r.status;
}
else if (class == 5) {
if (best >= 500) best = r.status;
}
else if (class == 6) {
best = r.status;
break;
}
}
/* We haven't heard anything useful from anybody. */
if (best == 1000) {
best = 404;
}
if (best/100 != 3) locs = 0;
response(best, locs, locations);
8. any other 6xx error response not yet defined. When operating in this mode, a proxy server MUST ignore any responses
received for Call-IDs that it does not have a pending transaction
for. (If server were to forward responses not belonging to a current
transaction using the Via field, the requesting client would get
confused if it has just issued another request using the same Call-
ID.)
If a proxy has exhausted its search and the only response it has 13 Security Considerations
received has been "603 Alternative Address", then the proxy should
send a "600 Search Failure" response if any connection attempt timed
out or failed, or it should send "602 Not Currently Here" if two or
more "603 Alternative Address" responses only provide references to
each other.
12.4 Change of Transport at a Proxy 13.1 Confidentiality
Editors note: this section is still incomplete. Several Unless SIP transactions are protected by lower-layer security
options exist for where the responsibility should lie for mechanisms such as SSL , an attacker may be able to eavesdrop on call
retransmissions from proxies between TCP and UDP transport. establishment and invitations and, through the Subject header field
This section generally assumes local retransmission, but or the session description, gain insights into the topic of
end-to-end transmission through a chain of proxies is also conversation.
possible.
It is possible that a proxy server will receiver a request using TCP 13.2 Integrity
and relay it onwards using UDP or vice-versa. SIP does not assume
end-to-end reliability even when the initiating client is using TCP,
but a SIP client sending a request over TCP MAY assume that the
request has been received by the server it sent the request to.
Retransmission of the request is then not the responsibility of the
client. However, a called user agent SHOULD NOT assume that a 2xx
success response has been received by the invitation initiator, even
if all the path fields in the request indicated TCP transport because
it cannot be certain all those TCP connections still exist. If the
called user agent requires knowledge that the response did reach the
invitation initiator, it MAY add a Confirm: required field to the
reply as it would if the response was sent using UDP.
In the following, the term "TCP-UDP proxy" is used to mean a proxy Unless SIP transactions are protected by lower-layer security
that received a request using TCP and relayed it using UDP. Similarly mechanisms such as SSL , an active attacker may be able to modify SIP
a "TCP-UDP proxy" receives a reply using UDP and should relay it requests.
using TCP.
12.4.1 Retransmission from a TCP-UDP Proxy 13.3 Access Control
A proxy receiving a request with TCP transport and forwarding that SIP requests are not authenticated unless the SIP Authorization and
request using UDP becomes responsible for retransmission of the WWW-Authenticate headers are being used. The strengths and weaknesses
request as required and for timing out the request if no answer is of these authentication mechanisms are the same as for HTTP.
forthcoming.
12.4.2 Retransmissions arriving at a UDP-TCP Proxy 13.4 Privacy
A proxy receiving a request using UDP transport and forwarding that User location and SIP-initiated calls may violate a callee's privacy.
request using TCP transport may have have SIP request state An implementation SHOULD be able to restrict, on a per-user basis,
associated with that TCP connection or SIP response state associated what kind of location and availability information is given out to
with it. certain classes of callers.
If such a proxy receives a retransmission of the UDP request whilst A Summary of Augmented BNF
in the state or awaiting a response (i.e, has request state), it
SHOULD NOT forward the duplicate request into the TCP connection
unless the request has been modified, but instead SHOULD respond with
a "100 Trying" response sent back towards the initiator.
Note: This behavior is different from a UDP-UDP proxy which MUST In this specification we use the Augmented Backus-Naur Form notation
forward the retransmitted request and MAY additionally respond with a described in [19]. For quick reference, the following is a brief
"100 Trying" response sent back towards the initiator. summary of the main features of this ABNF.
If such a proxy receives a retransmission of the UDP request in "abc"
response state (i.e, it has already sent a definitive response) then The case-insensitive string of characters "abc" (or "Abc",
the proxy MAY retransmit that response if it has cached it. "aBC", etc.);
Alternatively if it has not cached the response, it SHOULD resend the
request towards the called user agent, either via an existing TCP
connection if there is one or via a new TCP connection if there is
not, to obtain a retransmission of the response. In the latter case,
the proxy MAY additionally respond with a "100 Trying" response sent
back towards the initiator.
Note: This behavior is the same as a UDP-to-UDP proxy in the same %d32
circumstances. The character with ASCII code decimal 32 (space);
12.4.3 Confirmation arriving at a TCP-UDP Proxy *term
zero of more instances of term;
One possible event that may occur is that whilst performing a search 3*term
using UDP, a response may arrive that should be relayed back towards three or more instances of term;
the initiator using TCP, but the TCP connection has been terminated
by the initiator. In this case the proxy MUST NOT attempt to relay
the response (by opening a TCP connection) and should terminate any
outstanding search. In this circumstance only, if the response was a
"200 OK" response with a Confirm: required field, the proxy MAY
resend the request to the Contact Host with a Confirm: false field
to speed hang-up discovery at the called user agent.
12.4.4 Confirmation sent from a UDP-TCP Proxy 2*4term
two, three or four instances of term;
Normally a response that arrives at a proxy using TCP that should be [ term ]
sent back towards the initiator using UDP should be sent once, and term is optional;
should only be resent if the request is resent from the UDP proxy
closer to the initiator. However, this does not allow for reliable
confirmation.
13 Using Variants for Terminal Negotiation term1 term2 term3
set notation: term1, term2 and term3 must all appear but
their order is unimportant;
Redirection allows the called party to indicate several communication term1 | term2
alternatives to the caller using the 300 (Multiple Choices) response, either term1 or term2 may appear but not both;
all reachable using a single published communication identifier.
The Alternates header in the response contains the variant list. #term
The response may contain an entity, typically of content type a comma separated list of term;
text/html, providing guidance to the user. The calling user agent is
free to ignore this part and solely rely on the Alternates header.
SIP/2.0 300 Multiple Choices 2#term
Date: Thu, 06 Mar 1997 10:08:55 GMT a comma separated list of term containing at least 2 items;
Alternates:
{"hgs@erlang.cs.columbia.edu" 0.9 {mobility fixed} {class business}
{service IP, voice-mail} {media all} {duplex full}},
{"+12129397042" 0.8 {mobility fixed} {class business}
{service POTS} {media audio} {duplex full}},
{"+12129397000" 0.7 {mobility fixed} {class business}
{service ISDN, attendant} {media audio} {duplex full}
{language en, es, iw}},
{"+12125551212" 0.6 {mobility mobile} {class personal}
{service POTS} {media audio} {duplex full}}
}
Content-Type: text/html
Content-Length: 283
<html> 2#4term
You can reach <a href="http://www.cs.columbia.edu/~doe">John Doe</a> at a comma separated list of term containing 2 to 4 items.
<ul> Common Tokens
<li><a href="sip://hgs@erlang.cs.columbia.edu">Internet telephony</a>
<li><a href="phone://+1219397042">analog phone</a> Certain tokens are used frequently in the BNF this document, and not
defined elsewhere. Their meaning is well understood but we include it
here for completeness.
<li>... CR = %d13 ; carriage return character
LF = %d10 ; line feed character
CRLF = CR LF ; typically the end of a line
SP = %d32 ; space character
TAB = %d09 ; tab character
LWS = *( SP | TAB) ; linear whitespace
DIGIT = "0" .. "9" ; a single decimal digit
</dl> Changes
</html>
13.1 Variant Description Since version -01, the following things have changed:
A variant can be described in a machine-readable way with a variant o Added personal note to "Searching" section indicating that 6xx
description [7]. codes may not be necessary. Added figures.
variant-description = o Initial author's note removed; dated.
"{" <"> UCI <"> communication-quality *variant-attribute "}"
communications-quality = qvalue o Introduction rewritten to give quick, concise overview as to
variant-attribute = "{" "mobility" ( "fixed" | "mobile" ) "}" what SIP does.
| "{" "class" ( "personal" | "business" ) "}"
| "{" "language" 1#language-tag "}"
| "{" "service" 1#service-tag "}"
| "{" "media" 1#media-tag "}"
| "{" "features" feature-list "}"
| "{" "description" quoted-string "}"
| "{" "duplex" ( "full" | "half" | "receive-only" |
"send-only" ) "}"
| extension-attribute
extension-attribute = "{" extension-name extension-value "}" o Conference control (tight vs. loose) seems less and less
extension-name = token appropriate. All share some state such as notions of
extension-value = *( token | quoted-string | LWS | membership; some (ITU versions) tend to keep it in a central
extension-specials ) server, others distribute it. Some state is synchronized at
extension-specials = <any element of tspecials except <"> and "}"> 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.)
language-tag = <see [H3.10]> o Added list of related protocols to emphasize that this is part
service-tag = fax | IP | POTS | pager | voice-mail | of a whole architecture.
attendant
media-tag = <see SDP: audio | video | ... >
feature-list =
Attributes which are unknown should be omitted. New tags for class- o Terminology: user always reminds me of controlled substances;
tag and service-tag can be registered with IANA. The media tag uses thus, this term is avoided where better terminology exists.
Internet media types, e.g., audio, video, application/x-wb, etc. This Since this protocol sits at the boundary between traditional
is meant for indicating general communication capability, not the Internet and telephony services, some of the terminology
support for specific encodings. It should be sufficient to allow the familiar in that realm is introduced.
caller to choose an appropriate communication address.
14 Acknowledgments 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
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
between items. Missing ":" added. CRLF was already in the
message definition. Also, unlike RFC 822 and HTTP, the
definition did not allow spaces between the field name and the
colon.
o Added (reluctantly) password to URL. It's no worse than ftp
and needed to easily call from a secure web page, without
having to type in a password manually.
o Added port to SIP URL to specify non-standard port.
o CAPABILITIES to OPTIONS for closer alignment with HTTP and
RTSP;
o Path to Via for closer alignment with HTTP and RTSP;
o Content type meta changed to application, since "meta" doesn't
exist as a top-level Internet media type.
o Formatting closer to HTTP and RTSP.
o Explain relationship to H.323.
B Open Issues
RELIABLE: How to provide reliability?
BYE: Use of BYE method?
REGISTER: Use of REGISTER method?
H.323: Interaction with H.323 and H.245.
TRANSACTION: Should we have a transaction id in addition to a call
ID? Call-IDs are for the end system, but a transaction ID is for
a single SIP exchange. This is useful for Internet telephony,
where a single call may trigger several transactions.
C Acknowledgments
We wish to thank the members of the IETF MMUSIC WG for their comments We wish to thank the members of the IETF MMUSIC WG for their comments
and suggestions. This work is based, inter alia, on [8,9]. and suggestions. This work is based, inter alia, on [23,24].
15 Authors' Addresses Parameters of the terminal negotiation mechanism were influenced by
Scott Petrack's CMA design.
D 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
Dept. of Computer Science Dept. of Computer Science
Columbia University Columbia University
1214 Amsterdam Avenue 1214 Amsterdam Avenue
New York, MY 10027 New York, NY 10027
USA electronic mail: schulzrinne@cs.columbia.edu USA
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
16 Bibliography E Bibliography
[1] M. Handley, "SDP: Session description protocol," Internet Draft, [1] R. Pandya, "Emerging mobile and personal communication systems,"
Internet Engineering Task Force, Nov. 1996. Work in progress. IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.
[2] M. Handley, "Sap: Session announcement protocol," Internet Draft, [2] R. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin,
"Resource reservation protocol (RSVP) -- version 1 functional
specification," Internet Draft, Internet Engineering Task Force, June
1997. Work in progress.
[3] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
transport protocol for real-time applications," RFC 1889, Internet
Engineering Task Force, Jan. 1996.
[4] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
Mar. 1997. Work in progress.
[5] M. Handley, "SAP: Session announcement protocol," Internet Draft,
Internet Engineering Task Force, Nov. 1996. Work in progress. Internet Engineering Task Force, Nov. 1996. Work in progress.
[3] P. Lantz, "Usage of H.323 on the Internet," Internet Draft, [6] M. Handley and V. Jacobson, "SDP: Session description protocol,"
Internet Draft, Internet Engineering Task Force, Mar. 1997. Work in
progress.
[7] P. Lantz, "Usage of H.323 on the Internet," Internet Draft,
Internet Engineering Task Force, Feb. 1997. Work in progress. Internet Engineering Task Force, Feb. 1997. Work in progress.
[4] S. Bradner, "Key words for use in RFCs to indicate requirement [8] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," Internet Draft, Internet Engineering Task Force, Jan. 1997. levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
Work in progress.
[5] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the [9] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-Lee,
"Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
Engineering Task Force, Jan. 1997.
[10] C. Partridge, "Mail routing and the domain system," STD 14, RFC
974, Internet Engineering Task Force, Jan. 1986.
[11] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the
location of services (DNS SRV)," RFC 2052, Internet Engineering Task location of services (DNS SRV)," RFC 2052, Internet Engineering Task
Force, Oct. 1996. Force, Oct. 1996.
[6] D. Crocker, "Augmented BNF for syntax specifications: ABNF," [12] P. Mockapetris, "Domain names - implementation and
specification," STD 13, RFC 1035, Internet Engineering Task Force,
Nov. 1987.
[13] R. Braden, "Requirements for internet hosts - application and
support," STD 3, RFC 1123, Internet Engineering Task Force, Oct.
1989.
[14] D. Zimmerman, "The finger user information protocol," RFC 1288,
Internet Engineering Task Force, Dec. 1991.
[15] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
protocol," RFC 1777, Internet Engineering Task Force, Mar. 1995.
[16] T. Berners-Lee, "Universal resource identifiers in WWW: a
unifying syntax for the expression of names and addresses of objects
on the network as used in the world-wide web," RFC 1630, Internet
Engineering Task Force, June 1994.
[17] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
locators (URL): Generic syntax and semantics," Internet Draft,
Internet Engineering Task Force, May 1997. Work in progress.
[18] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
1994.
[19] 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.
[7] K. Holtman and A. Muntz, "Transparent Content Negotiation in [20] J. Mogul and S. Deering, "Path MTU discovery," RFC 1191,
HTTP," Internet Draft, Internet Engineering Task Force, Nov. 1997. Internet Engineering Task Force, Nov. 1990.
Work in progress.
[8] E. M. Schooler, "Case study: multimedia conference control in a [21] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1.
Reading, Massachusetts: Addison-Wesley, 1994.
[22] D. Crocker, "Standard for the format of ARPA internet text
messages," STD 11, RFC 822, Internet Engineering Task Force, Aug.
1982.
[23] E. M. Schooler, "Case study: multimedia conference control in a
packet-switched teleconferencing system," Journal of Internetworking: 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.
[9] H. Schulzrinne, "Personal mobility for multimedia services in the [24] H. Schulzrinne, "Personal mobility for multimedia services in
Internet," in European Workshop on Interactive Distributed Multimedia the Internet," in European Workshop on Interactive Distributed
Systems and Services , (Berlin, Germany), Mar. 1996. Multimedia Systems and Services , (Berlin, Germany), Mar. 1996.
Table of Contents
1 Introduction ........................................ 2
1.1 Overview of SIP Functionality ....................... 2
1.2 Finding Multimedia Sessions ......................... 3
1.3 Terminology ......................................... 4
1.4 Definitions ......................................... 4
1.5 Protocol Properties ................................. 6
1.5.1 Minimal State ....................................... 6
1.5.2 Transport-Protocol Neutral .......................... 6
1.5.3 Text-Based .......................................... 6
1.6 SIP Addressing ...................................... 6
1.7 Locating a SIP Server ............................... 8
1.8 SIP Transactions .................................... 9
1.9 Locating a User ..................................... 9
2 SIP Uniform Resource Locators ....................... 12
3 SIP Message Overview ................................ 14
4 Request ............................................. 15
4.1 Request-Line ........................................ 16
4.1.1 Methods ............................................. 17
4.1.2 Request-URI ......................................... 18
4.1.3 SIP Version ......................................... 18
5 Response ............................................ 18
5.1 Status-Line ......................................... 19
5.1.1 Status Codes and Reason Phrases ..................... 19
6 Header Field Definitions ............................ 20
6.1 General Header Fields ............................... 22
6.2 Entity Header Fields ................................ 22
6.3 Request Header Fields ............................... 22
6.4 Response Header Fields .............................. 22
6.5 Header Field Format ................................. 23
6.6 Accept .............................................. 23
6.7 Accept-Language ..................................... 24
6.8 Allow ............................................... 24
6.9 Authorization ....................................... 24
6.10 Authentication ...................................... 24
6.11 Call-ID ............................................. 24
6.12 Content-Length ...................................... 25
6.13 Content-Type ........................................ 25
6.14 Date ................................................ 26
6.15 Expires ............................................. 26
6.16 From ................................................ 27
6.17 Location ............................................ 27
6.18 Organization ........................................ 29
6.19 PEP ................................................. 29
6.20 Priority ............................................ 29
6.21 Proxy-Authenticate .................................. 29
6.22 Proxy-Authorization ................................. 29
6.23 Public .............................................. 30
6.24 Reach ............................................... 30
6.25 Retry-After ......................................... 30
6.26 Sequence ............................................ 31
6.27 Server .............................................. 31
6.28 Subject ............................................. 31
6.29 To .................................................. 32
6.30 User-Agent .......................................... 32
6.31 Via ................................................. 32
6.32 Warning ............................................. 33
6.33 WWW-Authenticate .................................... 34
7 Status Code Definitions ............................. 34
7.1 Informational 1xx ................................... 35
7.1.1 100 Trying .......................................... 35
7.1.2 180 Ringing ......................................... 35
7.2 Successful 2xx ...................................... 35
7.2.1 200 OK .............................................. 35
7.3 Redirection 3xx ..................................... 35
7.3.1 300 Multiple Choices ................................ 35
7.3.2 301 Moved Permanently ............................... 36
7.3.3 302 Moved Temporarily ............................... 36
7.3.4 380 Alternative Service ............................. 36
7.4 Request Failure 4xx ................................. 36
7.4.1 400 Bad Request ..................................... 36
7.4.2 401 Unauthorized .................................... 37
7.4.3 402 Payment Required ................................ 37
7.4.4 403 Forbidden ....................................... 37
7.4.5 404 Not Found ....................................... 37
7.4.6 405 Method Not Allowed .............................. 37
7.4.7 407 Proxy Authentication Required ................... 37
7.4.8 408 Request Timeout ................................. 37
7.4.9 420 Bad Extension ................................... 37
7.4.10 480 Temporarily Unavailable ......................... 38
7.5 Server Failure 5xx .................................. 38
7.5.1 500 Server Internal Error ........................... 38
7.5.2 501 Not implemented ................................. 38
7.5.3 502 Bad Gateway ..................................... 38
7.5.4 503 Service Unavailable ............................. 38
7.5.5 504 Gateway Timeout ................................. 39
7.6 Global Failures ..................................... 39
7.6.1 600 Busy ............................................ 39
7.6.2 603 Decline ......................................... 39
7.6.3 604 Does not exist anywhere ......................... 39
7.6.4 606 Not Acceptable .................................. 39
8 SIP Message Body .................................... 40
8.1 Body Inclusion ...................................... 40
8.2 Message Body Length ................................. 40
9 Examples ............................................ 41
9.1 Invitation .......................................... 41
9.1.1 Request ............................................. 41
9.1.2 Reply ............................................... 42
9.1.3 Aborting a Call ..................................... 43
9.1.4 Redirects ........................................... 44
9.1.5 Alternative Services ................................ 44
9.1.6 Negotiation ......................................... 45
9.2 OPTIONS Request ..................................... 46
10 Compact Form ........................................ 47
11 SIP Transport ....................................... 48
11.1 Achieving Reliability For UDP Transport ............. 48
11.1.1 General Operation ................................... 48
11.1.2 INVITE .............................................. 49
11.2 Connection Management for TCP ....................... 50
12 Behavior of SIP Servers ............................. 50
12.1 Redirect Server ..................................... 53
12.2 Proxies Issuing Single Unicast Requests ............. 53
12.2.1 UDP--UDP Proxy Server ............................... 53
12.2.2 UDP--TCP Proxy Server ............................... 53
12.3 Proxy Server Issuing Several Requests ............... 54
13 Security Considerations ............................. 56
13.1 Confidentiality ..................................... 56
13.2 Integrity ........................................... 56
13.3 Access Control ...................................... 56
13.4 Privacy ............................................. 56
A Summary of Augmented BNF ............................ 57
B Open Issues ......................................... 60
C Acknowledgments ..................................... 60
D Authors' Addresses .................................. 61
E Bibliography ........................................ 61
 End of changes. 

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