draft-ietf-mmusic-sip-06.txt   draft-ietf-mmusic-sip-07.txt 
Internet Engineering Task Force MMUSIC WG Internet Engineering Task Force MMUSIC WG
Internet Draft Handley/Schulzrinne/Schooler Internet Draft Handley/Schulzrinne/Schooler/Rosenberg
draft-ietf-mmusic-sip-06.txt ISI/Columbia U./Caltech ietf-mmusic-sip-07.txt ISI/Columbia U./Caltech/Bell Labs.
June 13, 1998 July 16, 1998
Expires: November 1998 Expires: December 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
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Distribution of this document is unlimited. Distribution of this document is unlimited.
ABSTRACT ABSTRACT
Many styles of multimedia conferencing are likely to co- The Session Initiation Protocol (SIP) is an application-
exist on the Internet, and many of them share the need to layer control (signaling) protocol for creating,
invite users to participate. The Session Initiation modifying and terminating sessions with one or more
Protocol (SIP) is a simple protocol designed to enable participants. These sessions include Internet multimedia
the invitation of users to participate in such multimedia conferences, Internet telephone calls and multimedia
sessions. It is not tied to any specific conference distribution. Members in a session can communicate via
control scheme. In particular, it aims to enable user multicast or via a mesh of unicast relations, or a
mobility by relaying and redirecting invitations to a combination of these.
user's current location.
SIP invitations used to create sessions carry session
descriptions which allow participants to agree on a set
of compatible media types. It supports user mobility by
proxying and redirecting requests to the user's current
location. Users can register their current location. SIP
is not tied to any particular conference control
protocol. SIP is designed to be independent of the
lower-layer transport protocol and can be extended with
additional capabilities.
This document is a product of the Multi-party 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.
1 Introduction 1 Introduction
1.1 Overview of SIP Functionality 1.1 Overview of SIP Functionality
The Session Initiation Protocol (SIP) is an application-layer The Session Initiation Protocol (SIP) is an application-layer control
protocol that can establish, modify and terminate multimedia sessions protocol that can establish, modify and terminate multimedia sessions
or calls. These multimedia sessions include multimedia conferences, or calls. These multimedia sessions include multimedia conferences,
distance learning, Internet telephony and similar applications. SIP distance learning, Internet telephony and similar applications. SIP
can invite a person to both unicast and multicast sessions; the can invite both persons and "robots", such as a media storage
initiator does not necessarily have to be a member of the session to service. SIP can invite parties to both unicast and multicast
which it is inviting users. Media and participants can be added to an sessions; the initiator does not necessarily have to be a member of
existing session. SIP can be used to "call" both persons and the session to which it is inviting. Media and participants can be
"robots", for example, to invite a media storage device to record an added to an existing session.
ongoing conference or to invite a video-on-demand server to play a
video into a conference. (SIP does not directly control these
services, however; see RTSP [1].)
SIP can be used to initiate sessions as well as invite members to SIP can be used to initiate sessions as well as invite members to
sessions that have been advertised and established by other means. sessions that have been advertised and established by other means.
(Sessions may be advertised using multicast protocols such as SAP Sessions may be advertised using multicast protocols such as SAP,
[2], electronic mail, news groups, web pages or directories (LDAP), electronic mail, news groups, web pages or directories (LDAP), among
among others.) others.
SIP transparently supports name mapping and redirection services, SIP transparently supports name mapping and redirection services,
allowing the implementation of ISDN and Intelligent Network telephony allowing the implementation of ISDN and Intelligent Network telephony
subscriber services. These facilities also enable personal mobility subscriber services. These facilities also enable personal mobility
services, this is defined as: "Personal mobility is the ability of services, this is defined as: "Personal mobility is the ability of
end users to originate and receive calls and access subscribed end users to originate and receive calls and access subscribed
telecommunication services on any terminal in any location, and the telecommunication services on any terminal in any location, and the
ability of the network to identify end users as they move. Personal ability of the network to identify end users as they move. Personal
mobility is based on the use of a unique personal identity (i.e., mobility is based on the use of a unique personal identity (i.e.,
mobility complements terminal mobility, i.e., the ability to maintain mobility complements terminal mobility, i.e., the ability to maintain
communications when moving a single end system from one network to communications when moving a single end system from one subnet to
another. another.
SIP supports some or all of five facets of establishing and SIP supports five facets of establishing and terminating multimedia
terminating multimedia communications: communications:
User location: determination of the end system to be used for User location: determination of the end system to be used for
communication; communication;
User capabilities: determination of the media and media parameters to User capabilities: determination of the media and media parameters to
be used; be used;
User availability: determination of the willingness of the called User availability: determination of the willingness of the called
party to engage in communications; party to engage in communications;
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called and calling party; called and calling party;
Call handling: including transfer and termination of calls. Call handling: including transfer and termination of calls.
SIP can also initiate multi-party calls using a multipoint control SIP can also initiate multi-party calls using a multipoint control
unit (MCU) or fully-meshed interconnection instead of multicast. unit (MCU) or fully-meshed interconnection instead of multicast.
Internet telephony gateways that connect PSTN parties may also use Internet telephony gateways that connect PSTN parties may also use
SIP to set up calls between them. SIP to set up calls between them.
SIP is designed as part of the overall IETF multimedia data and SIP is designed as part of the overall IETF multimedia data and
control architecture [4] currently incorporating protocols such as control architecture currently incorporating protocols such as RSVP
RSVP (RFC 2205 [5]) for reserving network resources, the real-time (RFC 2205 [2]) for reserving network resources, the real-time
transport protocol (RTP) (RFC 1889 [6]) for transporting real-time transport protocol (RTP) (RFC 1889 [3]) for transporting real-time
data and providing QOS feedback, the real-time streaming protocol data and providing QOS feedback, the real-time streaming protocol
(RTSP) [1] for controlling delivery of streaming media, the session (RTSP) (RFC 2326 [4]) for controlling delivery of streaming media,
announcement protocol (SAP) [2] for advertising multimedia sessions the session announcement protocol (SAP) for advertising multimedia
via multicast and the session description protocol (SDP) (RFC 2327 sessions via multicast and the session description protocol (SDP)
[7]) for describing multimedia sessions, but the functionality and (RFC 2327 [5]) for describing multimedia sessions. However, the
operation of SIP does not depend on any of these protocols. functionality and operation of SIP does not depend on any of these
protocols.
SIP may also be used in conjunction with other call setup and SIP may also be used in conjunction with other call setup and
signaling protocols. In that mode, an end system uses SIP protocol signaling protocols. In that mode, an end system uses SIP exchanges
exchanges to determine the appropriate end system address and to determine the appropriate end system address and protocol from a
protocol from a given address that is protocol-independent. For given address that is protocol-independent. For example, SIP could be
example, SIP may be used to determine that the party may be reached used to determine that the party may be reached via H.323, obtain the
via H.323, obtain the H.245 gateway and user address and then use H.245 gateway and user address and then use H.225.0 to establish the
H.225.0 to establish the call [8]. In another example, it may be used call.
to determine that the callee is reachable via the public switched
telephone network (PSTN) and indicate the phone number to be called, In another example, it may be used to determine that the callee is
possibly suggesting an Internet-to-PSTN gateway to be used. reachable via the public switched telephone network (PSTN) and
indicate the phone number to be called, possibly suggesting an
Internet-to-PSTN gateway to be used.
SIP does not offer conference control services such as floor control SIP does not offer conference control services such as floor control
or voting and does not prescribe how a conference is to be managed, or voting and does not prescribe how a conference is to be managed,
but SIP can be used to introduce conference control protocols. SIP but SIP can be used to introduce conference control protocols. SIP
does not allocate multicast addresses. does not allocate multicast addresses.
SIP can invite users to sessions with and without resource SIP can invite users to sessions with and without resource
reservation. SIP does not reserve resources, but may convey to the reservation. SIP does not reserve resources, but may convey to the
invited system the information necessary to do this. Quality-of- invited system the information necessary to do this. Quality-of-
service guarantees, if required, may depend on knowing the full service guarantees, if required, may depend on knowing the full
membership of the session; this information may or may not be known membership of the session; this information may or may not be known
to the agent performing session invitation. to the agent performing session invitation.
1.2 Terminology 1.2 Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED", In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [9] and and "OPTIONAL" are to be interpreted as described in RFC 2119 [6] and
indicate requirement levels for compliant SIP implementations. indicate requirement levels for compliant SIP implementations.
1.3 Definitions 1.3 Definitions
This specification uses a number of terms to refer to the roles This specification uses a number of terms to refer to the roles
played by participants in SIP communications. The definitions of played by participants in SIP communications. The definitions of
client, server and proxy are similar to those used by the Hypertext client, server and proxy are similar to those used by the Hypertext
Transport Protocol (HTTP) (RFC 2068 [10]). The following terms have Transport Protocol (HTTP) (RFC 2068 [7]). The following terms have
special significance for SIP. special significance for SIP.
Call: A call consists of all participants in a conference invited by Call: A call consists of all participants in a conference invited by
a common source. A SIP call is identified by a globally unique a common source. A SIP call is identified by a globally unique
call-id (Section 6.12). Thus, if a user is, for example, invited call-id (Section 6.12). Thus, if a user is, for example, invited
to the same multicast session by several people, each of these to the same multicast session by several people, each of these
invitations will be a unique call. A point-to-point Internet invitations will be a unique call. A point-to-point Internet
telephony conversation maps into a single SIP call. In a MCU- telephony conversation maps into a single SIP call. In a MCU-
based call-in conference, each participant uses a separate call based call-in conference, each participant uses a separate call
to invite himself to the MCU. to invite himself to the MCU.
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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. User agents and proxies contain directly with a human user. User agents and proxies contain
clients (and servers). clients (and servers).
Conference: A multimedia session (see below), identified by a common Conference: A multimedia session (see below), identified by a common
session description. A conference may have zero or more members session description. A conference may have zero or more members
and includes the cases of a multicast conference, a full-mesh and includes the cases of a multicast conference, a full-mesh
conference and a two-party "telephone call", as well as conference and a two-party "telephone call", as well as
combinations of these. combinations of these. Any number of calls may be used to
create a conference.
Downstream: Requests sent in the direction from the caller to the Downstream: Requests sent in the direction from the caller to the
callee. callee.
Final response: A response that terminates a SIP transaction, as Final response: A response that terminates a SIP transaction, as
opposed to a provisional response that does not. All 2xx, 3xx, opposed to a provisional response that does not. All 2xx, 3xx,
4xx, 5xx and 6xx responses are final. 4xx, 5xx and 6xx responses are final.
Initiator, calling party, caller: The party initiating a conference Initiator, calling party, caller: The party initiating a conference
invitation. Note that the calling party does not have to be the invitation. Note that the calling party does not have to be the
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before forwarding it. before forwarding it.
Redirect server: A redirect server is a server that accepts a SIP Redirect server: A redirect server is a server that accepts a SIP
request, maps the address into zero or more new addresses and request, maps the address into zero or more new addresses and
returns these addresses to the client. Unlike a proxy server , returns these addresses to the client. Unlike a proxy server ,
it does not initiate its own SIP request. Unlike a user agent it does not initiate its own SIP request. Unlike a user agent
server , it does not accept calls. server , it does not accept calls.
Registrar: A registrar is server that accepts REGISTER requests. A Registrar: A registrar is server that accepts REGISTER requests. A
registrar is typically co-located with a proxy or redirect registrar is typically co-located with a proxy or redirect
server. server and may offer location services.
Ringback: Ringback is the signaling tone produced by the calling Ringback: Ringback is the signaling tone produced by the calling
client's application indicating that a called party is being client's application indicating that a called party is being
alerted (ringing). alerted (ringing).
Server: A server is an application program that accepts requests in Server: A server is an application program that accepts requests in
order to service requests and sends back responses to those order to service requests and sends back responses to those
requests. Servers are either proxy, redirect or user agent requests. Servers are either proxy, redirect or user agent
servers. servers or registrars.
Session: "A multimedia session is a set of multimedia senders and Session: "A multimedia session is a set of multimedia senders and
receivers and the data streams flowing from senders to receivers and the data streams flowing from senders to
receivers. A multimedia conference is an example of a multimedia receivers. A multimedia conference is an example of a multimedia
session." (RFC 2327, [7]) (A session as defined for SDP may session." (RFC 2327 [5]) (A session as defined for SDP may
comprise one or more RTP sessions.) As defined, a callee may be comprise one or more RTP sessions.) As defined, a callee may be
invited several times, by different calls, to the same session. invited several times, by different calls, to the same session.
If SDP is used, a session is defined by the concatenation of the If SDP is used, a session is defined by the concatenation of the
user name , session id , network type , address type and address user name , session id , network type , address type and address
elements in the origin field. elements in the origin field.
(SIP) transaction: A SIP transaction occurs between a client and a (SIP) transaction: A SIP transaction occurs between a client and a
server and comprises all messages from the first request sent server and comprises all messages from the first request sent
from the client to the server up to a final (non-1xx) response from the client to the server up to a final (non-1xx) response
sent from the server to the client. A transaction is identified sent from the server to the client. A transaction is identified
by the CSeq sequence number (Section 6.16) within a single call by the CSeq sequence number (Section 6.16) within a single call
leg The ACK request has the same CSeq number as the leg The ACK request has the same CSeq number as the
corresponding INVITE request, but comprises a transaction on corresponding INVITE request, but comprises a transaction of
its own. its own.
Upstream: Responses sent in the direction from the called client to Upstream: Responses sent in the direction from the called client to
the caller. the caller.
URL-encoded: A character string encoded according to RFC 1738, URL-encoded: A character string encoded according to RFC 1738,
Section 2.2 [11]. Section 2.2 [8].
User agent client (UAC), calling user agent: A user agent client is a User agent client (UAC), calling user agent: A user agent client is a
client application that initiates the SIP request. client application that initiates the SIP request.
User agent server (UAS), called user agent: A user agent server is a User agent server (UAS), called user agent: A user agent server is a
server application that contacts the user when a SIP request is server application that contacts the user when a SIP request is
received and that returns a response on behalf of the user. The received and that returns a response on behalf of the user. The
response may accept, reject or redirect the call. response may accept, reject or redirect the request.
An application program may be capable of acting both as a client and An application program may be capable of acting both as a client and
a server. For example, a typical multimedia conference control a server. For example, a typical multimedia conference control
application would act as a client to initiate calls or to invite application would act as a user agent client to initiate calls or to
others to conferences and as a user agent server to accept invite others to conferences and as a user agent server to accept
invitations. The properties of the different SIP server types are invitations. The properties of the different SIP server types are
summarized in Table 1. summarized in Table 1.
property redirect proxy user agent property redirect proxy user agent registrar
server server server server server server
_______________________________________________________ __________________________________________________________________________
also acts as client no yes no also acts as a SIP client no yes no no
return 1xx status yes yes yes returns 1xx status yes yes yes rare
return 2xx status no yes yes returns 2xx status no yes yes yes
return 3xx status yes yes yes returns 3xx status yes yes yes yes
return 4xx status yes yes yes returns 4xx status yes yes yes yes
return 5xx status yes yes yes returns 5xx status yes yes yes yes
return 6xx status no yes yes returns 6xx status no yes yes no
insert Via header no yes no inserts Via header no yes no no
accept ACK no yes yes accepts ACK yes yes yes no
Table 1: Properties of the different SIP server types Table 1: Properties of the different SIP server types
1.4 Summary of SIP Operation 1.4 Summary of SIP Operation
This section explains the basic protocol functionality and operation. This section explains the basic protocol functionality and operation.
Callers and callees are identified by SIP addresses, described in Callers and callees are identified by SIP addresses, described in
Section 1.4.1. When making a SIP call, a caller first locates the Section 1.4.1. When making a SIP call, a caller first locates the
appropriate server (Section 1.4.2) and then sends a SIP request appropriate server (Section 1.4.2) and then sends a SIP request
(Section 1.4.3). The most common SIP operation is the invitation (Section 1.4.3). The most common SIP operation is the invitation
(Section 1.4.4). Instead of directly reaching the intended callee, a (Section 1.4.4). Instead of directly reaching the intended callee, a
SIP request may be redirected or may trigger a chain of new SIP SIP request may be redirected or may trigger a chain of new SIP
requests by proxies (Section 1.4.5). Users can register their requests by proxies (Section 1.4.5). Users can register their
location(s) with SIP servers (Section 4.2.6). location(s) with SIP servers (Section 4.2.6).
1.4.1 SIP Addressing 1.4.1 SIP Addressing
SIP addresses contain a user and host part. The user part is a user The "objects" addressed by SIP are users at hosts, identified by a
name, a civil name or a telephone number. The host part is either a SIP URL. The SIP URL takes the form similar to a mailto or telnet
domain name having a DNS SRV (RFC 2052 [12]), MX (RFC 974 [13], CNAME URL, i.e., user@host The user part is a user name, a civil name or a
or A record (RFC 1035 [14]), or a numeric network address. telephone number. The host part is either a domain name having a DNS
SRV (RFC 2052 [9]), MX (RFC 974 [10], CNAME or A record (RFC 1035
[11]), or a numeric network address.
A user's SIP address can be obtained out-of-band, can be learned via A user's SIP address can be obtained out-of-band, can be learned via
existing media agents, can be included in some mailers' message existing media agents, can be included in some mailers' message
headers, or can be recorded during previous invitation interactions. headers, or can be recorded during previous invitation interactions.
In many cases, the SIP address can be the same as a user's electronic In many cases, a user's SIP URL can be guessed from his email
mail address, but this is not required. SIP can thus leverage off the address.
domain name system (DNS) to provide a first-stage location
mechanisms.
Examples include: Examples of SIP URLs include:
mjh@metro.isi.edu sip:mjh@metro.isi.edu
watson@bell-telephone.com sip:watson@bell-telephone.com
root@[193.175.132.42] sip:root@193.175.132.42
root@193.175.132.42 sip:info@ietf.org
An address can designate an individual (possibly located at one of A SIP URL address can designate an individual (possibly located at
several end systems), the first available person from a group of one of several end systems), the first available person from a group
individuals or a whole group. The form of the address, e.g., of individuals or a whole group. The form of the address, e.g.,
sales@example.com , is not sufficient, in general, to determine the sip:sales@example.com , is not sufficient, in general, to determine
intent of the caller. the intent of the caller.
If a user or service chooses to be reachable at an address that is If a user or service chooses to be reachable at an address that is
guessable from the person's name and organizational affiliation, the guessable from the person's name and organizational affiliation, the
traditional method of ensuring privacy by having an unlisted "phone" traditional method of ensuring privacy by having an unlisted "phone"
number is compromised. However, unlike traditional telephony, SIP number is compromised. However, unlike traditional telephony, SIP
offers authentication and access control mechanisms and can avail offers authentication and access control mechanisms and can avail
itself of lower-layer security mechanisms, so that client software itself of lower-layer security mechanisms, so that client software
can reject unauthorized or undesired call attempts. can reject unauthorized or undesired call attempts.
Since SIP requests and responses may also contain non-SIP addresses,
e.g., telephone numbers, SIP addresses are written as SIP URLs
(Section 2) when used within SIP headers. For example,
sip:info@ietf.org
1.4.2 Locating a SIP Server 1.4.2 Locating a SIP Server
A SIP client MUST follow the following steps to resolve the host part A SIP client MUST follow the following steps to resolve the host part
of a callee address. If a client supports only TCP or UDP, but not of a callee address. If a client supports only TCP or UDP, but not
both, the client omits the respective address type. If the SIP both, the client omits the respective address type. If the SIP
address contains a port number, that number is to be used, otherwise, address contains a port number, that number is to be used, otherwise,
the default port number 5060 is to be used. The default port number the default port number 5060 is to be used. The default port number
is the same for UDP and TCP. In all cases, the client first attempts is the same for UDP and TCP. In all cases, the client first attempts
to contact the server using UDP, then TCP. to contact the server using UDP, then TCP.
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address. (For socket-based programs: For TCP, connect() returns address. (For socket-based programs: For TCP, connect() returns
ECONNREFUSED if there is no server at the designated address; for ECONNREFUSED if there is no server at the designated address; for
UDP, the socket should be bound to the destination address using UDP, the socket should be bound to the destination address using
connect() rather than sendto() or similar so that a second write() connect() rather than sendto() or similar so that a second write()
fails with ECONNREFUSED. ) fails with ECONNREFUSED. )
If the SIP address contains a numeric IP address, the client contacts If the SIP address contains a numeric IP address, the client contacts
the SIP server at that address. Otherwise, the client follows the the SIP server at that address. Otherwise, the client follows the
steps below. steps below.
1. If there is a SRV DNS resource record (RFC 2052 [12]) of 1. If there is a SRV DNS resource record (RFC 2052 [9]) of
type sip.udp, contact the listed SIP servers in the order type sip.udp, contact the listed SIP servers in the order
of the preference values contained in those resource of the preference values contained in those resource
records, using UDP as a transport protocol at the port records, using UDP as a transport protocol at the port
number given in the URL or, if none provided, the one number given in the URL or, if none provided, the one
listed in the DNS resource record. listed in the DNS resource record.
2. If there is a SRV DNS resource record (RFC 2052 [12]) of 2. If there is a SRV DNS resource record (RFC 2052 [9]) of
type sip.tcp, contact the listed SIP servers in the order type sip.tcp, contact the listed SIP servers in the order
of the preference value contained in those resource of the preference value contained in those resource
records, using TCP as a transport protocol at the port records, using TCP as a transport protocol at the port
number given in the URL or, if none provided, the one number given in the URL or, if none provided, the one
listed in the DNS resource record. listed in the DNS resource record.
3. If there is a DNS MX record (RFC 974 [13]), contact the 3. If there is a DNS MX record (RFC 974 [10]), contact the
hosts listed in their order of preference at the port hosts listed in their order of preference at the port
number listed in the URL or the default SIP port number if number listed in the URL or the default SIP port number if
none. For each host listed, first try to contact the SIP none. For each host listed, first try to contact the SIP
server using UDP, then TCP. server using UDP, then TCP.
4. Finally, check if there is a DNS CNAME or A record for the 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 given host and try to contact a SIP server at the one or
more addresses listed, again trying first UDP, then TCP. more addresses listed, again trying first UDP, then TCP.
If all of the above methods fail to locate a server, the caller MAY If all of the above methods fail to locate a server, the caller MAY
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a last resort, a client MAY choose to deliver the session description a last resort, a client MAY choose to deliver the session description
to the callee using electronic mail. to the callee using electronic mail.
A client MAY cache the result of the reachability steps for a A client MAY cache the result of the reachability steps for a
particular address and retry that host address for the next call. If particular address and retry that host address for the next call. If
the client does not find a SIP server at the cached address, it MUST the client does not find a SIP server at the cached address, it MUST
start the search at the beginning of the sequence. start the search at the beginning of the sequence.
This sequence is modeled after that described for SMTP, This sequence is modeled after that described for SMTP,
where MX records are to be checked before A records (RFC where MX records are to be checked before A records (RFC
1123 [15]). 1123 [12]).
1.4.3 SIP Transaction 1.4.3 SIP Transaction
Once the host part has been resolved to a SIP server, the client Once the host part has been resolved to a SIP server, the client
sends one or more SIP requests to that server and receives one or sends one or more SIP requests to that server and receives one or
more responses from the server. A request (and its retransmissions) more responses from the server. A request (and its retransmissions)
together with the responses triggered by that request make up a SIP together with the responses triggered by that request make up a SIP
transaction. The ACK request following an INVITE is not part of transaction. The ACK request following an INVITE is not part of
the transaction since it may traverse a different set of hosts. the transaction since it may traverse a different set of hosts.
skipping to change at page 10, line 31 skipping to change at page 10, line 34
A successful SIP invitation consists of two requests, INVITE A successful SIP invitation consists of two requests, INVITE
followed by ACK. The INVITE (Section 4.2.1) request asks the callee followed by ACK. The INVITE (Section 4.2.1) request asks the callee
to join a particular conference or establish a two-party to join a particular conference or establish a two-party
conversation. After the callee has agreed to participate in the call, conversation. After the callee has agreed to participate in the call,
the caller confirms that it has received that response by sending an the caller confirms that it has received that response by sending an
ACK (Section 4.2.2) request. If the caller no longer wants to ACK (Section 4.2.2) request. If the caller no longer wants to
participate in the call, it sends a BYE request instead of an ACK. participate in the call, it sends a BYE request instead of an ACK.
The INVITE request typically contains a session description, for The INVITE request typically contains a session description, for
example written in SDP (RFC 2327, [7]) format, that provides the example written in SDP (RFC 2327 [5]) format, that provides the
called party with enough information to join the session. For called party with enough information to join the session. For
multicast sessions, the session description enumerates the media multicast sessions, the session description enumerates the media
types and formats that may be distributed to that session. For a types and formats that may be distributed to that session. For a
unicast session, the session description enumerates the media types unicast session, the session description enumerates the media types
and formats that the caller is willing to receive and where it wishes and formats that the caller is willing to receive and where it wishes
the media data to be sent. In either case, if the callee wishes to the media data to be sent. In either case, if the callee wishes to
accept the call, it responds to the invitation by returning a similar accept the call, it responds to the invitation by returning a similar
description listing the media it wishes to receive. For a multicast description listing the media it wishes to receive. For a multicast
session, the callee should only return a session description if it is session, the callee should only return a session description if it is
unable to receive the media indicated in the caller's description or unable to receive the media indicated in the caller's description or
wants to receive data via unicast. wants to receive data via unicast.
The protocol exchanges for the INVITE method are shown in Fig. 1 for The protocol exchanges for the INVITE method are shown in Fig. 1 for
a proxy server and in Fig. 2 for a redirect server. In Fig. 1, the a proxy server and in Fig. 2 for a redirect server. (Note that the
proxy server accepts the INVITE request (step 1), contacts the messages shown in the figures have been abbreviated slightly.) In
location service with all or parts of the address (step 2) and Fig. 1, the proxy server accepts the INVITE request (step 1),
obtains a more precise location (step 3). The proxy server then contacts the location service with all or parts of the address (step
issues a SIP INVITE request to the address(es) returned by the 2) and obtains a more precise location (step 3). The proxy server
then issues a SIP INVITE request to the address(es) returned by the
location service (step 4). The user agent server alerts the user location service (step 4). The user agent server alerts the user
(step 5) and returns a success indication to the proxy server (step (step 5) and returns a success indication to the proxy server (step
6). The proxy server then returns the success result to the original 6). The proxy server then returns the success result to the original
caller (step 7). The receipt of this message is confirmed by the caller (step 7). The receipt of this message is confirmed by the
caller using an ACK request, which is forwarded to the callee (steps caller using an ACK request, which is forwarded to the callee (steps
8 and 9). All requests and responses have the same Call-ID. 8 and 9). All requests and responses have the same Call-ID.
+....... cs.columbia.edu .......+ +....... cs.columbia.edu .......+
: : : :
: (~~~~~~~~~~) : : (~~~~~~~~~~) :
skipping to change at page 12, line 4 skipping to change at page 12, line 8
The redirect server shown in Fig. 2 accepts the INVITE request (step The redirect server shown in Fig. 2 accepts the INVITE request (step
1), contacts the location service as before (steps 2 and 3) and, 1), contacts the location service as before (steps 2 and 3) and,
instead of contacting the newly found address itself, returns the instead of contacting the newly found address itself, returns the
address to the caller (step 4), which is then acknowledged via an address to the caller (step 4), which is then acknowledged via an
ACK request (step 5). The caller issues a new request, with the same ACK request (step 5). The caller issues a new request, with the same
call-ID but a higher CSeq, to the address returned by the first call-ID but a higher CSeq, to the address returned by the first
server (step 6). In the example, the call succeeds (step 7). The server (step 6). In the example, the call succeeds (step 7). The
caller and callee complete the handshake with an ACK (step 8). caller and callee complete the handshake with an ACK (step 8).
The next section discusses what happens if the location service The next section discusses what happens if the location service
returns more than one possible alternative.
1.4.5 Locating a User
A callee may move between a number of different end systems over
time. These locations can be dynamically registered with the SIP
server (Sections 1.4.7, 4.2.6). A location server may also use one or
more other protocols, such as finger (RFC 1288 [13]), rwhois (RFC
2167 [14]), LDAP (RFC 1777 [15]), multicast-based protocols [16] or
operating-system dependent mechanisms to actively determine the end
system where a user might be reachable. A location server may return
several locations because the user is logged in at several hosts
simultaneously or because the location server has (temporarily)
inaccurate information. The SIP server combines the results to yield
a list of a zero or more locations. It is recommended that each
location server sorts results according to the likelihood of success.
The action taken on receiving a list of locations varies with the
type of SIP server. A SIP redirect server returns the list to the
client as Location headers (Section 6.22). A SIP proxy server can
sequentially or in parallel try the addresses until the call is
successful (2xx response) or the callee has declined the call (6xx
response). With sequential attempts, a proxy server can implement an
"anycast" service.
If a proxy server forwards a SIP request, it MUST add itself to the
end of the list of forwarders noted in the Via (Section 6.40)
headers. The Via trace ensures that replies can take the same path
back, ensuring correct operation through compliant firewalls and
avoiding request loops. On the response path, each host MUST remove
its Via, so that routing internal information is hidden from the
callee and outside networks. When a multicast request is made, first
the host making the request, then the multicast address itself are
added to the path. A proxy server MUST check that it does not
generate a request to a host listed in the Via list. (Note: If a
host has several names or network addresses, this may not always
work. Thus, each host also checks if it is part of the Via list.)
A SIP invitation may traverse more than one SIP proxy server. If one
of these "forks" the request, i.e., issues more than one request in
response to receiving the invitation request, it is possible that a
client is reached, independently, by more than one copy of the
invitation request. Each of these copies bears the same Call-ID. The
+....... cs.columbia.edu .......+ +....... cs.columbia.edu .......+
: : : :
: (~~~~~~~~~~) : : (~~~~~~~~~~) :
: ( location ) : : ( location ) :
: ( service ) : : ( service ) :
: (~~~~~~~~~~) : : (~~~~~~~~~~) :
: ^ | : : ^ | :
: | hgs@play : : | hgs@play :
: 2| 3| : : 2| 3| :
: | | : : | | :
skipping to change at page 13, line 4 skipping to change at page 14, line 4
| : ( ) : | : ( ) :
| 8: ACK : ( ) : | 8: ACK : ( ) :
======================================================> (~~~~~~) : ======================================================> (~~~~~~) :
+...............................+ +...............................+
====> SIP request ====> SIP request
....> SIP response ....> SIP response
----> non-SIP protocols ----> non-SIP protocols
Figure 2: Example of SIP redirect server Figure 2: Example of SIP redirect server
returns more than one possible alternative.
1.4.5 Locating a User
A callee may move between a number of different end systems over
time. These locations can be dynamically registered with the SIP
server (Sections 1.4.7, 4.2.6). A location server may also use one or
more other protocols, such as finger (RFC 1288 [16]), rwhois (RFC
2167 [17]), LDAP (RFC 1777 [18]), multicast-based protocols or
operating-system dependent mechanism to actively determine the end
system where a user might be reachable. A location server may return
several locations because the user is logged in at several hosts
simultaneously or because the location server has (temporarily)
inaccurate information. The SIP server combines the results to yield
a list of a zero or more locations. It is recommended that each
location server sorts results according to the likelihood of success.
The action taken on receiving a list of locations varies with the
type of SIP server. A SIP redirect server returns the list to the
client sending the request as Location headers (Section 6.22). A SIP
proxy server can sequentially or in parallel try the addresses until
the call is successful (2xx response) or the callee has declined the
call (6xx response). With sequential attempts, a proxy server can
implement an "anycast" service.
If a proxy server forwards a SIP request, it MUST add itself to the
end of the list of forwarders noted in the Via (Section 6.40)
headers. The Via trace ensures that replies can take the same path
back, ensuring correct operation through compliant firewalls and
avoiding request loops. On the response path, each host MUST remove
its Via, so that routing internal information is hidden from the
callee and outside networks. When a multicast request is made, first
the host making the request, then the multicast address itself are
added to the path. A proxy server MUST check that it does not
generate a request to a host listed in the Via list. (Note: If a
host has several names or network addresses, this may not always
work. Thus, each host also checks if it is part of the Via list.)
A SIP invitation may traverse more than one SIP proxy server. If one
of these "forks" the request, i.e., issues more than one request in
response to receiving the invitation request, it is possible that a
client is reached, independently, by more than one copy of the
invitation request. Each of these copies bears the same Call-ID. The
user agent MUST return the appropriate status response. Duplicate user agent MUST return the appropriate status response. Duplicate
requests are not an error, so there is no need to alert the user. requests are not an error.
1.4.6 Changing an Existing Session 1.4.6 Changing an Existing Session
In some circumstances, it may be necessary to change the parameters In some circumstances, it may be necessary to change the parameters
of an existing session. For example, two parties may have been of an existing session. For example, two parties may have been
conversing and then want to add a third party, switching to multicast conversing and then want to add a third party, switching to multicast
for efficiency. One of the participants invites the third party with for efficiency. One of the participants invites the third party with
the new multicast address and simultaneously sends an INVITE to the the new multicast address and simultaneously sends an INVITE to the
second party, with the new multicast session description, but with second party, with the new multicast session description, but with
the old call identifier. the old call identifier.
1.4.7 Registration Services 1.4.7 Registration Services
skipping to change at page 14, line 15 skipping to change at page 14, line 20
of an existing session. For example, two parties may have been of an existing session. For example, two parties may have been
conversing and then want to add a third party, switching to multicast conversing and then want to add a third party, switching to multicast
for efficiency. One of the participants invites the third party with for efficiency. One of the participants invites the third party with
the new multicast address and simultaneously sends an INVITE to the the new multicast address and simultaneously sends an INVITE to the
second party, with the new multicast session description, but with second party, with the new multicast session description, but with
the old call identifier. the old call identifier.
1.4.7 Registration Services 1.4.7 Registration Services
The REGISTER request allows a client to let a proxy or redirect The REGISTER request allows a client to let a proxy or redirect
server know which address(es) it may be reached under. A client may server know at which address(es) it may be reached. A client may also
also use it to install call handling features at the server. use it to install call handling features at the server.
1.5 Protocol Properties 1.5 Protocol Properties
1.5.1 Minimal State 1.5.1 Minimal State
A single conference session or call may involve one or more SIP A single conference session or call may involve one or more SIP
request-response transactions. Proxy servers do not have to keep request-response transactions. Proxy servers do not have to keep
state for a particular call, however, they MAY maintain state for a state for a particular call, however, they MAY maintain state for a
single SIP transaction, as discussed in Section 11. single SIP transaction, as discussed in Section 11.
skipping to change at page 14, line 44 skipping to change at page 14, line 49
or a byte stream service, with reliable or unreliable service. or a byte stream service, with reliable or unreliable service.
In an Internet context, SIP is able to utilize both UDP and TCP as In an Internet context, SIP is able to utilize both UDP and TCP as
transport protocols, among others. UDP allows the application to more transport protocols, among others. UDP allows the application to more
carefully control the timing of messages and their retransmission, to carefully control the timing of messages and their retransmission, to
perform parallel searches without requiring TCP connection state for perform parallel searches without requiring TCP connection state for
each outstanding request, and to use multicast. Routers can more each outstanding request, and to use multicast. Routers can more
readily snoop SIP UDP packets. TCP allows easier passage through readily snoop SIP UDP packets. TCP allows easier passage through
existing firewalls, and given the similar protocol design, allows existing firewalls, and given the similar protocol design, allows
common servers for SIP, HTTP and the Real Time Streaming Protocol common servers for SIP, HTTP and the Real Time Streaming Protocol
(RTSP) [1]. (RTSP) (RFC 2326 [4]).
When TCP is used, SIP can use one or more connections to attempt to When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing conference. contact a user or to modify parameters of an existing conference.
Different SIP requests for the same SIP call may use different TCP Different SIP requests for the same SIP call may use different TCP
connections or a single persistent connection, as appropriate. connections or a single persistent connection, as appropriate.
User agents SHOULD implement both UDP and TCP transport, proxy and
redirect servers MUST.
For concreteness, this document will only refer to Internet For concreteness, this document will only refer to Internet
protocols. However, SIP may also be used directly with protocols protocols. However, SIP may also be used directly with protocols
such as ATM AAL5, IPX, frame relay or X.25. The necessary naming such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
conventions are beyond the scope of this document. conventions are beyond the scope of this document. User agents SHOULD
implement both UDP and TCP transport, proxy and redirect servers
MUST.
1.5.3 Text-Based 1.5.3 Text-Based
SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This
allows easy implementation in languages such as Java, Tcl and Perl, allows easy implementation in languages such as Java, Tcl and Perl,
allows easy debugging, and most importantly, makes SIP flexible and allows easy debugging, and most importantly, makes SIP flexible and
extensible. As SIP is used for initiating multimedia conferences extensible. As SIP is used for initiating multimedia conferences
rather than delivering media data, it is believed that the additional rather than delivering media data, it is believed that the additional
overhead of using a text-based protocol is not significant. overhead of using a text-based protocol is not significant.
2 SIP Uniform Resource Locators 2 SIP Uniform Resource Locators
SIP URLs are used within SIP messages to indicate the originator, SIP URLs are used within SIP messages to indicate the originator (
current destination and final recipient of a SIP request, and to From), current destination ( Request-URI) and final recipient ( To)
specify redirection addresses. A SIP URL can also be embedded in web of a SIP request, and to specify redirection addresses ( Location). A
pages or other hyperlinks to indicate that a user or service may be SIP URL can also be embedded in web pages or other hyperlinks to
called. indicate that a user or service may be called.
Because interaction with some resources may require message headers Because interaction with some resources may require message headers
or message bodies to be specified as well as the SIP address, the SIP or message bodies to be specified as well as the SIP address, the SIP
URL scheme is defined to allow setting SIP request-header fields and URL scheme is defined to allow setting SIP request-header fields and
the SIP message-body. (This is similar to the mailto: URL [19].) the SIP message-body.
A SIP URL follows the guidelines of RFC 1630, as revised, [20,21] and A SIP URL follows the guidelines of RFC 1630 [17], as revised [18],
takes the following form: and has the syntax shown in Fig. 3. Note that reserved characters
have to be escaped.
SIP-URL = "sip:" [ userinfo "@" ] hostport The URI character classes referenced above are described in Section
C. The URI specification is currently being revised. It is
anticipated that future versions of this specification will reference
the revised edition. Note that all URL reserved characters MUST be
encoded.
host: The mailto: URL and RFC 822 email addresses require that
numeric host addresses ("host numbers") are enclosed in square
brackets (presumably, since host names might be numeric), while
SIP-URL = "sip:" [ userinfo ] "@" hostport
url-parameters [ headers ] url-parameters [ headers ]
userinfo = user [ ":" password ]
user = *( unreserved | escaped
| ";" | "&" | "=" | "+" | "$" | "," )
password = *( unreserved | escaped
| ";" | "&" | "=" | "+" | "$" | "," )
hostport = host [ ":" port ]
host = hostname | IPv4address
hostname = *( domainlabel "." ) toplabel [ "." ]
domainlabel = alphanum | alphanum *( alphanum | "-" ) alphanum
toplabel = alpha | alpha *( alphanum | "-" ) alphanum
IPv4address = 1*digit "." 1*digit "." 1*digit "." 1*digit
port = *digit
url-parameters = *( ";" url-parameter ) url-parameters = *( ";" url-parameter )
url-parameter = transport-param | user-param url-parameter = transport-param | user-param
| ttl-param | maddr-param | tag-param | other-param | ttl-param | maddr-param | tag-param | other-param
transport-param = "transport=" ( "udp" | "tcp" ) transport-param = "transport=" ( "udp" | "tcp" )
ttl-param = "ttl=" ttl ttl-param = "ttl=" ttl
ttl = 1*3DIGIT ; 0 to 255 ttl = 1*3DIGIT ; 0 to 255
maddr-param = "maddr=" maddr maddr-param = "maddr=" maddr
maddr = IPv4address ; multicast address maddr = IPv4address ; multicast address
user-param = "user=" ( "phone" ) user-param = "user=" ( "phone" )
tag-param = "tag=" UUID tag-param = "tag=" UUID
UUID = 1*( hex | "-" )
other-param = *uric other-param = *uric
headers = "?" header *( "&" header ) headers = "?" header *( "&" header )
header = hname "=" hvalue header = hname "=" hvalue
hname = *uric hname = *uric
hvalue = *uric hvalue = *uric
uric = reserved | unreserved | escaped
reserved = ";" | "/" | "?" | ":" | "@" | "&" | "=" | "+" |
"$" | ","
digits = 1*DIGIT digits = 1*DIGIT
Note that all URL reserved characters MUST be encoded. The special
hname "body" indicates that the associated hvalue is the message-
body of the SIP INVITE request. Within sip URLs, the characters
"?", "=", "&" are reserved.
The mailto: URL and RFC 822 email addresses require that numeric Figure 3: SIP URL syntax
host addresses ("host numbers") are enclosed in square brackets
(presumably, since host names might be numeric), while host numbers
without brackets are used for all other URLs. The SIP URL requires
the latter form.
The elements userinfo, uric, hostport, IPv4address are defined in host numbers without brackets are used for all other URLs. The
[21]. SIP URL requires the latter form, without brackets.
The SIP scheme MAY use the format "user:password" in the userinfo userinfo: The SIP scheme MAY use the format " user:password" in the
field. The use of passwords in the userinfo is NOT RECOMMENDED, userinfo field. The use of passwords in the userinfo is NOT
because the passing of authentication information in clear text (such RECOMMENDED, because the passing of authentication information
as URI) has proven to be a security risk in almost every case where in clear text (such as URIs) has proven to be a security risk in
it has been used. almost every case where it has been used.
telephone-subscriber = global-phone-number | local-phone-number
global-phone-number = "+" 1*phonedigit [isdn-subaddress]
[post-dial]
local-phone-number = 1*(phonedigit | dtmf-digit |
pause-character) [isdn-subaddress]
[post-dial]
isdn-subaddress = ";isub=" 1*phonedigit
post-dial = ";postd=" 1*(phonedigit | dtmf-digit
| pause-character)
phonedigit = DIGIT | visual-separator
visual-separator = "-" | "."
pause-character = one-second-pause | wait-for-dial-tone
one-second-pause = "p"
wait-for-dial-tone = "w"
dtmf-digit = "*" | "#" | "A" | "B" | "C" | "D"
Figure 4: SIP URL syntax; telephone subscriber
If the host is an Internet telephony gateway, the userinfo field can If the host is an Internet telephony gateway, the userinfo field can
also encode a telephone number using the notation of telephone- also encode a telephone number using the notation of telephone-
subscriber defined in [22]. The telephone number is a special case of subscriber (Fig. 4). The telephone number is a special case of a
a user name and cannot be distinguished by a BNF. Thus, a URL user name and cannot be distinguished by a BNF. Thus, a URL
parameter, user, is added to distinguish telephone numbers from user parameter, user, is added to distinguish telephone numbers from user
names. The phone identifier is to be used when connecting to a names. The phone identifier is to be used when connecting to a
telephony gateway. Even without this parameter, recipients of SIP telephony gateway. Even without this parameter, recipients of SIP
URLs MAY interpret the pre-@ part as a phone number if local URLs MAY interpret the pre-@ part as a phone number if local
restrictions on the name space for user name allow to make this restrictions on the name space for user name allow it.
determination.
The tag parameter allows to distinguish several instances of a user
that share the same host and port values, for example, where these
designate a firewall. The tag value is a version-1 (time-based) or
version-4 (random) UUID [31]. The tag value is designed to be
globally unique within each Call-ID and only to be used within the
same Call-ID. It SHOULD NOT be included in long-lived SIP URLs,
e.g., those found on web pages or user databases. A single user
maintains the same tag throughout the call identified by the Call-ID.
If a server handles SIP addresses for another domain, it MUST URL- If a server handles SIP addresses for another domain, it MUST URL-
encode the "@" character (%40). encode the "@" character (%40).
SIP URLs can define specific parameters of the request, including the URL parameters: SIP URLs can define specific parameters of the
transport mechanism (UDP or TCP) and the use of multicast to make a request, including the transport mechanism (UDP or TCP) and the
request. These parameters are added after the host and are separated use of multicast to make a request. These parameters are added
by semi-colons. For example, to specify to call j.doe@big.com using after the host and are separated by semi-colons. For example, to
multicast to 239.255.255.1 with a ttl of 15, the following URL would specify to call j.doe@big.com using multicast to 239.255.255.1
be used: 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 The transport protocol UDP is to be assumed when a multicast address
is given. is given.
Transport parameters MUST NOT be used in the From and To header
fields and the Request-URI; they are ignored if present.
Headers: Headers of the SIP request can be defined with the "?"
mechanism within a SIP URL. The special hname " body" indicates
that the associated hvalue is the message-body of the SIP
INVITE request. Headers MUST NOT be used in the From and To
header fields and the Request-URI; they are ignored if present.
Tag: The tag parameter allows several instances of a user that share
the same host and port values to be distinguished from each
other, for example, where the host designates a firewall or
proxy. The tag value is a random string consisting of hex
digits. The use of version-1 (time-based) or version-4 (random)
UUID [19] is OPTIONAL. The tag value is designed to be
globally unique and cryptographically random with at least 32
bits of randomness. It SHOULD NOT be included in long-lived SIP
URLs, e.g., those found on web pages or user databases. A single
user maintains the same tag throughout the call identified by
the Call-ID. The tag parameter in To headers is ignored when
matching responses to requests that did not contain a tag in
their To header. (See Section 6.37.)
Table 2 summarizes where the components of the SIP URL can be used.
Request-URI To From Location external
user x x x x x
password x x x
host x x x x x
tag x x x x
headers x x
transport para. x x
Table 2: Use of URL elements for SIP headers, Request-URI and
references
Examples of SIP URLs are: Examples of SIP URLs are:
sip:j.doe@big.com sip:j.doe@big.com
sip:j.doe:secret@big.com;transport=tcp sip:j.doe:secret@big.com;transport=tcp
sip:j.doe@big.com?subject=project sip:j.doe@big.com?subject=project
sip:+1-212-555-1212:1234@gateway.com;user=phone sip:+1-212-555-1212:1234@gateway.com;user=phone
sip:1212@gateway.com sip:1212@gateway.com
sip:alice@10.1.2.3 sip:alice@10.1.2.3
sip:alice@example.com;tag=f81d4fae-7dec-11d0-a765-00a0c91e6bf6 sip:alice@example.com;tag=f81d4fae-7dec-11d0-a765-00a0c91e6bf6
sip:alice sip:alice
Within a SIP message, URLs are used to indicate the source and Within a SIP message, URLs are used to indicate the source and
intended destination of a request, redirection addresses and the intended destination of a request, redirection addresses and the
current destination of a request. Normally all these fields will current destination of a request. Normally all these fields will
contain SIP URLs. contain SIP URLs.
SIP URLs are case-insensitive, so that sip:j.doe@example.com and SIP URLs are case-insensitive, so that for example the two URLs
SIP:J.Doe@Example.com are equivalent. All URL parameters are included sip:j.doe@example.com and SIP:J.Doe@Example.com are equivalent. All
when comparing SIP URLs for equality. URL parameters are included when comparing SIP URLs for equality.
In some circumstances a non-SIP URL may be used in a SIP message. An In some circumstances a non-SIP URL may be used in a SIP message. An
example might be making a call from a telephone which is relayed by a example might be making a call from a telephone which is relayed by a
gateway onto the internet as a SIP request. In such a case, the gateway onto the internet as a SIP request. In such a case, the
source of the call is really the telephone number of the caller, and source of the call is really the telephone number of the caller, and
so a SIP URL is inappropriate and a phone URL might be used instead. so a SIP URL is inappropriate and a phone URL might be used instead.
To allow for this flexibility, SIP headers that specify user To allow for this flexibility, SIP headers that specify user
addresses allow these addresses to be SIP and non-SIP URLs. addresses allow these addresses to be SIP and non-SIP URLs.
Clearly not all URLs are appropriate to be used in a SIP message as a Clearly not all URLs are appropriate to be used in a SIP message as a
user address. The correct behavior when an unknown scheme is user address. The correct behavior when an unknown scheme is
encountered by a SIP server is defined in the context of each of the encountered by a SIP server is defined in the context of each of the
header fields that use a SIP URL. header fields that use a SIP URL.
3 SIP Message Overview 3 SIP Message Overview
SIP is a text-based protocol and uses the ISO 10646 character set in SIP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [23]). Lines are terminated by CRLF, but UTF-8 encoding (RFC 2279 [20]). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by receivers should be prepared to also interpret CR and LF by
themselves as line terminators. themselves as line terminators.
Except for the above difference in character sets, much of the Except for the above difference in character sets, much of the
message syntax is identical to HTTP/1.1, rather than repeating it 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 here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1
specification (RFC 2068 [10]). In addition, we describe SIP in both specification (RFC 2068 [7]). In addition, we describe SIP in both
prose and an augmented Backus-Naur form (BNF) [H2.1] described in prose and an augmented Backus-Naur form (BNF) [H2.1] described in
detail in RFC 2234 [24]. detail in RFC 2234 [21].
Unlike HTTP, SIP MAY use UDP. When sent over TCP or UDP, multiple SIP
transactions can be carried in a single TCP connection or UDP transactions can be carried in a single TCP connection or UDP
datagram. UDP datagrams, including all headers, should not normally datagram. UDP datagrams, including all headers, should not normally
be larger than the path maximum transmission unit (MTU) if the MTU is be larger than the path maximum transmission unit (MTU) if the MTU is
known, or 1400 bytes if the MTU is unknown. known, or 1400 bytes if the MTU is unknown.
The 1400 bytes accommodates lower-layer packet headers The 1400 bytes accommodates lower-layer packet headers
within the "typical" MTU of around 1500 bytes. Recent within the "typical" MTU of around 1500 bytes. Recent
studies [25] indicate that an MTU of 1500 bytes is a studies [22] indicate that an MTU of 1500 bytes is a
reasonable assumption. The next lower common MTU values are reasonable assumption. The next lower common MTU values are
1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191 1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
[26]). Thus, another reasonable value would be a message [23]). Thus, another reasonable value would be a message
size of 950 bytes, to accommodate packet headers within the size of 950 bytes, to accommodate packet headers within the
SLIP MTU without fragmentation. SLIP MTU without fragmentation.
A SIP message is either a request from a client to a server, or a A SIP message is either a request from a client to a server, or a
response from a server to a client. response from a server to a client.
SIP-message ___ Request | Response SIP-message ___ Request | Response
Both Request (section 4) and Response (section 5) messages use the Both Request (section 4) and Response (section 5) messages use the
generic-message format of RFC 822 [27] for transferring entities (the generic-message format of RFC 822 [24] for transferring entities (the
body of the message). Both types of message consist of a start-line, body of the message). Both types of messages consist of a start-
one or more header fields (also known as "headers"), an empty line line, one or more header fields (also known as "headers"), an empty
(i.e., a line with nothing preceding the carriage-return line-feed ( line (i.e., a line with nothing preceding the carriage-return line-
CRLF)) indicating the end of the header fields, and an optional feed ( CRLF)) indicating the end of the header fields, and an
message-body. To avoid confusion with similar-named headers in HTTP, optional message-body. To avoid confusion with similar-named headers
we refer to the header describing the message body as entity headers. in HTTP, we refer to the header describing the message body as entity
These components are described in detail in the upcoming sections. headers. These components are described in detail in the upcoming
sections.
generic-message = start-line generic-message = start-line
*message-header *message-header
CRLF CRLF
[ message-body ] [ message-body ]
start-line = Request-Line | Section 4.1 start-line = Request-Line | Section 4.1
Status-Line Section 5.1 Status-Line Section 5.1
message-header = *( general-header message-header = *( general-header
| request-header | request-header
| response-header | response-header
| entity-header ) | entity-header )
In the interest of robustness, any leading empty line(s) MUST be
general-header = Call-ID ; Section 6.12 general-header = Call-ID ; Section 6.12
| CSeq ; Section 6.16 | CSeq ; Section 6.16
| Date ; Section 6.17 | Date ; Section 6.17
| Encryption ; Section 6.18 | Encryption ; Section 6.18
| Expires ; Section 6.19 | Expires ; Section 6.19
| From ; Section 6.20 | From ; Section 6.20
| Record-Route ; Section 6.30 | Record-Route ; Section 6.29
| Timestamp ; Section 6.36 | Timestamp ; Section 6.36
| To ; Section 6.37 | To ; Section 6.37
| Via ; Section 6.40 | Via ; Section 6.40
entity-header = Content-Encoding ; Section 6.13 entity-header = Content-Encoding ; Section 6.13
| Content-Length ; Section 6.14 | Content-Length ; Section 6.14
| Content-Type ; Section 6.15 | Content-Type ; Section 6.15
request-header = Accept ; Section 6.7 request-header = Accept ; Section 6.7
| Accept-Encoding ; Section 6.8 | Accept-Encoding ; Section 6.8
| Accept-Language ; Section 6.9 | Accept-Language ; Section 6.9
| Authorization ; Section 6.11 | Authorization ; Section 6.11
| Hide ; Section 6.21 | Hide ; Section 6.21
| Location ; Section 6.22 | Location ; Section 6.22
| Max-Forwards ; Section 6.23 | Max-Forwards ; Section 6.23
| Organization ; Section 6.24 | Organization ; Section 6.24
| Priority ; Section 6.25 | Priority ; Section 6.25
| Proxy-Authorization ; Section 6.27 | Proxy-Authorization ; Section 6.27
| Proxy-Require ; Section 6.28 | Proxy-Require ; Section 6.28
| Route ; Section 6.32 | Route ; Section 6.33
| Require ; Section 6.29 | Require ; Section 6.30
| Response-Key ; Section 6.31 | Response-Key ; Section 6.31
| Subject ; Section 6.35 | Subject ; Section 6.35
| User-Agent ; Section 6.39 | User-Agent ; Section 6.39
response-header = Allow ; Section 6.10 response-header = Allow ; Section 6.10
| Location ; Section 6.22 | Location ; Section 6.22
| Proxy-Authenticate ; Section 6.26 | Proxy-Authenticate ; Section 6.26
| Retry-After ; Section 6.33 | Retry-After ; Section 6.32
| Server ; Section 6.34 | Server ; Section 6.34
| Unsupported ; Section 6.38 | Unsupported ; Section 6.38
| Warning ; Section 6.41 | Warning ; Section 6.41
| WWW-Authenticate ; Section 6.42 | WWW-Authenticate ; Section 6.42
Table 2: SIP headers Table 3: SIP headers
In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins ignored. In other words, if the Request or Response message begins
with a CRLF, the CRLF should be ignored. with a CRLF, CR, or LF, these characters should be ignored.
4 Request 4 Request
The Request message format is shown below: The Request message format is shown below:
Request = Request-Line ; Section 4.1 Request = Request-Line ; Section 4.1
*( general-header *( general-header
| request-header | request-header
| entity-header ) | entity-header )
CRLF CRLF
[ message-body ] ; Section 8 [ message-body ] ; Section 8
4.1 Request-Line 4.1 Request-Line
skipping to change at page 20, line 19 skipping to change at page 22, line 20
CRLF CRLF
[ message-body ] ; Section 8 [ message-body ] ; Section 8
4.1 Request-Line 4.1 Request-Line
The Request-Line begins with a method token, followed by the The Request-Line begins with a method token, followed by the
Request-URI and the protocol version, and ending with CRLF. The Request-URI and the protocol version, and ending with CRLF. The
elements are separated by SP characters. No CR or LF are allowed elements are separated by SP characters. No CR or LF are allowed
except in the final CRLF sequence. except in the final CRLF sequence.
Request-Line ___ Method SP Request-URI SP SIP-Version CRLF Request-Line = Method SP Request-URI SP SIP-Version CRLF
4.2 Methods 4.2 Methods
The methods are defined below. Methods that are not supported by a The methods are defined below. Methods that are not supported by a
proxy or redirect server are treated by that server as if they were proxy or redirect server are treated by that server as if they were
an INVITE method and forwarded accordingly. an INVITE method and forwarded accordingly. Methods that are not
supported by a user agent server cause a 501 (Not Implemented)
Methods that are not supported by a user agent server cause a 501 response to be returned (Section 7).
(Not Implemented) response to be returned (Section 7).
Method = "ACK" | "BYE" | "CANCEL" | "INVITE" Method = "ACK" | "BYE" | "CANCEL" | "INVITE"
| "OPTIONS" | "REGISTER" | "OPTIONS" | "REGISTER"
4.2.1 INVITE 4.2.1 INVITE
The INVITE method indicates that the user or service is being The INVITE method indicates that the user or service is being
invited to participate in a session. The message body contains a invited to participate in a session. The message body contains a
description of the session to which the callee is being invited. For description of the session to which the callee is being invited. For
two-party calls, the caller indicates the type of media it is able to two-party calls, the caller indicates the type of media it is able to
receive as well as their parameters such as network destination. If receive as well as their parameters such as network destination. If
the session description format allows this, it may also indicate the session description format allows this, it may also indicate
"send-only" media. A success response indicates in its message body "send-only" media. A success response indicates in its message body
which media the callee wishes to receive. which media the callee wishes to receive.
A server MAY automatically respond to an invitation for a conference A server MAY automatically respond to an invitation for a conference
the user is already participating in, identified either by the SIP the user is already participating in, identified either by the SIP
Call-ID or a globally unique identifier within the session Call-ID or a globally unique identifier within the session
description, with a 200 (OK) response. description, with a 200 (OK) response.
If a user agent receives an INVITE request for an existing Call-ID
with a higher CSeq sequence number than any previous INVITE for the
same Call-ID, it MUST check any version identifiers in the session same Call-ID, it MUST check any version identifiers in the session
description or, if there are no version identifiers, the content of description or, if there are no version identifiers, the content of
the session description to see if it has changed. It MUST also the session description to see if it has changed. It MUST also
inspect any other header fields for changes and act accordingly. If inspect any other header fields for changes and act accordingly. If
the session description has changed, the user agent server MUST the session description has changed, the user agent server MUST
adjust the session parameters accordingly, possibly after asking the adjust the session parameters accordingly, possibly after asking the
user for confirmation. (Versioning of the session description may be user for confirmation. (Versioning of the session description may be
used to accommodate the capabilities of new arrivals to a conference, used to accommodate the capabilities of new arrivals to a conference,
add or delete media or change from a unicast to a multicast add or delete media or change from a unicast to a multicast
conference.) conference.)
This method MUST be supported by a SIP server and client. This method MUST be supported by SIP proxy, redirect and user agent
servers as well as clients.
4.2.2 ACK 4.2.2 ACK
The ACK request confirms that the client has received a final The ACK request confirms that the client has received a final
response to an INVITE request. ( ACK is used only with INVITE response to an INVITE request. ( ACK is used only with INVITE
requests.) 2xx responses are acknowledged by client user agents, all requests.) 2xx responses are acknowledged by client user agents, all
other final responses by the first proxy or client user agent to other final responses by the first proxy or client user agent to
receive the response. The Via is always initialized to the host that receive the response. The Via is always initialized to the host that
originates the ACK request, i.e., the client user agent after a 2xx originates the ACK request, i.e., the client user agent after a 2xx
response or the first proxy to receive a non-2xx final response. The response or the first proxy to receive a non-2xx final response. The
ACK request is forwarded as the corresponding INVITE request, based ACK request is forwarded as the corresponding INVITE request, based
on its Request-URI. See Section 10 for details. This method MUST be on its Request-URI. See Section 10 for details.
supported by a SIP server and client.
The ACK request MAY contain a message body with the final session The ACK request MAY contain a message body with the final session
description to be used by the callee. If the ACK message body is description to be used by the callee. If the ACK message body is
empty, the callee uses the session description in the INVITE empty, the callee uses the session description in the INVITE
request. request.
This method MUST be supported by SIP proxy, redirect and user agent
servers as well as clients.
4.2.3 OPTIONS 4.2.3 OPTIONS
The client is being queried as to its capabilities. A server that 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 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 is logged in and has been recently active, MAY respond to this
request with a capability set. Support of this method is REQUIRED. request with a capability set. A called user agent MAY return a
status reflecting how it would have responded to an invitation, e.g.,
600 (Busy).
A called user agent MAY return a status reflecting how it would have This method MUST be supported by SIP proxy, redirect and user agent
responded to an invitation, e.g., 600 (Busy). servers, registrars and clients.
4.2.4 BYE 4.2.4 BYE
The user agent client uses BYE to indicate to the server that it The user agent client uses BYE to indicate to the server that it
wishes to abort the call. A BYE request is forwarded like an INVITE wishes to abort the call. A BYE request is forwarded like an INVITE
request. It terminates any on-going searches for the named call. A request. A caller SHOULD issue a BYE request before aborting a call
caller SHOULD issue a BYE request before aborting a call ("hanging ("hanging up"). Note that a BYE request may also be issued by the
up"). Note that a BYE request may also be issued by the callee. callee.
If the INVITE request contained a Location header, the callee sends If the INVITE request contained a Location header, the callee sends
the BYE request to that address rather than the From address. the BYE request to that address rather than the From address.
supported by all other SIP server types. This method MUST be supported by proxy servers and SHOULD be
supported by redirect and user agent SIP servers.
4.2.5 CANCEL 4.2.5 CANCEL
The CANCEL request cancels any pending searches, but does not The CANCEL request cancels a pending request with the same Call-ID,
terminate an accepted call at a particular user agent. (A call is To, From and CSeq (sequence number only) header values, but does
considered accepted if the callee has returned a 200 (OK) status not affect a completed request. (A request is considered completed if
response.) Any client MAY issue a CANCEL request at any time. A the server has returned a final status response.)
proxy, in particular, MAY choose to send a CANCEL to destinations
that have not yet returned a final response after it has received a A user agent client or proxy client MAY issue a CANCEL request at
2xx or 6xx response for one or more of the parallel-search requests. any time. A proxy, in particular, MAY choose to send a CANCEL to
A proxy that receives a CANCEL request forwards the request to all destinations that have not yet returned a final response after it has
destinations with pending requests triggered by an INVITE. The received a 2xx or 6xx response for one or more of the parallel-search
Call-ID, To and From in the CANCEL request are identical to those requests. A proxy that receives a CANCEL request forwards the
contained in the INVITE request, but the Via header field is request to all destinations with pending requests. The Call-ID, To
initialized to the proxy issuing the CANCEL request. and From in the CANCEL request are identical to those contained in
the request being canceled, but the Via header field is initialized
to the proxy issuing the CANCEL request. (Thus, responses to this
CANCEL request only reach the issuing proxy.)
Once a user agent server has received a CANCEL, it MUST NOT issue a Once a user agent server has received a CANCEL, it MUST NOT issue a
2xx response for the cancelled invitation. 2xx response for the cancelled original request.
A redirect server or user agent server returns 200 (OK) if the Call- A redirect server or user agent server returns 200 (OK) if the Call-
ID exists and 481 (Invalid Call-ID) if not, but takes no further ID exists and 481 (Invalid Call-ID) if not, but takes no further
action. In particular, any existing call is unaffected. action. In particular, any existing call is unaffected.
The BYE request cannot be used to cancel branches of a The BYE request cannot be used to cancel branches of a
parallel search, since several branches may, through parallel search, since several branches may, through
intermediate proxies, find the same user agent server and intermediate proxies, find the same user agent server and
then terminate the call. then terminate the call. To terminate a call instead of
just pending searches, the UAC must use BYE instead of or
in addition to CANCEL. While CANCEL can terminate any
pending request other than ACK or CANCEL, it is typically
useful only for INVITE. 200 responses to INVITE and 200
responses to CANCEL are distinguished by the method in the
Cseq header field, so there is no ambiguity.
This method MUST be supported by proxy servers and SHOULD be This method MUST be supported by proxy servers and SHOULD be
supported by all other SIP server types. supported by all other SIP server types.
4.2.6 REGISTER 4.2.6 REGISTER
A client uses the REGISTER method to register the address listed in A client uses the REGISTER method to register the address listed in
the To header to a SIP server. the To header with a SIP server.
A user agent SHOULD register with a local server on startup by A user agent SHOULD register with a local server on startup by
sending a REGISTER request to the well-known "all SIP servers" sending a REGISTER request to the well-known "all SIP servers"
multicast address, 224.0.1.75, with a time-to-live value of 1. multicast address, 224.0.1.75, with a time-to-live value of 1. SIP
user agents on the same subnet MAY listen to that address and use it
SIP user agents on the same subnet MAY listen to that address and use to become aware of the location of other local users [16]; however,
it to become aware of the location of other local users [28]; they do not respond to the request.
however, they do not respond to the request.
The REGISTER request interprets header fields as follows. We define The REGISTER request-header fields are defined as follows. We define
"address-of-record" as the SIP address that the registry knows the "address-of-record" as the SIP address that the registry knows the
registrand under, typically of the form "user@domain" rather than registrand, typically of the form "user@domain" rather than
"user@host". In third-party registration, the entity issuing the "user@host". In third-party registration, the entity issuing the
request is different from the entity being registered. request is different from the entity being registered.
To: The To header field contains the address-of-record whose
registration is to be created or updated. registration is to be created or updated.
From: The From header field contains the address-of-record of the From: The From header field contains the address-of-record of the
person responsible for the registration. For first-party person responsible for the registration. For first-party
registration, it is identical to the To header field value. registration, it is identical to the To header field value.
Request-URI: The Request-URI names the destination of the Request-URI: The Request-URI names the destination of the
registration request, i.e., the domain of the registrar. The registration request, i.e., the domain of the registrar. The
user name MUST be empty. Generally, the domains in the user name MUST be empty. Generally, the domains in the
Request-URI and the To header have the same value; however, it Request-URI and the To header have the same value; however, it
is possible to register as a "visitor", while maintaining one's is possible to register as a "visitor", while maintaining one's
name. For example, a traveller sip:alice@acme.com may register name. For example, a traveller sip:alice@acme.com ( To) may
under sip:@atlanta.ayh.org , with the former as the To field and register under the Request-URI sip:@atlanta.ayh.org , with the
the latter as the Request-URI. The request is no longer former as the To field and the latter as the Request-URI. The
forwarded once it reached the server whose authoritative domain request is no longer forwarded once it reached the server whose
is the one listed in the Request-URI. authoritative domain is the one listed in the Request-URI.
Location: If the request contains a Location header field, requests Location: The request MUST contain a Location header field; requests
for the Request-URI will also be directed to the address(es) for the Request-URI will be directed to the address(es) given.
given. It is recommended that user agents include both SIP UDP It is RECOMMENDED that user agents include SIP URLs with both
and TCP addresses in their registration. Registrations are UDP and TCP transport parameters in their registration. If the
additive. registration contains a Location field whose URL includes a
transport parameter, future requests will use that protocol.
Otherwise, requests use the same transport protocol as used by
the registration. However, a multicast REGISTER request still
causes future requests to be unicast unless the maddr URL
parameter explicitly requests otherwise. If the Location header
does not contain a port number, the default SIP port number is
used for future requests.
We cannot require that registration and requests use the We cannot require that registration and subsequent INVITE
same transport protocol, as multicast registrations may be requests use the same transport protocol, as multicast
quite useful. registrations may be quite useful.
Otherwise, future call control requests will be directed to the Registrations are additive, but all current locations must share the
network source address of the REGISTER request, using the To same action value. A proxy server ignores the q parameter, while a
address in the REGISTER request as the Request-URI. If the redirect server simply returns the parameter in its Location header.
registration is changed while a user agent or proxy server processes
an invitation, the new information should be used. Having the proxy server interpret the q parameter is not
sufficient to guide proxy behavior, as it is not clear, for
example, how long it should wait between trying addresses.
If the registration is changed while a user agent or proxy server
processes an invitation, the new information should be used.
This allows a service known as "directed pick-up". This allows a service known as "directed pick-up".
After registration, the server MAY forward incoming SIP requests to A server SHOULD silently drop the registration after one hour, unless
the network source address and port that originated the registration refreshed by the client. A client may request a lower or higher
request. A server SHOULD silently drop the registration after one refresh interval through the Expires header (Section 6.19). Based on
hour, unless refreshed by the client. A client may request a lower or this request and its configuration, the server chooses the expiration
higher refresh interval through the Expires header (Section 6.19). interval and indicates it through the Expires header in the
Based on this request and its configuration, the server chooses the response. A single address (if host-independent) may be registered
expiration interval and indicates it through the Expires header in from several different clients.
the response. A single address (if host-independent) may be
registered from several different clients.
A client cancels an existing registration by sending a REGISTER A client cancels an existing registration by sending a REGISTER
request with an expiration time ( Expires) of zero seconds for a request with an expiration time ( Expires) of zero seconds for a
particular Location or the wildcard Location designated by a "*" particular Location or the wildcard Location designated by a "*"
for all registrations. for all registrations.
The server SHOULD return the current list of registrations in the 200
response as Location header fields.
It is particularly important that REGISTER requests are
authenticated since they allow to redirect future requests.
Beyond its use as a simple location service, this method is Beyond its use as a simple location service, this method is
needed if there are several SIP servers on a single host, needed if there are several SIP servers on a single host.
so that some cannot use the default port number. Each such In that case, only one of the servers can use the default
server would register with a server for the administrative port number. Each server that cannot would register with a
domain. Since a client may not have easy access to the host server for the administrative domain. Since a client may
address or port number, using the source address and port not have easy access to the host address or port number,
from the request itself seems simpler. using the source address and port from the request itself
seems simpler.
Support of this method is RECOMMENDED. Support of this method is RECOMMENDED.
4.3 Request-URI 4.3 Request-URI
The Request-URI field is a SIP URL as described in Section 2 or a The Request-URI field is a SIP URL as described in Section 2 or a
general URI. It indicates the user or service to which this request general URI. It indicates the user or service to which this request
is being addressed. Unlike the To field, the Request-URI field may is being addressed. Unlike the To field, the Request-URI field may
be re-written by proxies. For example, a proxy may perform a lookup be re-written by proxies.
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 The SIP-URL MUST NOT contain the transport-param, maddr-param,
of requests it generates. ttl-param, or headers elements. A server that receives a SIP-URL
with these elements removes them before further processing.
Typically, the UAC sets the Request-URI and To to the same
SIP URL, presumed to remain unchanged over long time
periods. However, if the UAC has cached a more direct path
to the callee, e.g., from the Location header of a
response to a previous request, the To would still contain
the long-term, "public" address, while the Request-URI
would be set to the cached address.
Proxy and redirect servers may use the information in the Request-URI
and request header fields to handle the request and possibly rewrite
the Request-URI. For example, a request addressed to the generic
address sip:sales@acme.com might be proxied to the particular person,
e.g., sip:bob@ny.acme.com , with the To remaining as sales@acme.com
ny.acme.com , Bob may have designated Alice as the temporary
substitute.
The host part of the Request-URI typically agrees with one of the The host part of the Request-URI typically agrees with one of the
host names of the server. If it does not, the server SHOULD proxy the host names of the server. If it does not, the server SHOULD proxy the
request to the address indicated or return a 404 (Not Found) response request to the address indicated or return a 404 (Not Found) response
if it is unwilling or unable to do so. The case where the Request-URI if it is unwilling or unable to do so. For example, the Request-URI
and server host name disagrees occurs, for example, for a firewall and server host name may disagree in the case of a firewall proxy
proxy that handles outgoing calls. It is similar to the operation of that handles outgoing calls. This mode of operation similar to that
HTTP proxies. of HTTP proxies.
If a SIP server receives a request with a URI indicating a scheme If a SIP server receives a request with a URI indicating a scheme
other than SIP which that server does not understand, the server MUST other than SIP which that server does not understand, the server MUST
return a 400 (Bad Request) response. It MUST do this even if the To return a 400 (Bad Request) response. It MUST do this even if the To
field contains a scheme it does understand. field contains a scheme it does understand.
4.3.1 SIP Version 4.3.1 SIP Version
Both request and response messages include the version of SIP in use, Both request and response messages include the version of SIP in use,
and basically follow [H3.1], with HTTP replaced by SIP. To be and basically follow [H3.1], with HTTP replaced by SIP. To be
conditionally compliant with this specification, applications sending conditionally compliant with this specification, applications sending
SIP messages MUST include a SIP-Version of "SIP/2.0". SIP messages MUST include a SIP-Version of "SIP/2.0".
4.4 Option Tags 4.4 Option Tags
Option tags are unique identifiers used to designate new options in Option tags are unique identifiers used to designate new options in
SIP. These tags are used in Require (Section 6.29) and Unsupported SIP. These tags are used in Require (Section 6.30) and Unsupported
(Section 6.38) fields. (Section 6.38) fields.
Syntax: Syntax:
option-tag ___ 1*urlc option-tag ___ 1*uric
The creator of a new SIP option should either prefix the option with
a reverse domain name or register the new option with the Internet a reverse domain name or register the new option with the Internet
Assigned Numbers Authority (IANA). For example, Assigned Numbers Authority (IANA). For example,
"com.foo.mynewfeature" is an apt name for a feature whose inventor "com.foo.mynewfeature" is an apt name for a feature whose inventor
can be reached at "foo.com". Options registered with IANA have the can be reached at "foo.com". Options registered with IANA have the
prefix "org.ietf.sip.", options described in RFCs have the prefix prefix "org.ietf.sip.", options described in RFCs have the prefix
"org.ietf.rfc.N", where N is the RFC number. Option tags are case- "org.ietf.rfc.N", where N is the RFC number. Option tags are case-
insensitive. insensitive.
4.4.1 Registering New Option Tags with IANA 4.4.1 Registering New Option Tags with IANA
skipping to change at page 26, line 4 skipping to change at page 29, line 31
Response = Status-Line ; Section 5.1 Response = Status-Line ; Section 5.1
*( general-header *( general-header
| response-header | response-header
| entity-header ) | entity-header )
CRLF CRLF
[ message-body ] ; Section 8 [ message-body ] ; Section 8
[H6] applies except that HTTP-Version is replaced by SIP-Version. [H6] applies except that HTTP-Version is replaced by SIP-Version.
Also, SIP defines additional response codes and does not use some Also, SIP defines additional response codes and does not use some
HTTP codes.
5.1 Status-Line 5.1 Status-Line
The first line of a Response message is the Status-Line, consisting The first line of a Response message is the Status-Line, consisting
of the protocol version (Section 4.3.1) followed by a numeric of the protocol version (Section 4.3.1) followed by a numeric
Status-Code and its associated textual phrase, with each element Status-Code and its associated textual phrase, with each element
separated by SP characters. No CR or LF is allowed except in the separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence. final CRLF sequence.
Status-Line ___ SIP-version SP Status-Code SP Reason-Phrase CRLF Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF
5.1.1 Status Codes and Reason Phrases 5.1.1 Status Codes and Reason Phrases
The Status-Code is a 3-digit integer result code that indicates the The Status-Code is a 3-digit integer result code that indicates the
outcome of the attempt to understand and satisfy the request. The outcome of the attempt to understand and satisfy the request. The
Reason-Phrase is intended to give a short textual description of the Reason-Phrase is intended to give a short textual description of the
Status-Code. The Status-Code is intended for use by automata, Status-Code. The Status-Code is intended for use by automata,
whereas the Reason-Phrase is intended for the human user. The client whereas the Reason-Phrase is intended for the human user. The client
is not required to examine or display the Reason-Phrase. is not required to examine or display the Reason-Phrase.
Status-Code = Informational Fig. 5
| Success Fig. 5
| Redirection Fig. 6
| Client-Error Fig. 7
| Server-Error Fig. 8
| Global-Failure Fig. 9
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
We provide an overview of the Status-Code below, and provide full We provide an overview of the Status-Code below, and provide full
definitions in Section 7. The first digit of the Status-Code defines definitions in Section 7. The first digit of the Status-Code defines
the class of response. The last two digits do not have any the class of response. The last two digits do not have any
categorization role. SIP/2.0 allows 6 values for the first digit: categorization role. SIP/2.0 allows 6 values for the first digit:
1xx: Informational -- request received, continuing process; 1xx: Informational -- request received, continuing to process the
request;
2xx: Success -- the action was successfully received, understood, and 2xx: Success -- the action was successfully received, understood, and
accepted; accepted;
3xx: Redirection -- further action must be taken in order to complete 3xx: Redirection -- further action must be taken in order to complete
the request; the request;
4xx: Client Error -- the request contains bad syntax or cannot be 4xx: Client Error -- the request contains bad syntax or cannot be
fulfilled at this server; fulfilled at this server;
5xx: Server Error -- the server failed to fulfill an apparently valid 5xx: Server Error -- the server failed to fulfill an apparently valid
request; request;
6xx: Global Failure - the request is invalid at any server. 6xx: Global Failure -- the request is invalid at any server.
Presented below are the individual values of the numeric response Figures 5 through 9 present the individual values of the numeric
codes, and an example set of corresponding reason phrases for response codes, and an example set of corresponding reason phrases
SIP/2.0. These reason phrases are only recommended; they may be for SIP/2.0. These reason phrases are only recommended; they may be
replaced by local equivalents without affecting the protocol. Note replaced by local equivalents without affecting the protocol. Note
that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
codes in the range starting at x80 to avoid conflicts with newly codes in the range starting at x80 to avoid conflicts with newly
defined HTTP response codes, and extends these response codes in the defined HTTP response codes, and adds a new class, 6xx, of response
6xx range. codes.
Status-Code = Informational Fig. 3 SIP response codes are extensible. SIP applications are not required
| Success Fig. 3 to understand the meaning of all registered response codes, though
| Redirection Fig. 4 such understanding is obviously desirable. However, applications MUST
| Client-Error Fig. 5 understand the class of any response code, as indicated by the first
| Server-Error Fig. 6 digit, and treat any unrecognized response as being equivalent to the
| Global-Failure Fig. 7 x00 response code of that class, with the exception that an
| extension-code unrecognized response MUST NOT be cached. For example, if a client
extension-code = 3DIGIT receives an unrecognized response code of 431, it can safely assume
Reason-Phrase = *<TEXT, excluding CR, LF> that there was something wrong with its request and treat the
response as if it had received a 400 (Bad Request) 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.
Informational = "100" ; Trying Informational = "100" ; Trying
| "180" ; Ringing | "180" ; Ringing
| "181" ; Call Is Being Forwarded | "181" ; Call Is Being Forwarded
| "182" ; Queued | "182" ; Queued
Success = "200" ; OK Success = "200" ; OK
Figure 3: Informational and success status codes Figure 5: Informational and success status codes
Redirection = "300" ; Multiple Choices Redirection = "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 | "380" ; Alternative Service
Figure 4: Redirection status codes Figure 6: Redirection status codes
SIP response codes are extensible. SIP applications are not required
to understand the meaning of all registered response codes, though
such understanding is obviously desirable. However, applications MUST
understand the class of any response code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 response code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if a client
receives an unrecognized response code of 431, it can safely assume
that there was something wrong with its request and treat the
response as if it had received a 400 (Bad Request) response code. In
Client-Error = "400" ; Bad Request Client-Error = "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 Timeout | "408" ; Request Timeout
| "409" ; Conflict | "409" ; Conflict
| "410" ; Gone | "410" ; Gone
| "411" ; Length Required | "411" ; Length Required
| "413" ; Request Message Body Too Large | "413" ; Request Message Body Too Large
| "414" ; Request-URI Too Large | "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type | "415" ; Unsupported Media Type
| "420" ; Bad Extension | "420" ; Bad Extension
| "480" ; Temporarily not available | "480" ; Temporarily not available
| "481" ; Invalid Call-ID | "481" ; Invalid Call-ID
| "482" ; Loop Detected | "482" ; Loop Detected
| "483" ; Too Many Hops | "483" ; Too Many Hops
| "484" ; Address Incomplete
| "485" ; Ambiguous
Figure 5: Client error status codes Figure 7: Client error status codes
Server-Error = "500" ; Internal Server Error Server-Error = "500" ; Internal Server Error
| "501" ; Not Implemented | "501" ; Not Implemented
| "502" ; Bad Gateway | "502" ; Bad Gateway
| "503" ; Service Unavailable | "503" ; Service Unavailable
| "504" ; Gateway Timeout | "504" ; Gateway Timeout
| "505" ; SIP Version not supported | "505" ; SIP Version not supported
Figure 6: Server error status codes Figure 8: Server error status codes
include human-readable information which will explain the unusual
status.
6 Header Field Definitions 6 Header Field Definitions
SIP header fields are similar to HTTP header fields in both syntax SIP header fields are similar to HTTP header fields in both syntax
and semantics [H4.2, H14]. In general the ordering of the header and semantics [H4.2, H14]. In general the ordering of the header
fields is not of importance (with the exception of Via fields, see fields is not of importance (with the exception of Via fields, see
below), but proxies MUST NOT reorder or otherwise modify header below), but proxies MUST NOT reorder or otherwise modify header
fields other than by adding a new Via or other hop-by-hop field.
Proxies MUST NOT, for example, change how header fields are broken
across lines. This allows an authentication field to be added after
Global-Failure | "600" ; Busy Global-Failure | "600" ; Busy
| "603" ; Decline | "603" ; Decline
| "604" ; Does not exist anywhere | "604" ; Does not exist anywhere
| "606" ; Not Acceptable | "606" ; Not Acceptable
Figure 7: Global failure status Codes Figure 9: Global failure status codes
fields other than by adding a new Via or other hop-by-hop field.
Proxies MUST NOT, for example, change how header fields are broken
across lines. This allows an authentication field to be added after
the Via fields that will not be invalidated by proxies.
The header fields required, optional and not applicable for each The header fields required, optional and not applicable for each
method are listed in Table 3. The table uses "o" to indicate method are listed in Table 4. The table uses "o" to indicate
optional, "m" mandatory and "-" for not applicable. A "*" indicates optional, "m" mandatory and "-" for not applicable. A "*" indicates
that the header fields are needed only if message body is not empty: that the header fields are needed only if message body is not empty:
The Content-Type and Content-Length headers are required when there The Content-Type and Content-Length headers are required when there
is a valid message body (of non-zero length) associated with the is a valid message body (of non-zero length) associated with the
message (Section 8). message (Section 8).
The "type" column describes the request and response types the header The "type" column describes the request and response types for which
field may be used for. A numeric value indicates the status code for the header field may be used. A numeric value indicates the status
a response, while "R" refers to any request header, "r" to any code for a response, while "R" refers to any request header, "r" to
response header. "g" and "e" designate general (Section 6.1) and any response header. "g" and "e" designate general (Section 6.1) and
entity header (Section 6.2) fields, respectively. entity header (Section 6.2) fields, respectively.
The "enc." column describes whether this message header may be The "enc." column describes whether this message header may be
encrypted end-to-end. A "n" designates fields that MUST NOT be encrypted end-to-end. A "n" designates fields that MUST NOT be
encrypted, while "c" designates fields that SHOULD be encrypted if encrypted, while "c" designates fields that SHOULD be encrypted if
encryption is used. encryption is used.
The "e-e" column has a value of "e" for end-to-end and a value of "h" The "e-e" column has a value of "e" for end-to-end and a value of "h"
for hop-by-hop headers. for hop-by-hop headers.
Other headers may be added as required; a server MAY ignore optional Other headers may be added as required; a server MAY ignore optional
headers that it does not understand. A compact form of these header headers that it does not understand. A compact form of these header
fields is also defined in Section 9 for use over UDP when the request fields is also defined in Section 9 for use over UDP when the request
has to fit into a single packet and size is an issue. has to fit into a single packet and size is an issue.
Table 4 in Appendix A indicates which system components should be Table 5 in Appendix A indicates which system components should be
capable of parsing which header fields. capable of parsing which header fields.
6.1 General Header Fields 6.1 General Header Fields
There are a few header fields that have general applicability for
both request and response messages. These header fields apply only to
the message being transmitted.
General-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields may be given the semantics of general
type enc. e-e ACK BYE CAN INV OPT REG type enc. e-e ACK BYE CAN INV OPT REG
________________________________________________________________________________ ________________________________________________________________________________
Accept R e o o o o o o Accept R e - - - o o o
Accept-Encoding R e o o o o o o Accept-Encoding R e - - - o o o
Accept-Language R n e o o o o o o Accept-Language R n e - o o o o o
Allow 405 e o o o o o o Allow 405 e o o o o o o
Authorization R e o o o o o o Authorization R e o o o o o o
Call-ID g n e m m m m o - Call-ID g n e m m m m m m
Content-Encoding e e * - - * * * Content-Encoding e e * - - * * *
Content-Length e e m - - m m m Content-Length e e m - - m m m
Content-Type e e * - - * * * Content-Type e e * - - * * *
CSeq g n e m m m m m o CSeq g n e m m m m m o
Date g e o o o o o o Date g e o o o o o o
Encryption g n e o o o o o o Encryption g n e o o o o o o
Expires g e - - - o o o Expires g e - - - o o o
From g e m m m m m m From g e m m m m m m
Hide R n h o o o o o o Hide R n h o o o o o o
Location R e - - - - - o Location R e o - - o - m
Location 3xx e - - o o o o Location 2xx e - - - o o -
Location 2xx e - - o o o - Location 3xx e - o - o o o
Location 485 e - o - o o o
Max-Forwards R n e o o o o o o Max-Forwards R n e o o o o o o
Organization R c e - - - o o o Organization R c e - - - o o o
Proxy-Authenticate 407 n h o o o o o o Proxy-Authenticate 407 n h o o o o o o
Proxy-Authorization R n h o o o o o o Proxy-Authorization R n h o o o o o o
Proxy-Require R n h o o o o o o Proxy-Require R n h o o o o o o
Priority R c e - - - o - - Priority R c e - - - o - -
Require R n e o o o o o o Require R n e o o o o o o
Retry-After R c e - - - - - o Retry-After R c e - - - - - o
Retry-After 600,603 c e - - - o - - Retry-After 600,603 c e - - - o - -
Response-Key R c e - o o o o o Response-Key R c e - o o o o o
skipping to change at page 30, line 47 skipping to change at page 34, line 48
Server r c e o o o o o o Server r c e o o o o o o
Subject R c e - - - o - - Subject R c e - - - o - -
Timestamp g e o o o o o o Timestamp g e o o o o o o
To g n e m m m m m m To g n e m m m m m m
Unsupported 420 e o o o o o o Unsupported 420 e o o o o o o
User-Agent R c e o o o o o o User-Agent R c e o o o o o o
Via g n e m m m m m m Via g n e m m m m m m
Warning r e o o o o o o Warning r e o o o o o o
WWW-Authenticate 401 c e o o o o o o WWW-Authenticate 401 c e o o o o o o
Table 3: Summary of header fields Table 4: Summary of header fields
General header fields apply to both request and response messages.
The general-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields may be given the semantics of general
header fields if all parties in the communication recognize them to
be general-header fields. Unrecognized header fields are treated as
entity-header fields.
6.2 Entity Header Fields 6.2 Entity Header Fields
Entity-header fields define meta-information about the message-body
or, if no body is present, about the resource identified by the The entity-header fields define meta-information about the message-
body or, if no body is present, about the resource identified by the
request. The term "entity header" is an HTTP 1.1 term where the request. The term "entity header" is an HTTP 1.1 term where the
response body may contain a transformed version of the message body.
The original message body is referred to as the "entity". We retain
the same terminology for header fields but usually refer to the the same terminology for header fields but usually refer to the
"message body" rather then the entity as the two are the same in SIP. "message body" rather then the entity as the two are the same in SIP.
6.3 Request Header Fields 6.3 Request Header Fields
The request-header fields allow the client to pass additional The request-header fields allow the client to pass additional
information about the request, and about the client itself, to the information about the request, and about the client itself, to the
server. These fields act as request modifiers, with semantics server. These fields act as request modifiers, with semantics
equivalent to the parameters on a programming language method equivalent to the parameters of a programming language method
invocation. invocation.
Request-header field names can be extended reliably only in The request-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of request- experimental header fields MAY be given the semantics of request-
header fields if all parties in the communication recognize them to header fields if all parties in the communication recognize them to
be request-header fields. Unrecognized header fields are treated as be request-header fields. Unrecognized header fields are treated as
entity-header fields. entity-header fields.
6.4 Response Header Fields 6.4 Response Header Fields
The response-header fields allow the server to pass additional The response-header fields allow the server to pass additional
information about the response which cannot be placed in the Status- information about the response which cannot be placed in the Status-
skipping to change at page 31, line 37 skipping to change at page 36, line 4
further access to the resource identified by the Request-URI. further access to the resource identified by the Request-URI.
Response-header field names can be extended reliably only in Response-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 response-
header fields if all parties in the communication recognize them to header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as be response-header fields. Unrecognized header fields are treated as
entity-header fields. entity-header fields.
6.5 End-to-end and Hop-by-hop Headers 6.5 End-to-end and Hop-by-hop Headers
End-to-end headers must be transmitted unmodified across all proxies, End-to-end headers must be transmitted unmodified across all proxies,
while hop-by-hop headers may be modified or added by proxies. while hop-by-hop headers may be modified or added by proxies.
6.6 Header Field Format 6.6 Header Field Format
Header fields ( general-header, request-header, response-header, and Header fields ( general-header, request-header, response-header, and
entity-header) follow the same generic header format as that given in entity-header) follow the same generic header format as that given in
Section 3.1 of RFC 822 [27,29]. Section 3.1 of RFC 822 [24]. Each header field consists of a name
followed by a colon (":") and the field value. Field names are case-
Each header field consists of a name followed by a colon (":") and insensitive. The field value may be preceded by any amount of leading
the field value. Field names are case-insensitive. The field value white space (LWS), though a single space (SP) is preferred. Header
may be preceded by any amount of leading white space (LWS), though a fields can be extended over multiple lines by preceding each extra
single space (SP) is preferred. Header fields can be extended over line with at least one SP or horizontal tab (HT). Applications SHOULD
multiple lines by preceding each extra line with at least one SP or follow HTTP "common form" when generating these constructs, since
horizontal tab (HT). Applications SHOULD follow HTTP "common form" there might exist some implementations that fail to accept anything
when generating these constructs, since there might exist some beyond the common forms.
implementations that fail to accept anything beyond the common forms.
message-header = field-name ":" [ field-value ] CRLF message-header = field-name ":" [ field-value ] CRLF
field-name = token field-name = token
field-value = *( field-content | LWS ) field-value = *( field-content | LWS )
field-content = < the OCTETs making up the field-value field-content = < the OCTETs making up the field-value
and consisting of either *TEXT or combinations and consisting of either *TEXT
of token, tspecials, and quoted-string> or combinations of token,
tspecials, and quoted-string>
The order in which header fields are received is not significant if The order in which header fields are received is not significant if
the header fields have different field names. Multiple header fields 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 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- 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 separated list (i.e., #(values)). It MUST be possible to combine the
multiple header fields into one "field-name: field-value" pair, multiple header fields into one "field-name: field-value" pair,
without changing the semantics of the message, by appending each without changing the semantics of the message, by appending each
subsequent field-value to the first, each separated by a comma. The subsequent field-value to the first, each separated by a comma. The
order in which header fields with the same field-name are received is order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field therefore significant to the interpretation of the combined field
value, and thus a proxy MUST NOT change the order of these field value, and thus a proxy MUST NOT change the order of these field
values when a message is forwarded. values when a message is forwarded.
Field names are not case-sensitive, although their values may be. Field names are not case-sensitive, although their values may be.
6.7 Accept 6.7 Accept
See [H14.1] for syntax. This request-header field is used only with See [H14.1] for syntax. This request-header field is used only with
the OPTIONS and INVITE request methods to indicate what media types the INVITE, OPTIONS and REGISTER request methods to indicate what
are acceptable in the response. media types are acceptable in the response.
Example: Example:
Accept: application/sdp;level=1, application/x-private, text/html Accept: application/sdp;level=1, application/x-private, text/html
6.8 Accept-Encoding 6.8 Accept-Encoding
The Accept-Encoding request-header field is similar to Accept, but The Accept-Encoding request-header field is similar to Accept, but
restricts the content-codings [H3.4.1] that are acceptable in the restricts the content-codings [H3.4.1] that are acceptable in the
response. See [H14.3]. response. See [H14.3].
6.9 Accept-Language 6.9 Accept-Language
See [H14.4] for syntax. The Accept-Language request header can be See [H14.4] for syntax. The Accept-Language request header can be
used to allow the client to indicate to the server in which language used to allow the client to indicate to the server in which language
it would prefer to receive reason phrases, session descriptions or it would prefer to receive reason phrases, session descriptions or
status responses carried as message bodies. This may also be used as status responses carried as message bodies. A proxy may use this
a hint by the proxy to which destination to connect the call to field to help select the destination for the call, for example, a
(e.g., for selecting a human operator). human operator conversant in a language spoken by the caller.
Example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7 Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.10 Allow 6.10 Allow
See [H14.7]. The Allow entity-header field lists the set of methods See [H14.7]. The Allow entity-header field lists the set of methods
supported by the resource identified by the Request-URI. The purpose supported by the resource identified by the Request-URI. The purpose
of this field is strictly to inform the recipient of valid methods of this field is strictly to inform the recipient of valid methods
associated with the resource. An Allow header field MUST be present associated with the resource. An Allow header field MUST be present
in a 405 (Method Not Allowed) response. in a 405 (Method Not Allowed) response.
6.11 Authorization 6.11 Authorization
See [H14.8] and [30]. A user agent that wishes to authenticate itself See [H14.8].
with a server -- usually, but not necessarily, after receiving a 401
response -- MAY do so by including an Authorization request-header A user agent that wishes to authenticate itself with a server --
field with the request. The Authorization field value consists of usually, but not necessarily, after receiving a 401 response -- MAY
credentials containing the authentication information of the user do so by including an Authorization request-header field with the
agent for the realm of the resource being requested. request. The Authorization field value consists of credentials
containing the authentication information of the user agent for the
realm of the resource being requested.
6.12 Call-ID 6.12 Call-ID
The Call-ID general header uniquely identifies a particular The Call-ID general header uniquely identifies a particular
invitation. Note that a single multimedia conference may give rise to invitation or all registrations of a particular client. Note that a
several calls with different Call-IDs, e.g., if a user invites a single multimedia conference may give rise to several calls with
single individual several times to the same (long-running) different Call-IDs, e.g., if a user invites a single individual
conference. several times to the same (long-running) conference.
For an INVITE request, a callee client application alerts the user For an INVITE request, a callee user agent server SHOULD NOT alert
only if the user has not responded previously to the Call-ID in the the user if the user has responded previously to the Call-ID in the
INVITE request. If the user is already a member of the conference and INVITE request. If the user is already a member of the conference and
the conference parameters contained in the session description have the conference parameters contained in the session description have
not changed, a callee client application MAY silently accept the not changed, a callee user agent server MAY silently accept the call,
call, regardless of the Call-ID. An invitation for an existing regardless of the Call-ID. An invitation for an existing Call-ID or
Call-ID or session may change the parameters of the conference. A session may change the parameters of the conference. A client
client application MAY decide to simply indicate to the user that the application MAY decide to simply indicate to the user that the
conference parameters have been changed and accept the invitation conference parameters have been changed and accept the invitation
automatically or it MAY require user confirmation. automatically or it MAY require user confirmation.
A user may be invited to the same conference or call using several A user may be invited to the same conference or call using several
different Call-IDs. If desired, the client may use identifiers different Call-IDs. If desired, the client may use identifiers
within the session description to detect this duplication. For within the session description to detect this duplication. For
example, SDP contains a session id and version number in the origin ( example, SDP contains a session id and version number in the origin (
o) field. o) field.
The REGISTER and OPTIONS methods use the Call-ID value to
unambiguously match requests and responses. All REGISTER requests
issued by a single client MUST use the same Call-ID.
The Call-ID may be any string consisting of the unreserved URI The Call-ID may be any string consisting of the unreserved URI
characters that can be guaranteed to be globally unique for the characters that can be guaranteed to be globally unique for the
duration of the request. Call-IDs are case-sensitive and are not duration of the request. Call-IDs are case-sensitive and are not
URL-encoded. URL-encoded.
Since the Call-ID is generated by and for SIP, there is no
reason to deal with the complexity of URL-encoding and reason to deal with the complexity of URL-encoding and
case-ignoring string comparison. case-ignoring string comparison.
The form UUID@host is recommended, where host is either the fully Call-ID = ( "Call-ID" | "i" ) ":" local-id "@" host
qualified domain name or a globally routable IP address. The UUID is local-id = *uric
a version-4 (random) UUID [31].
Using cryptographically random identifiers provides some host MUST be either a fully qualified domain name or a globally
protection against session hijacking. routable IP address, while the local-id is a random identifier
unique within host. The use of a UUID as local-id is OPTIONAL. The
UUID is a version-4 (random) UUID [19].
Call-ID = ( "Call-ID" | "i" ) ":" UUID "@" host Using cryptographically random identifiers provides some
protection against session hijacking. Call-ID, To and
From are needed to identify a call leg call leg matters in
calls with third-party control.
Example: Example:
Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
6.13 Content-Encoding 6.13 Content-Encoding
The Content-Encoding entity-header field is used as a modifier to The Content-Encoding entity-header field is used as a modifier to
the media-type. When present, its value indicates what additional the media-type. When present, its value indicates what additional
content codings have been applied to the entity-body, and thus what content codings have been applied to the entity-body, and thus what
decoding mechanisms MUST be applied in order to obtain the media-type decoding mechanisms MUST be applied in order to obtain the media-type
referenced by the Content-Type header field. Content-Encoding is referenced by the Content-Type header field. Content-Encoding is
primarily used to allow a document to be compressed without losing primarily used to allow a document to be compressed without losing
the identity of its underlying media type. See [H14.11]. the identity of its underlying media type. See [H14.12].
6.14 Content-Length 6.14 Content-Length
The Content-Length entity-header field indicates the size of the The Content-Length entity-header field indicates the size of the
message-body, in decimal number of octets, sent to the recipient. message-body, in decimal number of octets, sent to the recipient.
Content-Length = "Content-Length" ":" 1*DIGIT Content-Length = "Content-Length" ":" 1*DIGIT
An example is An example is
Content-Length: 3495 Content-Length: 3495
Applications MUST use this field to indicate the size of the Applications MUST use this field to indicate the size of the
message-body to be transferred, regardless of the media type of the message-body to be transferred, regardless of the media type of the
entity. Any Content-Length greater than or equal to zero is a valid entity. Any Content-Length greater than or equal to zero is a valid
value. If no body is present in a message, then the Content-Length value. If no body is present in a message, then the Content-Length
how to determine the length of the message body. header MUST be set to zero. If a server receives a message without
Content-Length, it MUST assume it to be zero. Section 8 describes how
to determine the length of the message body.
6.15 Content-Type 6.15 Content-Type
The Content-Type entity-header field indicates the media type of the The Content-Type entity-header field indicates the media type of the
message-body sent to the recipient. message-body sent to the recipient. The media-type element is
defined in [H3.7].
Content-Type = "Content-Type" ":" media-type Content-Type = "Content-Type" ":" media-type
Example of this header field are Examples of this header field are
Content-Type: application/sdp Content-Type: application/sdp
Content-Type: text/html; charset=ISO-8859-4 Content-Type: text/html; charset=ISO-8859-4
6.16 CSeq 6.16 CSeq
Clients MUST add the CSeq (command sequence) general-header field to Clients MUST add the CSeq (command sequence) general-header field to
every request. A CSeq request header field contains a single decimal every request. A CSeq request header field contains a single decimal
sequence number chosen by the requesting client, unique within a sequence number chosen by the requesting client, unique within a
single value of Call-ID or, for requests without Call-ID, within the single value of Call-ID. The sequence number MUST be expressible as
request type. The sequence number MUST be expressible as a 32-bit a 32-bit unsigned integer. The initial value of the sequence number
unsigned integer. The initial value of the sequence number is is arbitrary, but MUST be less than 2**31. Consecutive requests that
arbitrary, but a value of zero is RECOMMENDED. Consecutive requests differ in request method, headers or body, but have the same Call-ID
that differ in request method, headers or body, but have the same MUST contain strictly monotonically increasing and contiguous
Call-ID MUST contain strictly monotonically increasing and contiguous
sequence numbers; sequence numbers do not wrap around. sequence numbers; sequence numbers do not wrap around.
Retransmissions of the same request carry the same sequence number, Retransmissions of the same request carry the same sequence number,
but an INVITE with a different message body or different header but an INVITE with a different message body or different header
fields (a "re-invitation") acquires a new, higher sequence number. A fields (a "re-invitation") acquires a new, higher sequence number. A
server responding to a request containing a CSeq header MUST echo server MUST echo the CSeq value from the request in its response. If
the value in the response. If the Method value is missing, the the Method value is missing, the server fills it in appropriately.
server fills it it appropriately.
The ACK and CANCEL requests MUST contain the same CSeq value as The ACK and CANCEL requests MUST contain the same CSeq value as
the INVITE request that it refers to, while a BYE request the INVITE request that it refers to, while a BYE request
cancelling an invitation MUST have a higher sequence number. cancelling an invitation MUST have a higher sequence number.
A user agent server MUST remember the highest sequence number for any A user agent server MUST remember the highest sequence number for any
INVITE request with the same Call-ID value. The server MUST respond INVITE request with the same Call-ID value. The server MUST respond
to, but ignore any INVITE request with a lower sequence number. to, but ignore any INVITE request with a lower sequence number.
All requests spawned in a parallel search have the same CSeq value All requests spawned in a parallel search have the same CSeq value
as the request triggering the parallel search. as the request triggering the parallel search.
CSeq = "CSeq" ":" 1*DIGIT Method
Strictly speaking, CSeq header fields are needed for any Strictly speaking, CSeq header fields are needed for any
SIP request that can be cancelled by a BYE or CANCEL SIP request that can be cancelled by a BYE or CANCEL
request or where a client can issue several requests for request or where a client can issue several requests for
the same Call-ID in close succession. Without a sequence the same Call-ID in close succession. Without a sequence
number, the response to an INVITE could be mistaken for number, the response to an INVITE could be mistaken for
the response to the cancellation ( BYE or CANCEL). Also, the response to the cancellation ( BYE or CANCEL). Also,
if the network duplicates packets or if an ACK is delayed if the network duplicates packets or if an ACK is delayed
until the server has sent an additional response, the until the server has sent an additional response, the
client could interpret an old response as the response to a client could interpret an old response as the response to a
re-invitation issued shortly thereafter. Using CSeq also re-invitation issued shortly thereafter. Using CSeq also
skipping to change at page 36, line 27 skipping to change at page 41, line 26
body. body.
The Method value allows the client to distinguish the response to an The Method value allows the client to distinguish the response to an
INVITE request from that of a CANCEL response. CANCEL requests can INVITE request from that of a CANCEL response. CANCEL requests can
be generated by proxies; if they were to increase the sequence be generated by proxies; if they were to increase the sequence
number, it might conflict with a later request issued by the user number, it might conflict with a later request issued by the user
agent for the same call. agent for the same call.
With a length of 32 bits, a server could generate, within a single With a length of 32 bits, a server could generate, within a single
call, one request a second for about 136 years before needing to wrap call, one request a second for about 136 years before needing to wrap
around. around. The initial value of the sequence number is chosen so that
subsequent requests within the same call will not wrap around. A
non-zero initial value allows to use a time-based initial sequence
number, which protects against ambiguities when clients are re-
invited to the same call after rebooting. A client could, for
example, choose the 31 most significant bits of a 32-bit second clock
as an initial sequence number.
Forked requests must have the same CSeq as there would be ambiguity Forked requests must have the same CSeq as there would be ambiguity
otherwise between these forked requests and later BYE issued by the otherwise between these forked requests and later BYE issued by the
client user agent. client user agent.
Example: Example:
CSeq: 4711 INVITE CSeq: 4711 INVITE
6.17 Date 6.17 Date
General header field. See [H14.19]. General header field. See [H14.19].
The Date header field is useful for simple devices without The Date header field can be used by simple end systems
their own clock. without a battery-backed clock to acquire a notion of
current time.
6.18 Encryption 6.18 Encryption
The Encryption general-header field specifies that the content has The Encryption general-header field specifies that the content has
been encrypted. Section 12 describes the overall SIP security been encrypted. Section 12 describes the overall SIP security
architecture and algorithms. It is intended for end-to-end encryption architecture and algorithms. This header field is intended for end-
of requests and responses. Requests are encrypted with a public key to-end encryption of requests and responses. Requests are encrypted
belonging to the entity named in the To header field. Responses are with a public key belonging to the entity named in the To header
encrypted with the public key conveyed in the Response-Key header field. Responses are encrypted with the public key conveyed in the
Response-Key header field.
SIP chose not to adopt HTTP's Content-Transfer-Encoding SIP chose not to adopt HTTP's Content-Transfer-Encoding
header because the encrypted body may contain additional header because the encrypted body may contain additional
SIP header fields as well as the body of the message. SIP header fields as well as the body of the message.
For any encrypted message, at least the message body and possibly For any encrypted message, at least the message body and possibly
other message header fields are encrypted. An application receiving a other message header fields are encrypted. An application receiving a
request or response containing an Encryption header field decrypts request or response containing an Encryption header field decrypts
the body and then concatenates the plaintext to the request line and the body and then concatenates the plaintext to the request line and
headers of the original message. Message headers in the decrypted headers of the original message. Message headers in the decrypted
part completely replace those with the same field name in the part completely replace those with the same field name in the
skipping to change at page 37, line 42 skipping to change at page 43, line 11
The following example for a message encrypted with ASCII-armored PGP The following example for a message encrypted with ASCII-armored PGP
was generated by applying "pgp -ea" to the payload to be encrypted. was generated by applying "pgp -ea" to the payload to be encrypted.
INVITE sip:watson@boston.bell-telephone.com SIP/2.0 INVITE sip:watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5 Via: SIP/2.0/UDP 169.130.12.5
From: <sip:a.g.bell@bell-telephone.com> From: <sip:a.g.bell@bell-telephone.com>
To: T. A. Watson <sip:watson@bell-telephone.com> To: T. A. Watson <sip:watson@bell-telephone.com>
Call-ID: 187602141351@worcester.bell-telephone.com Call-ID: 187602141351@worcester.bell-telephone.com
Content-Length: 885 Content-Length: 885
Encryption: PGP,version=2.6.2,encoding=ascii Encryption: PGP version=2.6.2,encoding=ascii
hQEMAxkp5GPd+j5xAQf/ZDIfGD/PDOM1wayvwdQAKgGgjmZWe+MTy9NEX8O25Red hQEMAxkp5GPd+j5xAQf/ZDIfGD/PDOM1wayvwdQAKgGgjmZWe+MTy9NEX8O25Red
h0/pyrd/+DV5C2BYs7yzSOSXaj1C/tTK/4do6rtjhP8QA3vbDdVdaFciwEVAcuXs h0/pyrd/+DV5C2BYs7yzSOSXaj1C/tTK/4do6rtjhP8QA3vbDdVdaFciwEVAcuXs
ODxlNAVqyDi1RqFC28BJIvQ5KfEkPuACKTK7WlRSBc7vNPEA3nyqZGBTwhxRSbIR ODxlNAVqyDi1RqFC28BJIvQ5KfEkPuACKTK7WlRSBc7vNPEA3nyqZGBTwhxRSbIR
RuFEsHSVojdCam4htcqxGnFwD9sksqs6LIyCFaiTAhWtwcCaN437G7mUYzy2KLcA RuFEsHSVojdCam4htcqxGnFwD9sksqs6LIyCFaiTAhWtwcCaN437G7mUYzy2KLcA
zPVGq1VQg83b99zPzIxRdlZ+K7+bAnu8Rtu+ohOCMLV3TPXbyp+err1YiThCZHIu zPVGq1VQg83b99zPzIxRdlZ+K7+bAnu8Rtu+ohOCMLV3TPXbyp+err1YiThCZHIu
X9dOVj3CMjCP66RSHa/ea0wYTRRNYA/G+kdP8DSUcqYAAAE/hZPX6nFIqk7AVnf6 X9dOVj3CMjCP66RSHa/ea0wYTRRNYA/G+kdP8DSUcqYAAAE/hZPX6nFIqk7AVnf6
IpWHUPTelNUJpzUp5Ou+q/5P7ZAsn+cSAuF2YWtVjCf+SQmBR13p2EYYWHoxlA2/ IpWHUPTelNUJpzUp5Ou+q/5P7ZAsn+cSAuF2YWtVjCf+SQmBR13p2EYYWHoxlA2/
GgKADYe4M3JSwOtqwU8zUJF3FIfk7vsxmSqtUQrRQaiIhqNyG7KxJt4YjWnEjF5E
WUIPhvyGFMJaeQXIyGRYZAYvKKklyAJcm29zLACxU5alX4M25lHQd9FR9Zmq6Jed
wbWvia6cAIfsvlZ9JGocmQYF7pcuz5pnczqP+/yvRqFJtDGD/v3s++G2R+ViVYJO wbWvia6cAIfsvlZ9JGocmQYF7pcuz5pnczqP+/yvRqFJtDGD/v3s++G2R+ViVYJO
z/lxGUZaM4IWBCf+4DUjNanZM0oxAE28NjaIZ0rrldDQmO8V9FtPKdHxkqA5iJP+ z/lxGUZaM4IWBCf+4DUjNanZM0oxAE28NjaIZ0rrldDQmO8V9FtPKdHxkqA5iJP+
6vGOFti1Ak4kmEz0vM/Nsv7kkubTFhRl05OiJIGr9S1UhenlZv9l6RuXsOY/EwH2 6vGOFti1Ak4kmEz0vM/Nsv7kkubTFhRl05OiJIGr9S1UhenlZv9l6RuXsOY/EwH2
z8X9N4MhMyXEVuC9rt8/AUhmVQ== z8X9N4MhMyXEVuC9rt8/AUhmVQ==
=bOW+ =bOW+
Since proxies may base their forwarding decision on any combination Since proxies may base their forwarding decision on any combination
of SIP header fields, there is no guarantee that an encrypted request of SIP header fields, there is no guarantee that an encrypted request
"hiding" header fields will reach the same destination as an "hiding" header fields will reach the same destination as an
otherwise identical un-encrypted request. otherwise identical un-encrypted request.
6.19 Expires 6.19 Expires
The Expires entity-header field gives the date and time after which The Expires entity-header field gives the date and time after which
the message content expires. the message content expires.
This header field is currently defined only for the REGISTER and This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a request and response-header INVITE methods. For REGISTER, it is a request and response-header
field and allows the client to indicate how long the registration is field and allows the client to indicate how long the registration is
to be valid; the server uses it to indicate when the client has to to be valid; the server uses it to indicate when the client has to
re-register. The server's choice overrides that of the client. The re-register the addresses contained in the request. The server's
server MAY choose a shorter time interval than that requested by the choice overrides that of the client. The server MAY choose a shorter
client, but SHOULD not choose a longer one. time interval than that requested by the client, but SHOULD NOT
choose a longer one.
For INVITE, it is a request and response-header field. In a request, For INVITE, it is a request and response-header field. In a request,
the callee can limit the validity of an invitation. (For example, if the callee can limit the validity of an invitation. For example, if a
a client wants to limit how long a search should take at most or when client wants to limit how long a search should take at most or when a
a conference invitation is time-limited. A user interface may take conference invitation is time-limited. A user interface may take this
this is as a hint to leave the invitation window on the screen even as a hint to leave the invitation window on the screen even if the
if the user is not currently at the workstation.) This also limits user is not currently at the workstation. This also limits the
the duration of a search. If the request expires before the search duration of a search. If the request expires before the search
completes, the proxy returns a 408 (Request Timeout) status. In a 302 completes, the proxy returns a 408 (Request Timeout) status. In a 302
(Moved Temporarily) response, a server can advise the client of the (Moved Temporarily) response, a server can advise the client of the
maximal duration of the redirection. maximal duration of the redirection.
The value of this field can be either an HTTP-date or an integer The value of this field can be either an HTTP-date or an integer
number of seconds (in decimal), measured from the receipt of the number of seconds (in decimal), measured from the receipt of the
request. The latter approach is preferable for short durations, as it request. The latter approach is preferable for short durations, as it
does not depend on clients and servers sharing a synchronized clock. does not depend on clients and servers sharing a synchronized clock.
Expires = "Expires" ":" ( HTTP-date | delta-seconds ) Expires = "Expires" ":" ( HTTP-date | delta-seconds )
Two example of its use are Two examples of its use are
Expires: Thu, 01 Dec 1994 16:00:00 GMT Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5 Expires: 5
6.20 From 6.20 From
Requests and responses MUST contain a From general-header field, Requests and responses MUST contain a From general-header field,
indicating the initiator of the request. The server copies the To and indicating the initiator of the request. The server copies the To and
From header fields from the request to the response. The optional From header fields from the request to the response. The optional
display-name is meant to be rendered by a human user interface. display-name is meant to be rendered by a human-user interface.
The SIP-URL MUST NOT contain the transport-param, maddr-param,
ttl-param, or headers elements. A server that receives a SIP-URL
with these elements removes them before further processing.
From = ( "From" | "f" ) ":" ( name-addr | addr-spec ) From = ( "From" | "f" ) ":" ( name-addr | addr-spec )
name-addr = [ display-name ] "<" addr-spec ">" name-addr = [ display-name ] "<" addr-spec ">"
addr-spec = SIP-URL | URI addr-spec = SIP-URL | URI
display-name = *token | quoted-string display-name = *token | quoted-string
Examples: Examples:
From: A. G. Bell <sip:agb@bell-telephone.com> From: A. G. Bell <sip:agb@bell-telephone.com>
From: sip:+12125551212@server.phone2net.com From: sip:+12125551212@server.phone2net.com
From: Anonymous <sip:c8oqz84zk7z@privacy.org> From: Anonymous <sip:c8oqz84zk7z@privacy.org>
Call-ID, To and From are needed to identify a call leg Call-ID, To and From are needed to identify a call leg
matters in calls with third-party control. The format is matters in calls with third-party control. The format is
similar to the equivalent RFC 822 header, but with a URI similar to the equivalent RFC 822 [24] header, but with a
instead of just an email address. URI instead of just an email address.
6.21 Hide 6.21 Hide
The Hide request header field indicates that the path comprised of The Hide request header field indicates that the path comprised of
the Via header fields (Section 6.40) should be hidden from the Via header fields (Section 6.40) should be hidden from
subsequent proxies and user agents. It can take two forms: Hide: subsequent proxies and user agents. It can take two forms: Hide:
route and Hide:hop. Hide header fields are typically added by the route and Hide:hop. Hide header fields are typically added by the
client user agent, but MAY be added by any proxy along the path. client user agent, but MAY be added by any proxy along the path.
If a request contains the " Hide: route" header field, all following If a request contains the " Hide: route" header field, all following
skipping to change at page 40, line 5 skipping to change at page 45, line 33
anonymous. anonymous.
A server hides the previous hop by encrypting the host and port A server hides the previous hop by encrypting the host and port
parts of the top-most Via header with an algorithm of its choice. parts of the top-most Via header with an algorithm of its choice.
Servers SHOULD add additional "salt" to the host and port Servers SHOULD add additional "salt" to the host and port
information prior to encryption to prevent malicious downstream information prior to encryption to prevent malicious downstream
proxies from guessing earlier parts of the path based on seeing proxies from guessing earlier parts of the path based on seeing
identical encrypted Via headers. Hidden Via fields are marked with identical encrypted Via headers. Hidden Via fields are marked with
the hidden Via option, as described in Section 6.40. the hidden Via option, as described in Section 6.40.
A server that is capable of hiding Via headers MUST attempt to
decrypt all Via headers marked as "hidden" to perform loop decrypt all Via headers marked as "hidden" to perform loop
detection. Servers that are not capable of hiding can ignore hidden detection. Servers that are not capable of hiding can ignore hidden
Via fields in their loop detection algorithm. Via fields in their loop detection algorithm.
If hidden headers were not marked, a proxy would have to If hidden headers were not marked, a proxy would have to
decrypt all headers to detect loops, just in case one was decrypt all headers to detect loops, just in case one was
encrypted, as the Hide: Hop option may have been removed encrypted, as the Hide: Hop option may have been removed
along the way. along the way.
A host MUST NOT add such a " Hide:hop" header field unless it can A host MUST NOT add such a " Hide:hop" header field unless it can
skipping to change at page 40, line 44 skipping to change at page 46, line 29
The Hide header field has the following syntax: The Hide header field has the following syntax:
Hide = "Hide" ":" ( "route" | "hop" ) Hide = "Hide" ":" ( "route" | "hop" )
6.22 Location 6.22 Location
The Location general-header field can appear in requests, 2xx The Location general-header field can appear in requests, 2xx
responses and 3xx responses. responses and 3xx responses.
REGISTER requests: REGISTER requests MAY contain Location header REGISTER requests: REGISTER requests MUST contain a Location header
fields. They indicate under which locations the user may be field indicating at which locations the user may be reachable.
reachable. The REGISTER request defines a wildcard Location The REGISTER request defines a wildcard Location field, "*",
field, "*". that is only used with Expires: 0 to remove all which is only used with Expires: 0 to remove all registrations
registrations for a particular user. for a particular user.
INVITE and ACK requests: INVITE and ACK requests MAY contain INVITE and ACK requests: INVITE and ACK requests SHOULD contain
Location headers indicating the location the request is Location headers indicating from which location the request is
originating from. If the SIP address does not refer to the user originating. If the SIP address does not refer to the user agent
server, the SIP URL MUST contain a tag parameter uniquely
identifying the user agent. (The same person may be logged on at
several locations within the same domain served by the proxy.) several locations within the same domain served by the proxy.)
This allows the callee to send a BYE directly to the This allows the callee to send a BYE directly to the
caller instead of through a series of proxies. The Via caller instead of through a series of proxies. The Via
header is not sufficient since the desired address may be header is not sufficient since the desired address may be
that of a proxy. that of a proxy.
INVITE 2xx responses: A user agent server sending a definitive, INVITE 2xx responses: A user agent server sending a definitive,
positive response (2xx), MAY insert a Location response header positive response (2xx) MAY insert a Location response header
indicating the SIP address under which it is reachable most indicating the SIP address under which it is reachable most
directly for future SIP requests, such as ACK. This may be the directly for future SIP requests, such as ACK. This may be the
address of the server itself or that of a proxy, e.g., if the address of the server itself or that of a proxy, e.g., if the
host is behind a firewall. If the SIP address does not refer to host is behind a firewall. If the SIP address does not refer to
the user agent server, the SIP URL MUST contain a tag parameter the user agent server, the SIP URL MUST contain a tag parameter
uniquely identifying the user agent. (The same person may be uniquely identifying the user agent. (The same person may be
logged on at several locations within the same domain served by logged on at several locations within the same domain served by
the proxy.) The value of this Location header is copied into the proxy.) The value of this Location header is copied into
the Request-URI of subsequent ACK and BYE requests for this the Request-URI of subsequent requests for this call.
call.
REGISTER 2xx responses: Similarly, a REGISTER response SHOULD return REGISTER 2xx responses: Similarly, a REGISTER response SHOULD return
all locations that a user is currently reachable under. all locations at which the user is currently reachable.
3xx responses: The Location response-header field can be used with a 3xx responses: The Location response-header field can be used with a
3xx response codes to indicate one or more addresses to try. It 3xx response codes to indicate one or more addresses to try. It
can appear in responses to INVITE and OPTIONS methods. The can appear in responses to BYE, INVITE and OPTIONS methods.
Location header field contains URIs giving the new locations or The Location header field contains URIs giving the new
user names to try, or may simply specify additional transport locations or user names to try, or may simply specify additional
parameters. A 301 (Moved Permanently) or 302 (Moved Temporarily) transport parameters. A 301 (Moved Permanently) or 302 (Moved
response SHOULD contain a Location field containing URIs of new Temporarily) response SHOULD contain a Location field
addressed to be tried. A 301 or 302 response may also give the containing URIs of new addressed to be tried. A 301 or 302
same location and username that was being tried but specify response may also give the same location and username that was
additional transport parameters such as a multicast address to being tried but specify additional transport parameters such as
try or a change of SIP transport from UDP to TCP or vice versa. a multicast address to try or a change of SIP transport from UDP
to TCP or vice versa.
Note that the Location header may also refer to a different entity Note that the Location header may also refer to a different entity
than the one originally called. For example, a SIP call connected to than the one originally called. For example, a SIP call connected to
GSTN gateway may need to deliver a special information announcement GSTN gateway may need to deliver a special information announcement
such as "The number you have dialed has been changed." such as "The number you have dialed has been changed."
A Location response header may contain any suitable URI indicating A Location response header may contain any suitable URI indicating
where the called party may be reached, not limited to SIP URLs. For where the called party may be reached, not limited to SIP URLs. For
example, it may contain a phone or fax URL [22], a mailto: URL [19] example, it may contain a phone or fax
or irc: URL.
a mailto: (RFC 2368, [25]) or irc: URL.
The following parameters are defined. Additional parameters may be The following parameters are defined. Additional parameters may be
defined in other specifications. defined in other specifications.
q: The qvalue indicates the relative preference among the locations q: The qvalue indicates the relative preference among the locations
given. qvalue values are decimal numbers from 0.0 to 1.0, with given. qvalue values are decimal numbers from 0.0 to 1.0, with
higher values indicating higher preference.
action: The action is only used when registering with the REGISTER action: The action is only used when registering with the REGISTER
request. It indicates how the client wishes forwarding to occur, request. It indicates whether the client wishes that the server
by proxying or by redirection. The action taken if this proxies or redirects future requests intended for the client.
parameter is not specified depends on server configuration. In The action taken if this parameter is not specified depends on
its response, the registrar SHOULD indicate the mode used. This server configuration. In its response, the registrar SHOULD
parameter is ignored for other requests. indicate the mode used. This parameter is ignored for other
requests.
Location = ( "Location" | "m" ) ":" ("*" | (1# (( SIP-URL | URI ) Location = ( "Location" | "m" ) ":"
*( ";" location-params ))) ("*" | (1# (( SIP-URL | URI )
[ LWS *( ";" location-params ) ] ))
location-params = "q" "=" qvalue location-params = "q" "=" qvalue
| "action" "=" "proxy" | "redirect" | "action" "=" "proxy" | "redirect"
| extension-attribute | extension-attribute
extension-attribute = extension-name [ "=" extension-value ] extension-attribute = extension-name [ "=" extension-value ]
Example: Example:
Location: sip:watson@worcester.bell-telephone.com;tag=123 Location: sip:watson@worcester.bell-telephone.com;tag=123
;q=0.7, ;q=0.7,
skipping to change at page 43, line 4 skipping to change at page 49, line 5
Max-Forwards = "Max-Forwards" ":" 1*DIGIT Max-Forwards = "Max-Forwards" ":" 1*DIGIT
The Max-Forwards value is a decimal integer indicating the remaining The Max-Forwards value is a decimal integer indicating the remaining
number of times this request message may be forwarded. number of times this request message may be forwarded.
Each proxy or gateway recipient of a request containing a Max- Each proxy or gateway recipient of a request containing a Max-
Forwards header field MUST check and update its value prior to Forwards header field MUST check and update its value prior to
forwarding the request. If the received value is zero (0), the forwarding the request. If the received value is zero (0), the
recipient MUST NOT forward the request. Instead, for the OPTIONS and recipient MUST NOT forward the request. Instead, for the OPTIONS and
REGISTER methods, it MUST respond as the final recipient. For all REGISTER methods, it MUST respond as the final recipient. For all
other methods, the server returns 483 (Too many hops).
If the received Max-Forwards value is greater than zero, then the If the received Max-Forwards value is greater than zero, then the
forwarded message MUST contain an updated Max-Forwards field with a forwarded message MUST contain an updated Max-Forwards field with a
value decremented by one (1). value decremented by one (1).
Example: Example:
Max-Forwards: 6 Max-Forwards: 6
6.24 Organization 6.24 Organization
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The value of "emergency" should only be used when life, limb or The value of "emergency" should only be used when life, limb or
property are in imminent danger. property are in imminent danger.
Examples: Examples:
Subject: A tornado is heading our way! Subject: A tornado is heading our way!
Priority: emergency Priority: emergency
Subject: Weekend plans Subject: Weekend plans
Priority: non-urgent Priority: non-urgent
These are the values of RFC 2076 [26], with the addition of
These are the values of RFC 2076, with the addition of
"emergency". "emergency".
6.26 Proxy-Authenticate
The Proxy-Authenticate response-header field MUST be included as The Proxy-Authenticate response-header field MUST be included as
part of a 407 (Proxy Authentication Required) response. The field part of a 407 (Proxy Authentication Required) response. The field
value consists of a challenge that indicates the authentication value consists of a challenge that indicates the authentication
scheme and parameters applicable to the proxy for this Request-URI. scheme and parameters applicable to the proxy for this Request-URI.
See [H14.33] for further details. See [H14.33] for further details.
A client SHOULD cache the credentials used for a particular proxy A client SHOULD cache the credentials used for a particular proxy
server and realm for the next request to that server. Credentials server and realm for the next request to that server. Credentials
are, in general, valid for a specific value of the Request-URI at a are, in general, valid for a specific value of the Request-URI at a
particular proxy server. If a client contacts a proxy server that has particular proxy server. If a client contacts a proxy server that has
required authentication in the past, but the client does not have required authentication in the past, but the client does not have
credentials for the particular Request-URI, it MAY attempt to use credentials for the particular Request-URI, it MAY attempt to use
the most-recently used credential. The server responds with 401 the most-recently used credential. The server responds with 401
(Unauthorized) if the client guessed wrong. (Unauthorized) if the client guessed wrong.
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[H14.34] for further details. [H14.34] for further details.
6.28 Proxy-Require 6.28 Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive The Proxy-Require header is used to indicate proxy-sensitive
features that MUST be supported by the proxy. Any Proxy-Require features that MUST be supported by the proxy. Any Proxy-Require
header features that are not supported by the proxy MUST be header features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client if not supported. negatively acknowledged by the proxy to the client if not supported.
Servers treat this field identically to the Require field. Servers treat this field identically to the Require field.
See Section 6.29 for more details on the mechanics of this message See Section 6.30 for more details on the mechanics of this message
and a usage example. and a usage example.
6.29 Require 6.29 Record-Route
The Record-Route request and response header field is added to an
INVITE request by any proxy that insists on being in the path of
subsequent ACK and BYE requests for the same call. It contains a
globally reachable Request-URI that identifies the proxy server.
Each proxy server adds its Request-URI to the beginning of the list.
The server copies the Record-Route header unchanged into the
response. ( Record-Route is only relevant for 2xx responses.)
The calling user agent client copies the Record-Route header into a
Route header of subsequent requests, reversing the order of requests,
so that the first entry is closest to the caller. If the response
contained a Location header field, the calling user agent adds its
content as the last Route header. Unless this would cause a loop,
any client MUST send any subsequent requests for this Call-ID to the
first Request-URI in the Route request header and remove that entry.
Some proxies, such as those controlling firewalls or in an
automatic call distribution (ACD) system, need to maintain
call state and thus need to receive any BYE and ACK
packets for the call.
The Record-Route header field has the following syntax:
Record-Route = "Record-Route" ":" 1# request-uri
Example for a request that has traversed the hosts ieee.org and
bell-telephone.com , in that order:
Record-Route: sip:a.g.bell@bell-telephone.com, sip:a.bell@ieee.org
6.30 Require
The Require request header is used by clients to tell user agent The Require request header is used by clients to tell user agent
servers about options that the client expects the server to support servers about options that the client expects the server to support
in order to properly process the request. If a server does not in order to properly process the request. If a server does not
understand the option, it MUST respond by returning status code 420 understand the option, it MUST respond by returning status code 420
(Bad Extension) and list those options it does not understand in the (Bad Extension) and list those options it does not understand in the
Unsupported header. Unsupported header.
Require = "Require" ":" 1#option-tag Require = "Require" ":" 1#option-tag
Example: Example:
C->S: INVITE sip:watson@bell-telephone.com SIP/2.0 C->S: INVITE sip:watson@bell-telephone.com SIP/2.0
Require: com.example.billing Require: com.example.billing
Payment: sheep_skins, conch_shells Payment: sheep_skins, conch_shells
S->C: SIP/2.0 420 Bad Extension S->C: SIP/2.0 420 Bad Extension
Unsupported: com.example.billing Unsupported: com.example.billing
This is to make sure that the client-server interaction This is to make sure that the client-server interaction
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mechanisms. In addition, it also removes ambiguity when the mechanisms. In addition, it also removes ambiguity when the
client requires features that the server does not client requires features that the server does not
understand. Some features, such as call handling fields, understand. Some features, such as call handling fields,
are only of interest to end systems. are only of interest to end systems.
Proxy and redirect servers MUST ignore features that are not Proxy and redirect servers MUST ignore features that are not
understood. If a particular extension requires that intermediate understood. If a particular extension requires that intermediate
devices support it, the extension should be tagged in the Proxy- devices support it, the extension should be tagged in the Proxy-
Require field instead (see Section 6.28). Require field instead (see Section 6.28).
6.30 Record-Route
The Record-Route request and response header field is added to an
INVITE request by any proxy that insists on being in the path of
subsequent ACK and BYE requests for the same call. It contains a
globally reachable Request-URI that identifies the proxy server.
Each proxy server adds its Request-URI to the beginning of the list.
The server copies the Record-Route header unchanged into the
response. ( Record-Route is only relevant for 2xx responses.)
The calling user agent client copies the Record-Route header into a
Route header of subsequent requests, reversing the order of requests,
so that the first entry is closest to the caller. If the response
contained a Location header field, the calling user agent adds its
content as the last Route header. Unless this would cause a loop,
any clientMUST send any subsequent requests for this Call-ID to the
first Request-URI in the Route request header and remove that entry.
call state and thus need to receive any BYE and ACK
packets for the call.
The Record-Route header field has the following syntax:
Record-Route = "Record-Route" ":" 1# request-uri
Example for a request that has traversed the hosts ieee.org and
bell-telephone.com , in that order:
Record-Route: sip:a.g.bell@bell-telephone.com, sip:a.bell@ieee.org
6.31 Response-Key 6.31 Response-Key
The Response-Key request header field can be used by a client to The Response-Key request header field can be used by a client to
request the key that the called user agent SHOULD use to encrypt the request the key that the called user agent SHOULD use to encrypt the
response with. The syntax is: response with. The syntax is:
Response-Key = "Response-Key" ":" key-scheme 1*SP #key-param Response-Key = "Response-Key" ":" key-scheme 1*SP #key-param
key-scheme = token key-scheme = token
key-param = token "=" ( token | quoted-string ) key-param = token "=" ( token | quoted-string )
The key-scheme gives the type of encryption to be used for response. The key-scheme gives the type of encryption to be used for the
Section 12 describes security schemes. response. Section 12 describes security schemes.
If the client insists that the server return an encrypted response, If the client insists that the server return an encrypted response,
it includes a it includes a
Require: org.ietf.sip.encrypt-response Require: org.ietf.sip.encrypt-response
header field in its request. If the client cannot encrypt for header field in its request. If the client cannot encrypt for
whatever reason, it MUST follow normal Require header field whatever reason, it MUST follow normal Require header field
procedures and return an 420 (Bad Extension) response. If this procedures and return a 420 (Bad Extension) response. If this Require
Require header is not present, a client SHOULD still encrypt, but MAY header is not present, a client SHOULD still encrypt, but MAY return
return an unencrypted response if unable to. an unencrypted response if unable to.
6.32 Route
The Route request header determines the route taken by a request.
Each host removes the first entry and then proxies the request to the
host listed in that entry, also using it as the Request-URI. The
operation is further described in Section 6.30.
The Route header field has the following syntax:
6.33 Retry-After 6.32 Retry-After
The Retry-After response header field can be used with a 503 The Retry-After response header field can be used with a 503
(Service Unavailable) response to indicate how long the service is (Service Unavailable) response to indicate how long the service is
expected to be unavailable to the requesting client and with a 404 expected to be unavailable to the requesting client and with a 404
(Not Found), 600 (Busy), or 603 (Decline) response to indicate when (Not Found), 600 (Busy), or 603 (Decline) response to indicate when
the called party may be available again. The value of this field can the called party may be available again. The value of this field can
be either an HTTP-date or an integer number of seconds (in decimal) be either an HTTP-date or an integer number of seconds (in decimal)
after the time of the response. after the time of the response.
A REGISTER request may include this header field when deleting A REGISTER request may include this header field when deleting
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then indicates when the user might again be reachable. The registrar then indicates when the user might again be reachable. The registrar
MAY then include this information in responses to future calls. MAY then include this information in responses to future calls.
An optional comment can be used to indicate additional information An optional comment can be used to indicate additional information
about the time of callback. An optional duration parameter indicates about the time of callback. An optional duration parameter indicates
how long the called party will be reachable starting at the initial how long the called party will be reachable starting at the initial
time of availability. If no duration parameter is given, the service time of availability. If no duration parameter is given, the service
is assumed to be available indefinitely. is assumed to be available indefinitely.
Retry-After = "Retry-After" ":" ( HTTP-date | delta-seconds ) Retry-After = "Retry-After" ":" ( HTTP-date | delta-seconds )
[ comment ] [ ";duration" "=" delta-seconds [ comment ] [ ";duration" "=" delta-seconds ]
Examples of its use are Examples of its use are
Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting) Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
Retry-After: Mon, 1 Jan 9999 00:00:00 GMT Retry-After: Mon, 1 Jan 9999 00:00:00 GMT
(Dear John: Don't call me back, ever) (Dear John: Don't call me back, ever)
Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600 Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600
Retry-After: 120 Retry-After: 120
In the third example, the callee is reachable for one hour starting In the third example, the callee is reachable for one hour starting
at 21:00 GMT. In the last example, the delay is 2 minutes. at 21:00 GMT. In the last example, the delay is 2 minutes.
6.33 Route
The Route request header determines the route taken by a request.
Each host removes the first entry and then proxies the request to the
host listed in that entry, also using it as the Request-URI. The
operation is further described in Section 6.29.
The Route header field has the following syntax:
Route = "Route" ":" 1# request-uri
6.34 Server 6.34 Server
The Server response-header field contains information about the The Server response-header field contains information about the
software used by the user agent server to handle the request. See software used by the user agent server to handle the request. See
[H14.39]. [H14.39].
6.35 Subject 6.35 Subject
This is intended to provide a summary, or indicate the nature, of the This is intended to provide a summary, or to indicate the nature, of
call, allowing call filtering without having to parse the session the call, allowing call filtering without having to parse the session
description. (Also, the session description may not necessarily use description. (Also, the session description may not necessarily use
the same subject indication as the invitation.)
Subject = ( "Subject" | "s" ) ":" *text Subject = ( "Subject" | "s" ) ":" *text
Example: Example:
Subject: Tune in - they are talking about your work! Subject: Tune in - they are talking about your work!
6.36 Timestamp 6.36 Timestamp
The timestamp general header describes when the client sent the The timestamp general header describes when the client sent the
request to the server. The value of the timestamp is of significance request to the server. The value of the timestamp is of significance
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this, add a floating point number indicating the number of seconds this, add a floating point number indicating the number of seconds
that have elapsed since it has received the request. The timestamp is that have elapsed since it has received the request. The timestamp is
used by the client to compute the round-trip time to the server so used by the client to compute the round-trip time to the server so
that it can adjust the timeout value for retransmissions. that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ] Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ] delay = *(DIGIT) [ "." *(DIGIT) ]
6.37 To 6.37 To
The To general-header field specifies the invited user, with the The To general-header field specifies recipient of the request, with
same SIP URL syntax as the From field. the same SIP URL syntax as the From field.
To = ( "To" | "t" ) ":" ( name-addr | addr-spec ) To = ( "To" | "t" ) ":" ( name-addr | addr-spec )
The UAS copies the To header into its response, but SHOULD add a
tag parameter if not already present. It MAY forego adding the tag
parameter if there is no chance that another UAS responds to the same
request.
A SIP server returns a 400 (Bad Request) response if it receives a A SIP server returns a 400 (Bad Request) response if it receives a
request with a To header field containing a URI with a scheme it request with a To header field containing a URI with a scheme it
does not recognize. does not recognize.
Example: Example:
To: The Operator <sip:operator@cs.columbia.edu> To: The Operator <sip:operator@cs.columbia.edu>
To: sip:+12125551212@server.phone2net.com To: sip:+12125551212@server.phone2net.com
Call-ID, To and From are needed to identify a call leg Call-ID, To and From are needed to identify a call leg
matters in calls with third-party control. matters in calls with third-party control. The tag is
added to the To header in the response to allow forking of
future requests for the same call by proxies, while
addressing only one of the possibly several responding user
agent servers. It also allows several instances of the
callee to send requests that can be distinguished.
6.38 Unsupported
The Unsupported response header lists the features not supported by The Unsupported response header lists the features not supported by
the server. See Section 6.29 for a usage example and motivation. the server. See Section 6.30 for a usage example and motivation.
6.39 User-Agent 6.39 User-Agent
The User-Agent request-header field contains information about the The User-Agent request-header field contains information about the
client user agent originating the request. See [H14.42]. client user agent originating the request. See [H14.42].
6.40 Via 6.40 Via
The Via field indicates the path taken by the request so far. This The Via field indicates the path taken by the request so far. This
prevents request looping and ensures replies take the same path as prevents request looping and ensures replies take the same path as
the requests, which assists in firewall traversal and other unusual the requests, which assists in firewall traversal and other unusual
routing situations. routing situations.
6.40.1 Requests 6.40.1 Requests
The client originating the request MUST insert into the request a Via The client originating the request MUST insert into the request a Via
field containing its host name or network address and, if not the field containing its host name or network address and, if not the
default port number, the port number it wishes to receive responses default port number, the port number at which it wishes to receive
at. (Note that this port number may differ from the UDP source port responses. (Note that this port number may differ from the UDP source
number of the request.) A fully-qualified domain name is RECOMMENDED. port number of the request.) A fully-qualified domain name is
Each subsequent proxy server that sends the request onwards MUST add RECOMMENDED. Each subsequent proxy server that sends the request
its own additional Via field before any existing Via fields. onwards MUST add its own additional Via field before any existing
Via fields. A proxy that receives a redirection (3xx) response and
A proxy that receives a redirection (3xx) response and then searches then searches recursively, MUST use the same Via headers as on the
recursively, MUST use the same Via headers as on the original original request.
request.
A proxy SHOULD check the top-most Via header to ensure that it A proxy SHOULD check the top-most Via header to ensure that it
contains the sender's correct network address, as seen from that contains the sender's correct network address, as seen from that
proxy. If the sender's address is incorrect, the proxy should add an proxy. If the sender's address is incorrect, the proxy should add an
additional received attribute, as described below. additional received attribute, as described 6.40.2.
A host behind a network address translator (NAT) or A host behind a network address translator (NAT) or
firewall may not be able to insert a network address into firewall may not be able to insert a network address into
the Via header that can be reached by the next hop beyond the Via header that can be reached by the next hop beyond
the NAT. Hosts behind NATs or NAPTs should insert the local the NAT. Hosts behind NATs or NAPTs should insert the local
port number of the outgoing socket, rather than the port port number of the outgoing socket, rather than the port
number for incoming requests, as NAPTs assume that number for incoming requests, as NAPTs assume that
responses return with reversed source and destination responses return with reversed source and destination
ports. ports.
Additionally, if the message goes to a multicast address, an extra Additionally, if the message goes to a multicast address, an extra
Via field is added by the sender before all the other Via fields Via field is added by the sender before all the other Via fields
giving the multicast address and TTL. giving the multicast address and TTL.
If a proxy server receives a request which contains its own address, If a proxy server receives a request which contains its own address,
it MUST respond with a 482 (Loop Detected) status code. it MUST respond with a 482 (Loop Detected) status code.
This prevents a malfunctioning proxy server from causing
loops. Also, it cannot be guaranteed that a proxy server loops. Also, it cannot be guaranteed that a proxy server
can always detect that the address returned by a location can always detect that the address returned by a location
service refers to a host listed in the Via list, as a service refers to a host listed in the Via list, as a
single host may have aliases or several network interfaces. single host may have aliases or several network interfaces.
6.40.2 Receiver-tagged Via Fields 6.40.2 Receiver-tagged Via Fields
Normally every host that sends or forwards a SIP message adds a Via Normally, every host that sends or forwards a SIP message adds a Via
field indicating the path traversed. However, it is possible that field indicating the path traversed. However, it is possible that
Network Address Translators (NAT) may change the source address of Network Address Translators (NAT) may change the source address of
the request, in which case the Via field cannot be relied on to the request (e.g., from net-10 to a globally routable address), in
route replies. To prevent this, a proxy SHOULD check the top-most which case the Via field cannot be relied on to route replies. To
Via header to ensure that it contains the sender's correct network prevent this, a proxy SHOULD check the top-most Via header to ensure
address, as seen from that proxy. If the sender's address is that it contains the sender's correct network address, as seen from
incorrect, the proxy should add a received tag to the Via field that proxy. If the sender's address is incorrect, the proxy should
inserted by the previous hop. Such a modified Via field is known as a add a received tag to the Via field inserted by the previous hop.
receiver-tagged Via field. An example is: Such a modified Via field is known as a receiver-tagged Via field.
An example is:
Via: SIP/2.0/UDP erlang.bell-telephone.com:5060 Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
Via: SIP/2.0/UDP 10.0.0.1:5060 ;received=199.172.136.3 Via: SIP/2.0/UDP 10.0.0.1:5060 ;received=199.172.136.3
In this example, the message went from 10.0.0.1 and through a NAT In this example, the message originated from 10.0.0.1 and traversed a
using external address border.ieee.org (199.172.136.3) to NAT with the external address border.ieee.org (199.172.136.3) to
erlang.bell-telephone.com tagged the previous hop's Via field with reach erlang.bell-telephone.com and tagged the previous hop's Via
the address that it actually came from. field with the address that it actually came from.
6.40.3 Responses 6.40.3 Responses
In the return path, Via fields are processed by a proxy or client In the return path, Via fields are processed by a proxy or client
according to the following rules: according to the following rules:
1. The first Via field should indicate the proxy or client 1. The first Via field should indicate the proxy or client
processing this message. If it does not, discard the processing this message. If it does not, discard the
message. Otherwise, remove this Via field. message. Otherwise, remove this Via field.
2. If the second Via field in a response is a multicast 2. If the second Via field is a receiver-tagged field
address, remove that Via field, and send the message to
the multicast address indicated.
3. If the second Via field is a receiver-tagged field
(Section 6.40.2), send the message to the address in the (Section 6.40.2), send the message to the address in the
received tag. Otherwise, send send the message to the received tag. Otherwise, if the Via header contains a
address indicated in the sent-by parameter. maddr multicast address, send the response to that
multicast address, using the value of the ttl parameter if
given. Otherwise, send the message to the address
indicated in the sent-by parameter.
4. If there is no second Via field, this response is destined 3. If there is no second Via field, this response is destined
for this client. for this client.
These rules ensure that a client only has to check the first Via These rules ensure that a client only has to check the first Via
field in a response to see if it needs processing.
A user agent server or redirect server returns the response to the A user agent server or redirect server returns the response to the
network address where the request came from. (Since these servers do network address where the request came from. (Since these servers do
not forward the request, they do not add a received tag.) not forward the request, they do not add a Via header field or
received tag.)
6.40.4 Syntax 6.40.4 Syntax
The format for a Via header is: The format for a Via header is shown in Fig. 10.
Via = ( "Via" $|$ "v") ":" 1#( sent-protocol sent-by Via = ( "Via" $|$ "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] ) *( ";" via-params ) [ comment ] )
via-params = via-hidden | via-ttl | via-received via-params = via-hidden | via-ttl | via-maddr
| via-branch | via-received | via-branch
via-hidden = "hidden" via-hidden = "hidden"
via-ttl = "ttl" "=" ttl via-ttl = "ttl" "=" ttl
via-maddr = "maddr" "=" maddr
via-received = "received" "=" host via-received = "received" "=" host
via-branch = "branch" "=" token via-branch = "branch" "=" token
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" $|$ token transport = "UDP" $|$ "TCP" $|$ token
sent-by = ( host [ ":" port ] ) $|$ ( concealed-host ) sent-by = ( host [ ":" port ] ) $|$ ( concealed-host )
concealed-host = token concealed-host = token
ttl = 1*3DIGIT ; 0 to 255 ttl = 1*3DIGIT ; 0 to 255
The " ttl" parameter is included only if the address is a multicast Figure 10: Syntax of Via header field
address. The " received" parameter is added only for receiver-added
The defaults for " protocol-name" and " transport" are "SIP" and
"UDP", respectively. The " maddr" parameter, designating the
multicast address, and the " ttl" parameter, designating the time-
to-live (TTL) value, are included only if the request was sent via
multicast. The " received" parameter is added only for receiver-added
Via fields (Section 6.40.2). For reasons of privacy, a client or Via fields (Section 6.40.2). For reasons of privacy, a client or
proxy may wish to hide its Via information by encrypting it (see proxy may wish to hide its Via information by encrypting it (see
Section 6.21). The " hidden" parameter is included if this header Section 6.21). The " hidden" parameter is included if this header was
was hidden by the upstream proxy (see 6.21). hidden by the upstream proxy (see 6.21).
The " branch" parameter is included by every forking proxy. The The " branch" parameter is included by every forking proxy. The
token uniquely identifies a branch of a particular search. The token uniquely identifies a branch of a particular search. The
identifier has to be unique only within a set of isomorphic requests. identifier has to be unique only within a set of isomorphic requests.
Note that privacy of the proxy relies on the cooperation of the next Note that privacy of the proxy relies on the cooperation of the next
hop, as the next-hop proxy will, by necessity, know the IP address hop, as the next-hop proxy will, by necessity, know the IP address
and port number of the source host. and port number of the source host.
Via: SIP/2.0/UDP first.example.com:4000 Via: SIP/2.0/UDP first.example.com:4000;ttl=16
;maddr=224.2.0.1 (Example)
Via: SIP/2.0/UDP adk8 Via: SIP/2.0/UDP adk8
6.41 Warning 6.41 Warning
The Warning response-header field is used to carry additional The Warning response-header field is used to carry additional
information about the status of a response. Warning headers are sent
with responses and have the following format:
Warning = "Warning" ":" 1#warning-value Warning = "Warning" ":" 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text warning-value = warn-code SP warn-agent SP warn-text
warn-code = 3DIGIT "." 2DIGIT warn-code = 3DIGIT
warn-agent = ( host [ ":" port ] ) | pseudonym warn-agent = ( host [ ":" port ] ) | pseudonym
; the name or pseudonym of the server adding ; the name or pseudonym of the server adding
; the Warning header, for use in debugging ; the Warning header, for use in debugging
warn-text = quoted-string warn-text = quoted-string
A response may carry more than one Warning header. A response may carry more than one Warning header.
The warn-text should be in a natural language that is most likely to The warn-text should be in a natural language that is most likely to
be intelligible to the human user receiving the response. This be intelligible to the human user receiving the response. This
decision may be based on any available knowledge, such as the decision may be based on any available knowledge, such as the
location of the cache or user, the Accept-Language field in a location of the cache or user, the Accept-Language field in a
request, the Content-Language field in a response, etc. The default request, the Content-Language field in a response, etc. The default
language is English. language is English.
Any server may add Warning headers to a response. New Warning Any server may add Warning headers to a response. Proxy servers MUST
headers MUST be added after any existing Warning headers. A proxy place additional Warning headers before any Authorization headers.
server MUST NOT delete any Warning header that it received with a Within that constraint, Warning headers MUST be added after any
response. existing Warning headers not covered by a signature. 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 When multiple Warning headers are attached to a response, the user
agent SHOULD display as many of them as possible, in the order that agent SHOULD display as many of them as possible, in the order that
they appear in the response. If it is not possible to display all of they appear in the response. If it is not possible to display all of
the warnings, the user agent first displays warnings that appear the warnings, the user agent first displays warnings that appear
early in the response. early in the response. Systems that generate multiple Warning
headers should order them with this user agent behavior in mind.
Systems that generate multiple Warning headers should order them The warn-code consists of three digits. The first digit indicates the
with this user agent behavior in mind. significance of the warning, with 3xx indicating a warning that did
not cause the request to fail and 4xx indicating a fatal error
condition that contributed to the failure of the request.
Example: This is a list of the currently-defined warn-codes, each with a
recommended warn-text in English, and a description of its meaning.
Additional warn-codes may be defined through IANA. Note that these
warnings describe failures induced by the session description.
Warning: 606.4 isi.edu Multicast not available x01 Insufficient bandwidth: The bandwidth specified in the session
Warning: 606.2 isi.edu Incompatible protocol (RTP/XXP) description or defined by the media exceeds that known to be
available.
x02 Incompatible transport protocol: One or more transport protocols
described in the request are not available.
x03 Incompatible network protocol: One or more network protocols
described in the request are not available.
x04 Incompatible network address formats: One or more network address
formats described in the request are not available.
x05 Incompatible media format: One or more media formats described in
the request are not available.
x06 Incompatible bandwidth description: One or more bandwidth
descriptions contained in the request were not understood.
x07 Multicast not available: The site where the user is located does
not support multicast.
x08 Unicast not available: The site where the user is located does
not support unicast communication (usually due to the presence
of a firewall).
x09 Media type not available: One or more media types contained in
the request are not available.
x10 Attribute not understood: One or more of the media attributes in
the request are not supported.
x09 Session description parameter not understood: A parameter other
than those listed above was not understood.
x99 Miscellaneous warning: The warning text may include arbitrary
information to be presented to a human user, or logged. A system
receiving this warning MUST NOT take any automated action.
1xx and 2xx have been taken by HTTP/1.1.
Examples:
Warning: 309 isi.edu "Session parameter 'foo' not understood"
Warning: 404 isi.edu "Incompatible network address type 'E.164'"
6.42 WWW-Authenticate 6.42 WWW-Authenticate
The WWW-Authenticate response-header field MUST be included in 401 The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages. The field value consists of at (Unauthorized) response messages. The field value consists of at
least one challenge that indicates the authentication scheme(s) and least one challenge that indicates the authentication scheme(s) and
parameters applicable to the Request-URI. parameters applicable to the Request-URI. See [H14.46] and [27].
See [H14.46] and [30].
The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages.
The content of the realm parameter SHOULD be displayed to the user.
A user agent SHOULD cache the authorization credentials for a given A user agent SHOULD cache the authorization credentials for a given
value of the destination ( To header) and realm and attempt to re-use value of the destination ( To header) and realm and attempt to re-use
these values on the next request for that destination. these values on the next request for that destination.
In addition to the "basic" and "digest" authentication schemes In addition to the "basic" and "digest" authentication schemes
defined in the specifications cited above, SIP defines a new scheme, defined in the specifications cited above, SIP defines a new scheme,
PGP (RFC 2015, [32]), Section 13. Other schemes, such as S-MIME, are PGP (RFC 2015, [28]), Section 13. Other schemes, such as S-MIME, are
for further study. for further study.
7 Status Code Definitions 7 Status Code Definitions
The response codes are consistent with, and extend, HTTP/1.1 response The response codes are consistent with, and extend, HTTP/1.1 response
codes. Not all HTTP/1.1 response codes are appropriate, and only codes. Not all HTTP/1.1 response codes are appropriate, and only
those that are appropriate are given here. Other HTTP/1.1 response those that are appropriate are given here. Other HTTP/1.1 response
codes should not be used. Response codes not defined by HTTP/1.1 have codes should not be used. Response codes not defined by HTTP/1.1 have
codes x80 upwards to avoid clashes with future HTTP response codes. codes x80 upwards to avoid clashes with future HTTP response codes.
Also, SIP defines a new class, 6xx. The default behavior for unknown Also, SIP defines a new class, 6xx. The default behavior for unknown
response codes is given for each category of codes. response codes is given for each category of codes.
7.1 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. Typically a server should send a 1xx response if it prompting. Typically a server should send a 1xx response if it
expects to take more than 200 ms to obtain a final response. A server expects to take more than 200 ms to obtain a final response. A
can issue zero or more 1xx responses, with no restriction on their server can issue zero or more 1xx responses, with no restriction on
ordering or uniqueness. Note that 1xx responses are not transmitted their ordering or uniqueness. Note that 1xx responses are not
reliably, that is, they do not cause the client to send an ACK. transmitted reliably, that is, they do not cause the client to send
Servers are free to retransmit informational responses and clients an ACK. Servers are free to retransmit informational responses and
can inquire about the current state of call processing by re-sending clients can inquire about the current state of call processing by
the request. re-sending the request.
7.1.1 100 Trying 7.1.1 100 Trying
Some unspecified action is being taken on behalf of this call (e.g., Some unspecified action is being taken on behalf of this call (e.g.,
a database is being consulted), but the user has not yet been a database is being consulted), but the user has not yet been
located. located.
7.1.2 180 Ringing 7.1.2 180 Ringing
The called user agent has located a possible location where the user The called user agent has located a possible location where the user
has been recently and is trying to alert them. has registered recently and is trying to alert the user.
7.1.3 181 Call Is Being Forwarded 7.1.3 181 Call Is Being Forwarded
A proxy server MAY use this status code to indicate that the call is A proxy server MAY use this status code to indicate that the call is
being forwarded to a different set of destinations. The new being forwarded to a different set of destinations. The new
destinations are listed in Location headers. Proxies SHOULD be destinations are listed in Location headers. Proxies SHOULD be
configurable not to reveal this information. configurable not to reveal this information.
7.1.4 182 Queued
The called party is temporarily unavailable, but the callee has The called party is temporarily unavailable, but the callee has
decided to queue the call rather than reject it. When the callee decided to queue the call rather than reject it. When the callee
becomes available, it will return the appropriate final status becomes available, it will return the appropriate final status
response. The reason phrase MAY give further details about the status response. The reason phrase MAY give further details about the status
of the call, e.g., "5 calls queued; expected waiting time is 15 of the call, e.g., "5 calls queued; expected waiting time is 15
minutes". The server MAY issue several 182 responses to update the minutes". The server MAY issue several 182 responses to update the
caller about the status of the queued call. caller about the status of the queued call.
7.2 Successful 2xx 7.2 Successful 2xx
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whether the address returned by a redirect server equals an address whether the address returned by a redirect server equals an address
tried earlier. tried earlier.
7.3.1 300 Multiple Choices 7.3.1 300 Multiple Choices
The address in the request resolved to several choices, each with its The address in the request resolved to several choices, each with its
own specific location, and the user (or user agent) can select a own specific location, and the user (or user agent) can select a
preferred communication end point and redirect its request to that preferred communication end point and redirect its request to that
location. location.
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, if allowed by the Accept request choose the one most appropriate, if allowed by the Accept request
header. The entity format is specified by the media type given in the header. The entity format is specified by the media type given in the
Content-Type header field. The choices SHOULD also be listed as Content-Type header field. The choices SHOULD also be listed as
Location fields (Section 6.22). Unlike HTTP, the SIP response may Location fields (Section 6.22). Unlike HTTP, the SIP response may
contain several Location fields or a list of addresses in a contain several Location fields or a list of addresses in a
Location field. User agents MAY use the Location field value for Location field. User agents MAY use the Location field value for
automatic redirection or MAY ask the user to confirm a choice. automatic redirection or MAY ask the user to confirm a choice.
However, this specification does not define any standard for such However, this specification does not define any standard for such
automatic selection. automatic selection.
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given by the Location header field (Section 6.22). The duration of given by the Location header field (Section 6.22). The duration of
the redirection can be indicated through an Expires (Section 6.19) the redirection can be indicated through an Expires (Section 6.19)
header. header.
7.3.4 380 Alternative Service 7.3.4 380 Alternative Service
The call was not successful, but alternative services are possible. The call was not successful, but alternative services are possible.
The alternative services are described in the message body of the The alternative services are described in the message body of the
response. response.
7.3.5 381 Ambiguous
The callee address provided in the request was ambiguous. The
response MAY contain a listing of possible unambiguous addresses in
Location headers.
Revealing alternatives may infringe on privacy concerns of the user
or the organization. It MUST be possible to configure a server to
respond with status 404 (Not Found) or to suppress the listing of
possible choices if the request address was ambiguous.
Example response to a request with the URL lee@example.com :
381 Ambiguous SIP/2.0
Location: lee.foote@example.com (Lee M. Foote)
Some email and voice mail systems provide this
functionality. A status code separate from 300 is used
since the semantics are different: for 300, it is assumed
that the same person or service will be reached by the
choices provided. While an automated choice or sequential
search makes sense for a 300 response, user intervention is
required for a 381 response.
7.4 Request Failure 4xx 7.4 Request Failure 4xx
4xx responses are definite failure responses from a particular 4xx responses are definite failure responses from a particular
server. The client SHOULD NOT retry the same request without server. The client SHOULD NOT retry the same request without
modification (e.g., adding appropriate authorization). However, the modification (e.g., adding appropriate authorization). However, the
same request to a different server may be successful. same request to a different server may be successful.
7.4.1 400 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.
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returned if the domain in the Request-URI does not match any of the returned if the domain in the Request-URI does not match any of the
domains handled by the recipient of the request. domains handled by the recipient of the request.
7.4.6 405 Method Not Allowed 7.4.6 405 Method Not Allowed
The method specified in the Request-Line is not allowed for the The method specified in the Request-Line is not allowed for the
address identified by the Request-URI. The response MUST include an address identified by the Request-URI. The response MUST include an
Allow header containing a list of valid methods for the indicated Allow header containing a list of valid methods for the indicated
address. address.
7.4.7 406 Not Acceptable
The resource identified by the request is only capable of generating
response entities which have content characteristics not acceptable
according to the accept headers sent in the request.
7.4.8 407 Proxy Authentication Required
This code is similar to 401 (Unauthorized), but indicates that the This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with the proxy. The proxy MUST client MUST first authenticate itself with the proxy. The proxy MUST
return a Proxy-Authenticate header field (section 6.26) containing a return a Proxy-Authenticate header field (section 6.26) containing a
challenge applicable to the proxy for the requested resource. The challenge applicable to the proxy for the requested resource. The
client MAY repeat the request with a suitable Proxy-Authorization client MAY repeat the request with a suitable Proxy-Authorization
header field (section 6.27). SIP access authentication is explained header field (section 6.27). SIP access authentication is explained
in section 12.3 and [H11]. in section 12.2 and [H11].
This status code should be used for applications where access to the This status code should be used for applications where access to the
communication channel (e.g., a telephony gateway) rather than the communication channel (e.g., a telephony gateway) rather than the
callee herself requires authentication. callee herself requires authentication.
7.4.8 408 Request Timeout 7.4.9 408 Request Timeout
The server could not produce a response, e.g., a user location, The server could not produce a response, e.g., a user location,
within the time indicated in the request via the Expires header. The within the time indicated in the Expires request-header field. The
client MAY repeat the request without modifications at any later client MAY repeat the request without modifications at any later
time. time.
7.4.9 420 Bad Extension 7.4.10 414 Request-URI Too Long
The server is refusing to service the request because the Request-URI
is longer than the server is willing to interpret.
7.4.11 415 Unsupported Media Type
The server is refusing to service the request because the message
body of the request is in a format not supported by the requested
resource for the requested method.
7.4.12 420 Bad Extension
The server did not understand the protocol extension specified in a The server did not understand the protocol extension specified in a
Require (Section 6.29) header field. Require (Section 6.30) header field.
7.4.10 480 Temporarily Unavailable 7.4.13 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is The callee's end system was contacted successfully but the callee is
currently unavailable (e.g., not logged in or logged in in such a currently unavailable (e.g., not logged in or logged in in such a
manner as to preclude communication with the callee). The response manner as to preclude communication with the callee). The response
may indicate a better time to call in the Retry-After header. The may indicate a better time to call in the Retry-After header. The
user may also be available elsewhere (unbeknownst to this host), user may also be available elsewhere (unbeknownst to this host),
thus, this response does not terminate any searches. The reason thus, this response does not terminate any searches. The reason
phrase SHOULD indicate the more precise cause as to why the callee is phrase SHOULD indicate a more precise cause as to why the callee is
unavailable. This value SHOULD be setable by the user agent. unavailable. This value SHOULD be setable by the user agent.
7.4.11 481 Invalid Call-ID 7.4.14 481 Invalid Call-ID
The server received a BYE or CANCEL request with a Call-ID (Section The server received a BYE or CANCEL request with a Call-ID (Section
6.12) value it does not recognize. (A server simply discards an ACK 6.12) value it does not recognize. (A server simply discards an ACK
with an invalid Call-ID.) with an invalid Call-ID.)
7.4.12 482 Loop Detected 7.4.15 482 Loop Detected
The server received a request with a Via (Section 6.40) path The server received a request with a Via (Section 6.40) path
containing itself. containing itself.
7.4.13 483 Too Many Hops 7.4.16 483 Too Many Hops
The server received a request that contains more Via entries (hops) The server received a request that contains more Via entries (hops)
(Section 6.40) than allowed by the Max-Forwards (Section 6.23) (Section 6.40) than allowed by the Max-Forwards (Section 6.23)
header field. header field.
7.4.17 484 Address Incomplete
The server received a request with a To (Section 6.37) address or The server received a request with a To (Section 6.37) address or
Request-URI that was incomplete. Additional information should be Request-URI that was incomplete. Additional information should be
provided. provided.
This status code allows overlapped dialing. This status code allows overlapped dialing. With overlapped
dialing, the client does not know the length of the dialing
string. It sends strings of increasing lengths, prompting
the user for more input, until it no longer receives a 484
status response.
7.4.18 485 Ambiguous
The callee address provided in the request was ambiguous. The
response MAY contain a listing of possible unambiguous addresses in
Location headers.
Revealing alternatives may infringe on privacy concerns of the user
or the organization. It MUST be possible to configure a server to
respond with status 404 (Not Found) or to suppress the listing of
possible choices if the request address was ambiguous.
Example response to a request with the URL lee@example.com :
485 Ambiguous SIP/2.0
Location: sip:carol.lee@example.com (Carol Lee)
Location: sip:p.lee@example.com (Ping Lee)
Location: sip:lee.foote@example.com (Lee M. Foote)
Some email and voice mail systems provide this
functionality. A status code separate from 3xx is used
since the semantics are different: for 300, it is assumed
that the same person or service will be reached by the
choices provided. While an automated choice or sequential
search makes sense for a 3xx response, user intervention is
required for a 485 response.
7.5 Server Failure 5xx 7.5 Server Failure 5xx
5xx responses are failure responses given when a server itself has 5xx responses are failure responses given when a server itself has
erred. They are not definitive failures, and MUST NOT terminate a erred. They are not definitive failures, and MUST NOT terminate a
search if other possible locations remain untried. search if other possible locations remain untried.
7.5.1 500 Server Internal Error 7.5.1 500 Server Internal Error
The server encountered an unexpected condition that prevented it from The server encountered an unexpected condition that prevented it from
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server has to use it when becoming overloaded. Some servers may wish server has to use it when becoming overloaded. Some servers may wish
to simply refuse the connection. to simply refuse the connection.
7.5.5 504 Gateway Timeout 7.5.5 504 Gateway Timeout
The server, while acting as a gateway, did not receive a timely The server, while acting as a gateway, did not receive a timely
response from the server (e.g., a location server) it accessed in response from the server (e.g., a location server) it accessed in
attempting to complete the request. attempting to complete the request.
7.5.6 505 Version Not Supported 7.5.6 505 Version Not Supported
The server does not support, or refuses to support, the SIP protocol
version that was used in the request message. The server is version that was used in the request message. The server is
indicating that it is unable or unwilling to complete the request indicating that it is unable or unwilling to complete the request
using the same major version as the client, other than with this using the same major version as the client, other than with this
error message. The response SHOULD contain an entity describing why error message. The response SHOULD contain an entity describing why
that version is not supported and what other protocols are supported that version is not supported and what other protocols are supported
by that server. by that server.
7.6 Global Failures 6xx 7.6 Global Failures 6xx
6xx responses indicate that a server has definitive information about 6xx responses indicate that a server has definitive information about
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7.6.4 606 Not Acceptable 7.6.4 606 Not Acceptable
The user's agent was contacted successfully but some aspects of the The user's agent was contacted successfully but some aspects of the
session description such as the requested media, bandwidth, or session description such as the requested media, bandwidth, or
addressing style were not acceptable. addressing style were not acceptable.
A 606 (Not Acceptable) response means that the user wishes to A 606 (Not Acceptable) response means that the user wishes to
communicate, but cannot adequately support the session described. The communicate, but cannot adequately support the session described. The
606 (Not Acceptable) response MAY contain a list of reasons in a 606 (Not Acceptable) response MAY contain a list of reasons in a
Warning header describing why the session described cannot be Warning header or headers describing why the session described cannot
supported. These reasons can be one or more of: be supported. Reasons are listed in Section 6.41. It is hoped that
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
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 negotiation will not frequently be needed, and when a new user is
being invited to join an already existing conference, negotiation may being invited to join an already existing conference, negotiation may
not be possible. It is up to the invitation initiator to decide not be possible. It is up to the invitation initiator to decide
whether or not to act on a 606 (Not Acceptable) response. whether or not to act on a 606 (Not Acceptable) response.
8 SIP Message Body 8 SIP Message Body
8.1 Body Inclusion 8.1 Body Inclusion
For a request message, the presence of a body is signaled by the For a request message, the presence of a body is signaled by the
inclusion of a Content-Length header. Only ACK, INVITE, OPTIONS inclusion of a Content-Length header. Only ACK, INVITE, OPTIONS
and REGISTER requests may contain message bodies. For ACK, INVITE and REGISTER requests may contain message bodies. For ACK, INVITE
and OPTIONS, the message body is always a session description. The and OPTIONS, the message body is always a session description. The
use of message bodies for REGISTER requests is for further study. use of message bodies for REGISTER requests is for further study.
For response messages, whether or not a body is included is dependent For response messages, whether or not a body is included is dependent
on both the request method and the response message's response code. on both the request method and the response message's response code.
All responses MAY include a body, although it may be of zero length. All responses MAY include a body, although it may be of zero length.
Message bodies for 1xx responses contain advisory information about Message bodies for 1xx responses contain advisory information about
the progress of the request, 2xx responses contain session the progress of the request. 2xx responses contain session
descriptions; for responses with status 300 or greater, the session descriptions. For responses with status 300 or greater, the messaage
body MAY contain additional, human-readable information about the body MAY contain additional, human-readable information about the
reasons for failure. It is RECOMMENDED that information in 1xx and reasons for failure. It is RECOMMENDED that information in 1xx and
300 and greater responses be of type text/plain or text/html 300 and greater responses be of type text/plain or text/html
8.2 Message Body Type 8.2 Message Body Type
The Internet media type of the message body MUST be given by the The Internet media type of the message body MUST be given by the
Content-Type header field, If the body has undergone any encoding Content-Type header field, If the body has undergone any encoding
(such as compression) then this MUST be indicated by the Content- (such as compression) then this MUST be indicated by the Content-
Encoding header field, otherwise Content-Encoding MUST be omitted. Encoding header field, otherwise Content-Encoding MUST be omitted. If
applicable, the character set of the message body is indicated as
If applicable, the character set of the message body is indicated as
part of the Content-Type header-field value. part of the Content-Type header-field value.
8.3 Message Body Length 8.3 Message Body Length
The body length in bytes MUST be given by the Content-Length header The body length in bytes MUST be given by the Content-Length header
field. If no body is present in a message, then the Content-Length field. If no body is present in a message, then the Content-Length
header MUST set to zero. If a server receives a message without header MUST set to zero. If a server receives a message without
Content-Length, it MUST assume it to be zero.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP. The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
(Note: The chunked encoding modifies the body of a message in order (Note: The chunked encoding modifies the body of a message in order
to transfer it as a series of chunks, each with its own size to transfer it as a series of chunks, each with its own size
indicator.) indicator.)
9 Compact Form 9 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
response is larger than the MTU. To reduce this problem, a more response is larger than the MTU. To reduce this problem, a more
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c Content-Type c Content-Type
e Content-Encoding e Content-Encoding
f From f From
i Call-ID i Call-ID
l Content-Length l Content-Length
m Location from "moved" m Location from "moved"
s Subject s Subject
t To t To
v Via v Via
Thus the header in section 14.2 could also be written: Thus, the header in section 14.2 could also be written:
INVITE schooler@vlsi.caltech.edu SIP/2.0 INVITE sip:schooler@vlsi.caltech.edu SIP/2.0
v:SIP/2.0/UDP 239.128.16.254 16 v:SIP/2.0/UDP 131.215.131.131;maddr=239.128.16.254;ttl=16
v:SIP/2.0/UDP 131.215.131.131
v:SIP/2.0/UDP 128.16.64.19 v:SIP/2.0/UDP 128.16.64.19
f:mjh@isi.edu f:sip:mjh@isi.edu
t:schooler@cs.caltech.edu t:sip:schooler@cs.caltech.edu
i:62729-27@128.16.64.19 i:62729-27@128.16.64.19
c:application/sdp c:application/sdp
CSeq: 4711 INVITE
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
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
Mixing short field names and long field names is allowed, but not
recommended. Servers MUST accept both short and long field names for
requests. Proxies MUST NOT translate a request between short and long requests. Proxies MUST NOT translate a request between short and long
forms if authentication fields are present. forms if authentication fields are present.
10 SIP Transport 10 SIP Transport
10.1 General Remarks 10.1 General Remarks
SIP is defined so it can use either UDP (unicast or multicast) or TCP SIP is defined so it can use either UDP (unicast or multicast) or TCP
as a transport protocol; it provides its own reliability mechanism. as a transport protocol; it provides its own reliability mechanism.
10.1.1 Requests 10.1.1 Requests
Stateful proxies mark outgoing requests with the branch parameter in
the Via header.
Servers ignore isomorphic requests, but retransmit the appropriate Servers ignore isomorphic requests, but retransmit the appropriate
response. (SIP requests are said to be idempotent , i.e., receiving response. (SIP requests are said to be idempotent , i.e., receiving
more than one copy of a request does not change the server state.) more than one copy of a request does not change the server state.)
If a stateful proxy, user agent or redirect server cannot respond to
a request with a final response within 200 ms, it MUST issue a
provisional (1xx) response as soon as possible. Stateless proxies
MUST NOT issue provisional responses on their own.
After receiving a CANCEL request from an upstream client, a stateful After receiving a CANCEL request from an upstream client, a stateful
proxy server SHOULD send a CANCEL on all branches where it has not proxy server SHOULD send a CANCEL on all branches where it has not
yet received a final response. yet received a final response.
If the To header user and host information does not match an address
supported by the server, the server returns a 404 (Not Found) error
response. Otherwise, it searches for the Call-ID value.
If the Call-ID was found, it compares the tag value of To with the
user's tag and rejects the request if the two do not match. If the
From header, including any tag value, matches the value for an
existing call leg, the server compares the CSeq header value. If less
than or equal to the current sequence number, the request is a
retransmission. Otherwise, it is a new request. If the From header
does not match an existing call leg, a new call leg is created.
If the Call-ID was not found, a new call leg is created, with
entries for the To, From and Call-ID headers. In this case, the
To header should not have contained a tag. The server returns a
response containing the same To value, but with a unique tag added.
The tag MAY be omitted if the To refers to a fully qualified host
name.
10.1.2 Responses 10.1.2 Responses
A server MAY issue one or more provisional responses at any time A server MAY issue one or more provisional responses at any time
before sending a final response. before sending a final response. If a stateful proxy, user agent
server, redirect server or registrar cannot respond to a request with
a final response within 200 ms, it MUST issue a provisional (1xx)
response as soon as possible. Stateless proxies MUST NOT issue
provisional responses on their own.
Responses are mapped to requests by the matching To, From, Call-ID, Responses are mapped to requests by the matching To, From, Call-ID,
CSeq headers and the branch parameter of the first Via header. CSeq headers and the branch parameter of the first Via header.
Responses terminate request retransmissions even if they have Via Responses terminate request retransmissions even if they have Via
headers that cause them to be delivered to an upstream client. headers that cause them to be delivered to an upstream client.
A stateful proxy may receive a response that it does not have state A stateful proxy may receive a response that it does not have state
for, that is, where it has no a record of an isomorphic request. If for, that is, where it has no a record of an isomorphic request. If
the Via header field indicates that the upstream server used TCP, the the Via header field indicates that the upstream server used TCP, the
proxy actively opens a TCP connection to that address. Thus, proxies proxy actively opens a TCP connection to that address. Thus, proxies
have to be prepared to receive responses on the incoming side of have to be prepared to receive responses on the incoming side of
passive TCP connections, even though most responses will arrive on passive TCP connections, even though most responses will arrive on
the incoming side of an active connection. (An active connection is a the incoming side of an active connection. (An active connection is a
TCP connection initiated by the proxy, a passive connection is one TCP connection initiated by the proxy, a passive connection is one
accepted by the proxy, but initiated by another entity.) accepted by the proxy, but initiated by another entity.)
100 responses are not forwarded, other 1xx responses MAY be 100 responses are not forwarded, other 1xx responses MAY be
forwarded, possibly after the server eliminates responses with status forwarded, possibly after the server eliminates responses with status
codes that had already been sent earlier. 2xx responses are forwarded codes that had already been sent earlier. 2xx responses are forwarded
according to the Via header. Once a stateful proxy has received a
2xx response, it MUST NOT forward non-2xx final responses. Responses
with status 300 and higher are retransmitted by each stateful proxy with status 300 and higher are retransmitted by each stateful proxy
until the next upstream proxy sends an ACK (see below for timing until the next upstream proxy sends an ACK (see below for timing
details) or CANCEL. details) or CANCEL.
A stateful proxy can remove state for a call attempt and close any A stateful proxy can remove state for a call attempt and close any
connections 20 seconds after receiving the first final response. connections 20 seconds after receiving the first final response.
The 20 second window is given by the maximum retransmission The 20 second window is given by the maximum retransmission
duration of 200 responses (10 times T4), in case the ACK duration of 200 responses (10 times T4), in case the ACK
is lost somewhere on the way to the called user agent or is lost somewhere on the way to the called user agent or
the next stateful proxy. the next stateful proxy.
10.2 Unicast UDP 10.2 Source Addresses, Destination Addresses and Connections
10.2.1 Unicast UDP
UDP packets MUST have a source address that is valid as a destination UDP packets MUST have a source address that is valid as a destination
for requests and responses. Responses are returned to the address for requests and responses. Responses are returned to the address
listed in the Via header field (Section 6.40), not the source listed in the Via header field (Section 6.40), not the source
address of the request. address of the request.
10.3 Multicast UDP 10.2.2 Multicast UDP
Requests may be multicast. Multicast requests SHOULD have a time-to- Requests may be multicast; multicast requests likely feature a host-
independent Request-URI. Multicast requests SHOULD have a time-to-
live value of no greater than one, i.e., be restricted to the local live value of no greater than one, i.e., be restricted to the local
network. network.
If the request was received via multicast, the response is also A client receiving a multicast query does not have to check whether
returned via multicast. The server delays its response by a random the host part of the Request-URI matches its own host or domain
interval between zero and one second. Servers do not return 404 (Not name. If the request was received via multicast, the response is also
Found) responses and SHOULD suppress responses if they hear a lower- returned via multicast. Responses to multicast requests are multicast
numbered or 6xx response from another group member prior to sending. with the same TTL as the request, where the TTL is derived from the
Servers do not respond to CANCEL requests received via multicast. ttl parameter in the Via header (Section 6.40).
10.4 BYE, CANCEL, OPTIONS To avoid response implosion, servers MUST NOT answer multicast
requests with a status code other than 2xx or 6xx. Servers only
return 6xx responses if the To represents a single individual rather
than a group of people. The server delays its response by a random
interval between zero and one second. Servers SHOULD suppress
responses if they hear a lower-numbered or 6xx response from another
group member prior to sending. Servers do not respond to CANCEL
requests received via multicast to avoid request implosion.
A SIP client SHOULD retransmit a BYE, CANCEL, or OPTIONS request 10.3 TCP
periodically with timer T1 until it receives a response, or until it
has reached a set limit on the number of retransmissions. If the A single TCP connection can serve one or more SIP transactions. A
response is provisional, the client continues to retransmit the transaction contains zero or more provisional responses followed by
request, albeit less frequently, using timer T2. The default values one or more final responses. (Typically, transactions contain exactly
of timer T1 and T2 are 1 and 5 seconds, respectively. The default one final response, but there are exceptional circumstances, where,
retransmit limit is 20 times. After the server sends a final for example, multiple 200 responses may be generated.)
response, it cannot be sure the client has received the response, and
thus SHOULD cache the results for at least 100 seconds to avoid The client MAY close the connection at any time, but SHOULD keep the
having to, for example, contact the user or user location server connection open at least until the first final response arrives. The
again upon receiving a retransmission. server SHOULD NOT close the TCP connection until it has sent its
final response, at which point it MAY close the TCP connection if it
wishes to. However, normally it is the client's responsibility to
close the connection.
If the server leaves the connection open, and if the client so
desires it may re-use the connection for further SIP requests or for
requests from the same family of protocols (such as HTTP or stream
control commands).
If a client closes a connection or the connection is reset (e.g.,
because the client has crashed and rebooted), the server treats this
as equivalent to having received a CANCEL request.
If a server needs to return a response to a client and no longer has
a connection open to that client, it MAY open a connection to the
address listed in the Via header. Thus, a proxy or user agent MUST
be prepared to receive both requests and responses on a "passive"
connection.
10.4 Reliability for BYE, CANCEL, OPTIONS, REGISTER Requests
10.4.1 UDP
A SIP client using UDP SHOULD retransmit a BYE, CANCEL, OPTIONS, or
REGISTER request periodically with timer T1 until it receives a
response, or until it has reached a set limit on the number of
retransmissions. If the response is provisional, the client continues
to retransmit the request, albeit less frequently, using timer T2.
The default values of timer T1 and T2 are 1 and 5 seconds,
respectively. The default retransmit limit is 20 times. After the
server sends a final response, it cannot be sure the client has
received the response, and thus SHOULD cache the results for at least
100 seconds to avoid having to, for example, contact the user or
location server again upon receiving a retransmission.
Each server in a proxy chain generates its own final response to a Each server in a proxy chain generates its own final response to a
CANCEL request. BYE and OPTIONS final responses are generated by CANCEL request; BYE and OPTIONS final responses are generated by
redirect and user agent servers. redirect and user agent servers; REGISTER final responses are
generated by registrars. Note that responses to these commands are
not acknowledged via ACK.
The value of the initial retransmission timer is smaller The value of the initial retransmission timer is smaller
than that that for TCP since it is expected that network than that that for TCP since it is expected that network
paths suitable for interactive communications have round- paths suitable for interactive communications have round-
trip times smaller than 1.5 seconds. To avoid flooding the trip times smaller than 1 second. To avoid flooding the
network with packets every second even if the destination network with packets every second even if the destination
network is unreachable, the retransmission count has to be network is unreachable, the retransmission count has to be
bounded. Given that most transactions should consist of one bounded. Given that most transactions should consist of one
request and a few responses, round-trip time estimation request and a few responses, round-trip time estimation is
seems less than helpful. If RTT estimation is desired to not likely to be very useful. If RTT estimation is desired
more quickly discover a missing final response, each to more quickly discover a missing final response, each
request retransmission needs to be labeled with its own request retransmission needs to be labeled with its own
Timestamp (Section 6.36), returned in the response. The Timestamp (Section 6.36), returned in the response. The
server caches the result until it can be sure that the server caches the result until it can be sure that the
client will not retransmit the same request again. client will not retransmit the same request again.
10.5 REGISTER 10.4.2 TCP
A client MAY repeat its registration attempts at intervals of 2, 4,
8, ..., 512, 512, ... seconds if it receives no response.
Retransmitting REGISTER indefinitely ensures that a user Clients using TCP do not need to retransmit requests.
will eventually be able to register after a registrar
recovers from a crash. The period is chosen so that even on
a large LAN, there will not be more than about one
REGISTER request per second.
10.6 ACK 10.5 Reliability for ACK Requests
The ACK request does not generate responses. It is only The ACK request does not generate responses. It is only
retransmitted when a response to an INVITE request arrives. retransmitted when a response to an INVITE request arrives. This
behavior is independent of the transport protocol.
10.7 INVITE 10.6 Reliability for INVITE Requests
Special considerations apply for the INVITE method. Special considerations apply for the INVITE method.
1. After receiving an invitation, considerable time may elapse 1. After receiving an invitation, considerable time may elapse
before the server can determine the outcome. For example, before the server can determine the outcome. For example,
the called party may be "rung" or extensive searches may be the called party may be "rung" or extensive searches may be
performed, so delays between the request and a definitive performed, so delays between the request and a definitive
response can reach several tens of seconds. If either response can reach several tens of seconds. If either
caller or callee are automated servers not directly caller or callee are automated servers not directly
controlled by a human being, a call attempt may be controlled by a human being, a call attempt may be
unbounded in time. unbounded in time.
2. If a telephony user interface is modeled or if we need to 2. If a telephony user interface is modeled or if we need to
interface to the PSTN, the caller's user interface will interface to the PSTN, the caller's user interface will
provide "ringback", a signal that the callee is being provide "ringback", a signal that the callee is being
alerted. (The status response 180 (Ringing) may be used to alerted. (The status response 180 (Ringing) may be used to
initiate ringback.) Once the callee picks up, the caller initiate ringback.) Once the callee picks up, the caller
needs to know so that it can enable the voice path and stop 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 caller will continue to hear ringback while the callee
assumes that the call exists. assumes that the call exists.
3. The client has to be able to terminate an on-going request, 3. The client has to be able to terminate an on-going request,
e.g., because it is no longer willing to wait for the e.g., because it is no longer willing to wait for the
connection or search to succeed. The server will have to connection or search to succeed. The server will have to
wait several round-trip times to interpret the lack of wait several round-trip times to interpret the lack of
request retransmissions as the end of a call. If the call request retransmissions as the end of a call. If the call
succeeds shortly after the caller has given up, the callee succeeds shortly after the caller has given up, the callee
will "pick up the phone" and not be "connected". will "pick up the phone" and not be "connected".
A SIP client SHOULD retransmits a SIP INVITE request periodically 10.6.1 UDP
with timer T1 until it receives a response, or until it has reached a For UDP, A SIP client SHOULD retransmits a SIP INVITE request
set limit on the number of retransmissions. If the response is periodically with timer T1 until it receives a response. If the
provisional, the client continues to retransmit the request, albeit client receives no response, it ceases retransmission after 20
less frequently, using timer T3. The default values of timer T1 and attempts. If the response is provisional, the client continues to
T3 are 1 and 30 seconds, respectively. The default retransmit limit retransmit the request, albeit less frequently, using timer T3. The
is 20. default values of timer T1 and T3 are 1 and 30 seconds, respectively.
The value of T3 was chosen so that for most normal phone The value of T3 was chosen so that for most normal phone
calls, only one INVITE request will be issued. Typically, calls, only one INVITE request will be issued. Typically,
a phone switches to an answering machine or voice mail a phone switches to an answering machine or voice mail
after about 20--22 seconds. after about 20--22 seconds. The number of retransmissions
after receiving a provisional response is unlimited to
Upon receiving a 2xx final response, the client sends an ACK to the allow call queueing. Clients may limit the number of
address listed in the Location header field contained in the invitations sent for each call attempt.
response. If the response did not contain a Location header, the
client uses the same To header field as for the INVITE request and
sends the ACK to the same destination as the original INVITE
request.
ACKs for final responses other than 2xx are sent to the source of the For 2xx final responses, only the user agent client generates an
response. ACK. If the response contained a Location header, the ACK is sent
to the address listed in that Location header field. If the response
did not contain a Location header, the client uses the same To
header field and Request-URI as for the INVITE request and sends the
ACK to the same destination as the original INVITE request. ACKs
for final responses other than 2xx are sent to the source of the
response by each client.
The server retransmits the final response at intervals of T4 (default The server retransmits the final response at intervals of T4 (default
value of T4 = 2 seconds) until it receives an ACK request for the value of T4 = 2 seconds) until it receives an ACK request for the
same Call-ID and CSeq from the same From source or until it has same Call-ID and CSeq from the same From source or until it has
retransmitted the final response 10 times. The ACK request MUSTNOT be retransmitted the final response 10 times. The ACK request MUST NOT
acknowledged to prevent a response- ACK feedback loop. be acknowledged to prevent a response- ACK feedback loop.
Fig. 8 and 9 show the client and server state diagram for Fig. 11 and 12 show the client and server state diagram for
invitations. invitations.
Using the mechanism in Sec. 10.4 does not work well for the The mechanism in Sec. 10.4 would not work well for INVITE
long delays between INVITE and a final response. If the because of the long delays between INVITE and a final
200 response gets lost, the callee would believe the call response. If the 200 response were to get lost, the callee
to exist, but the voice path would be dead since the caller would believe the call to exist, but the voice path would
be dead since the caller does not know that the callee has
picked up. Thus, the INVITE retransmission interval would
have to be on the order of a second or two to limit the
duration of this state confusion. Retransmitting the
response a fixed number of times increases the probability
of success, but at the cost of significantly higher
processing and network load.
+===========+ +===========+
| Initial | | Initial |
+===========+ +===========+
| |
| |
| - | -
| ------ | ------
| INVITE | INVITE
+------v v +------v v
T1 +-----------+ T1 +-----------+
skipping to change at page 66, line 43 skipping to change at page 77, line 44
+------v | v | +------v | v |
xxx +-----------+ | xxx +-----------+ |
--- | Completed |<-------+ --- | Completed |<-------+
ACK +-----------+ ACK +-----------+
+------| +------|
event event
------- -------
message message
Figure 8: State transition diagram of client for INVITE method Figure 11: State transition diagram of client for INVITE method
order of a second or two to limit the duration of this
state confusion.
Blindly retransmitting the response increases the probability of 10.6.2 TCP
success, but at the cost of significantly higher processing and
network load.
A client using TCP MUST NOT retransmit requests, but uses the same
algorithm as for UDP (Section 10.6.1) to retransmit responses until
+===========+ +===========+
| Initial |<-------------+ | Initial |<-------------+
+===========+ | +===========+ |
| | | |
| | | |
| INVITE | | INVITE |
| ------ | | ------ |
| 1xx | | 1xx |
+------v v | +------v v |
INVITE +-----------+ | INVITE +-----------+ |
skipping to change at page 67, line 45 skipping to change at page 78, line 44
ACK +-----------+ | ACK +-----------+ |
--- | Connected | | --- | Connected | |
- +-----------+ | - +-----------+ |
+------| | | +------| | |
+-----------------+ +-----------------+
event event
------- -------
message message
Figure 9: State transition diagram of server for INVITE method Figure 12: State transition diagram of server for INVITE method
A single TCP connection can serve one or more SIP transactions. A
transaction contains zero or more provisional responses followed by
one or more final responses. (Typically, transactions contain exactly
one final response, but there are exceptional circumstances, where,
for example, multiple 200 responses may be generated.)
status code of 300 or larger, the client sends an ACK. If the
response status code is 2xx and the client is a user agent client, it
sends an ACK. If the client is not a user agent, the response is
forwarded upstream.
The client MAY close the connection at any time. The server SHOULD
NOT close the TCP connection until it has sent its final response, at
which point it MAY close the TCP connection if it wishes to. However,
normally it is the client's responsibility to close the connection.
If the server leaves the connection open, and if the client so
desires it may re-use the connection for further SIP requests or for
requests from the same family of protocols (such as HTTP or stream
control commands).
If a server needs to return a response to a client and no longer has it receives an ACK. (An implementation can simply set T1 and T3 to
a connection open to that client, it MAY open a connection to the infinity and otherwise maintain the same state diagram.)
address listed in the Via header. Thus, a proxy MUST be prepared to It is necessary to retransmit 2xx responses as their
receive both requests and responses on a "passive" connection. reliability is assured end-to-end only. If the chain of
proxies has a UDP link in the middle, it could lose the
response, with no possibility of recovery. For simplicity,
we also retransmit non-2xx responses, although that is not
strictly necessary.
11 Behavior of SIP Servers 11 Behavior of SIP Servers
This section describes behavior of a SIP server in detail. Servers This section describes behavior of a SIP server in detail. Servers
can operate in proxy or redirect mode. Proxy servers can "fork" can operate in proxy or redirect mode. Proxy servers can "fork"
connections, i.e., a single incoming request spawns several outgoing connections, i.e., a single incoming request spawns several outgoing
(client) requests. (client) requests.
A proxy server always inserts a Via header field containing its own A proxy server always inserts a Via header field containing its own
address into those requests that are caused by an incoming request. address into those requests that are caused by an incoming request.
skipping to change at page 68, line 44 skipping to change at page 79, line 31
prevent loops, a server MUST check if its own address is already prevent loops, a server MUST check if its own address is already
contained in the Via header of the incoming request. contained in the Via header of the incoming request.
A proxy server MAY convert a version-x SIP request or response to a A proxy server MAY convert a version-x SIP request or response to a
version-y request or response, where x may be larger, smaller or version-y request or response, where x may be larger, smaller or
equal to y. equal to y.
This rule allows a proxy to serve as a go-between between This rule allows a proxy to serve as a go-between between
two servers that have no version of the protocol in common. two servers that have no version of the protocol in common.
We define an "A--B proxy" as a proxy that receives SIP requests over
transport protocol A and issues requests, acting as a SIP client,
using transport protocol B. If not stated explicitly, rules apply to
any combination of transport protocols. For conciseness, we only
describe behavior with UDP and TCP, but the same rules apply for any
unreliable datagram or reliable protocol, respectively.
The detailed connection behavior for UDP and TCP is described in
Section 10.
11.1 Redirect Server 11.1 Redirect Server
return a response that refuses or redirects the request. After
receiving an INVITE request, once the server has gathered the list A redirect server does not issue any SIP requests of its own. After
of alternative locations or has decided to refuse the call, it receiving a request, the server gathers the list of alternative
returns the final response of class 3xx or 6xx. This ends the SIP locations and returns a final response of class 3xx or it refuses the
transaction. The redirect server maintains transaction state for the request. For CANCEL requests, it may also return a 2xx response.
whole SIP transaction. It is up to the client to detect forwarding This response ends the SIP transaction. The redirect server maintains
loops between redirect servers. transaction state for the whole SIP transaction. It is up to the
client to detect forwarding loops between redirect servers.
11.2 User Agent Server 11.2 User Agent Server
User agent servers behave similarly to redirect servers, except that User agent servers behave similarly to redirect servers, except that
they may also accept a call with a response of class 2xx. they may also accept requests and return a response of class 2xx.
11.3 Stateless Proxy: Proxy Servers Issuing Single Unicast Requests 11.3 Stateless Proxy: Proxy Servers Issuing Single Unicast Requests
Proxies in this category issue at most a single unicast request for Proxies in this category issue at most a single unicast request for
each incoming SIP request, that is, they do not "fork" requests. each incoming SIP request, that is, they do not "fork" requests.
However, servers may choose to always operate in a mode that allows However, servers may choose to always operate in a mode that allows
issuing of several requests, as described in Section 11.4. issuing of several requests, as described in Section 11.4.
The server can forward the request and any responses. It does not The server can forward the request and any responses. It does not
have to maintain any state for the SIP transaction. Reliability is have to maintain any state for the SIP transaction. Reliability is
assured by the next redirect or stateful proxy server in the server assured by the next redirect or stateful proxy server in the server
chain. chain.
A proxy server SHOULD cache the result of any address translations A proxy server SHOULD cache the result of any address translations
and the response to speed forwarding of retransmissions. After the and the response to speed forwarding of retransmissions. After the
cache entry has been expired, the server cannot tell whether an cache entry has been expired, the server cannot tell whether an
incoming request is actually a retransmission of an older request. incoming request is actually a retransmission of an older request.
The server will treat it as a new request and commence another The server will treat it as a new request and commence another
search. search.
11.4 Proxy Server Issuing Several INVITE Requests 11.4 Proxy Server Issuing Several Requests
The server MUST respond to the request immediately with a 100 The server MUST respond to the request immediately with a 100
(Trying) response. (Trying) response.
All requests carry the same Call-ID. For unicast, each of the All outgoing requests carry the same Call-ID, To, From and CSeq
requests has a different (host-dependent) Request-URI. For headers as the request received. Each of the requests has a different
multicast, a single request is issued, likely with a host-independent (host-dependent) Request-URI.
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 MUST 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. Responses to multicast requests are multicast with the same TTL
as the request, where the TTL is derived from the ttl parameter in
the Via header (Section 6.40).
Successful responses to an INVITE request SHOULD contain a Location Successful responses to an INVITE request SHOULD contain a Location
header so that the following ACK or BYE bypasses the proxy search header so that the following ACK or BYE bypasses the proxy search
mechanism. If the proxy requires future requests to be routed through
it, it adds a Record-Route header to the request (Section 6.29).
The following pseudo-code describes the behavior of a proxy server The following pseudo-code describes the behavior of a proxy server
issuing several requests in response to an incoming INVITE request. issuing several requests in response to an incoming INVITE request.
The function request(r, a, b) sends a SIP request of type r to The function request(r, a, b) sends a SIP request of type r to
address a, with branch id b. await_response() waits until a response address a, with branch id b. await_response() waits until a response
is received and returns the response. close(a) closes the TCP is received and returns the response. close(a) closes the TCP
connection to client with address a. response(s, l, L) sends a connection to client with address a. response(r) sends a response to
response to the client with status s and list of locations L, with l the client. ismulticast() returns 1 if the location is a multicast
entries. ismulticast() returns 1 if the location is a multicast
address and zero otherwise. The variable timeleft indicates the address and zero otherwise. The variable timeleft indicates the
amount of time left until the maximum response time has expired. The amount of time left until the maximum response time has expired. The
variable recurse indicates whether the server will recursively try variable recurse indicates whether the server will recursively try
addresses returned through a 3xx response. A server MAY decide to addresses returned through a 3xx response. A server MAY decide to
recursively try only certain addresses, e.g., those which are within recursively try only certain addresses, e.g., those which are within
the same domain as the proxy server. Thus, an initial multicast the same domain as the proxy server. Thus, an initial multicast
request may trigger additional unicast requests. request may trigger additional unicast requests.
/* request type */ /* request type */
typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method; typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method;
skipping to change at page 70, line 34 skipping to change at page 81, line 12
{ {
struct { struct {
address_t address; /* address */ address_t address; /* address */
int branch; /* branch id */ int branch; /* branch id */
int done; /* has responded */ int done; /* has responded */
} outgoing[]; } outgoing[];
int done[]; /* address has responded */ int done[]; /* address has responded */
char *location[]; /* list of locations */ char *location[]; /* list of locations */
int heard = 0; /* number of sites heard from */ int heard = 0; /* number of sites heard from */
int class; /* class of status code */ int class; /* class of status code */
int best = 1000; /* best response so far */
int timeleft = 120; /* sample timeout value */ int timeleft = 120; /* sample timeout value */
int loc = 0; /* number of locations */ int loc = 0; /* number of locations */
struct { /* response */ struct { /* response */
int status; /* response status; -2: BYE; -1: CANCEL */ int status; /* response: BYE=-2; CANCEL=-1 */
int locations; /* number of redirect locations */ int locations; /* number of redirect locations */
char *location[]; /* redirect locations */ char *location[]; /* redirect locations */
address_t a; /* address of respondent */ address_t a; /* address of respondent */
int branch; /* branch identifier */ int branch; /* branch identifier */
} r; } r, best; /* response, best response */
int i; int i;
best.status = 1000;
for (i = 0; i < N; i++) { for (i = 0; i < N; i++) {
request(R, address[i], i); request(R, address[i], i);
outgoing[i].done = 0; outgoing[i].done = 0;
outgoing[i].branch = i; outgoing[i].branch = i;
} }
while (timeleft > 0 && heard < N) { while (timeleft > 0 && heard < N) {
r = await_response(); r = await_response();
class = r.status / 100;
if (r.status < 0) { if (r.status < 0) {
break; break;
} }
/* If final response, mark branch as done. */ /* If final response, mark branch as done. */
if (class >= 2) { if (class >= 2) {
heard++; heard++;
for (i = 0; i < N; i++) { for (i = 0; i < N; i++) {
if (r.branch == outgoing[i].branch) { if (r.branch == outgoing[i].branch) {
outgoing[i].done = 1; outgoing[i].done = 1;
break; break;
} }
} }
} }
if (class == 2) { if (class == 2) {
best = r.status; best = r;
break; break;
} }
else if (class == 3) { else if (class == 3) {
/* A server may optionally recurse. The server MUST check whether /* A server may optionally recurse. The server MUST check
* it has tried this location before and whether the location is * whether it has tried this location before and whether the
* part of the Via path of the incoming request. This check is * location is part of the Via path of the incoming request.
* omitted here for brevity. Multicast locations MUST NOT be * This check is omitted here for brevity. Multicast locations
* returned to the client if the server is not recursing. * MUST NOT be returned to the client if the server is not
* recursing.
*/ */
if (recurse) { if (recurse) {
multicast = 0; multicast = 0;
N += r.locations; N += r.locations;
for (i = 0; i < r.locations; i++) { for (i = 0; i < r.locations; i++) {
request(R, r.location[i]); request(R, r.location[i]);
} }
} else if (!ismulticast(r.location)) { } else if (!ismulticast(r.location)) {
locations[loc++] = r.location; best = r;
best = r.status;
} }
request(ACK, r.a, r.branch); if (R == INVITE} request(ACK, r.a, r.branch);
} }
else if (class == 4) { else if (class == 4) {
request(ACK, r.a, r.branch); if (R == INVITE} request(ACK, r.a, r.branch);
if (best >= 400) best = r.status; if (best.status >= 400) best = r;
} }
else if (class == 5) { else if (class == 5) {
request(ACK, r.a, r.branch); if (R == INVITE} request(ACK, r.a, r.branch);
if (best >= 500) best = r.status; if (best.status >= 500) best = r;
} }
else if (class == 6) { else if (class == 6) {
request(ACK, r.a, r.branch); if (R == INVITE} request(ACK, r.a, r.branch);
best = r.status; best = r;
break; break;
} }
}
/* CANCEL */ /* CANCEL */
if (r.status == -1) { if (r.status == -1) {
response(200, loc, 0); best.status = 200;
response(best);
} }
/* BYE */ /* BYE */
else if (r.status == -2) { else if (r.status == -2) {
for (i = 0; i < N; i++) { for (i = 0; i < N; i++) {
request(BYE, address[i], i); request(BYE, address[i], i);
} }
} }
/* INVITE */
else { else {
/* We haven't heard anything useful from anybody. */ /* We haven't heard anything useful from anybody. */
if (best == 1000) { if (best.status == 1000) {
best = 404; best.status = 404;
} }
if (best/100 != 3) loc = 0; if (best.status/100 != 3) loc = 0;
response(best, loc, locations); response(best);
} }
/* /*
* If complete or CANCELed, close the other pending transactions by * If complete or CANCELed, close the other pending transactions by
* sending CANCEL. * sending CANCEL.
*/ */
if (r.status > 0 || r.status == -1) { if (r.status > 0 || r.status == -1) {
for (i = 0; i < N; i++) { for (i = 0; i < N; i++) {
if (!outgoing[i].done) { if (!outgoing[i].done) {
request(CANCEL, address[i], outgoing[i].branch); request(CANCEL, address[i], outgoing[i].branch);
if (tcp) close(a); if (tcp) close(a);
} }
} }
} }
} }
Responses are processed as follows. The process completes when all Responses are processed as follows. The process completes (and state
requests have been answered by final status responses (for unicast) can be freed) when all requests have been answered by final status
or 60 seconds have elapsed (for multicast). A proxy MAY send a responses (for unicast) or 60 seconds have elapsed (for multicast). A
CANCEL to all branches and return a 408 (Timeout) to the client after proxy MAY send a CANCEL to all branches and return a 408 (Timeout)
120 seconds or more. to the client after 60 seconds or more.
1xx: The proxy MAY forward the response upstream towards the client. 1xx: The proxy MAY forward the response upstream towards the client.
2xx: The proxy MUST forward the response upstream towards the client, 2xx: The proxy MUST forward the response upstream towards the client,
without sending an ACK downstream. without sending an ACK downstream. After receiving a 2xx, the
server SHOULD terminate all other pending requests by sending a
CANCEL request and closing the TCP connection, if applicable.
(Terminating pending requests is advisable as searches consume
resources. Also, INVITE requests may "ring" on a number of
workstations if the callee is currently logged in more than
once.)
3xx: The proxy MUST send an ACK and MAY recurse on the listed 3xx: The proxy MUST send an ACK and MAY recurse on the listed
Location addresses. Otherwise, the locations in the response are Location addresses. Otherwise, the lowest-numbered response is
added to separate lists for 300, 301 and 302 responses returned if there were no 2xx responses.
maintained by the proxy. The lowest-numbered 300 response is
returned to the client on completion. Location lists are not merged as that would prevent
forwarding of authenticated responses. Also, some responses
may have message bodies, so that merging is not feasible.
4xx, 5xx: The proxy MUST send an ACK and remember the response if it
has a lower status code than any previous 4xx and 5xx responses. has a lower status code than any previous 4xx and 5xx responses.
On completion, the lowest-numbered response is returned if there On completion, the lowest-numbered response is returned if there
were no 2xx or 3xx responses. were no 2xx or 3xx responses.
6xx: The proxy MUST forward the response to the client and send an 6xx: The proxy MUST forward the response to the client and send an
ACK. Other pending requests SHOULD be terminated with CANCEL. ACK. Other pending requests SHOULD be terminated with CANCEL as
described for 2xx responses.
The proxy server SHOULD maintain state until all responses have been
received or for 60 seconds if the request was multicast.
After receiving a 2xx or 6xx response, the server SHOULD terminate
all other pending requests by sending a CANCEL request and closing
the TCP connection, if applicable. (Terminating pending requests is
advisable as searches consume resources. Also, INVITE requests may
"ring" on a number of workstations if the callee is currently logged
in more than once.)
When operating in this mode, a proxy server MUST ignore any responses When operating in this mode, a proxy server MUST ignore any responses
received for Call-IDs for which it does not have a pending received for Call-IDs for which it does not have a pending
transaction. (If server were to forward responses not belonging to a transaction. (If server were to forward responses not belonging to a
current transaction using the Via field, the requesting client would current transaction using the Via field, the requesting client would
get confused if it has just issued another request using the same get confused if it has just issued another request using the same
Call-ID.) Call-ID.)
If a proxy server receives a BYE request for a pending search, the If a proxy server receives a BYE request for a pending search, the
proxy MUST terminate all pending requests by sending a BYE request. proxy MUST terminate all pending requests by sending a BYE request.
11.5 Proxy Server Issuing Several ACK and BYE Requests Special considerations apply for choosing forwarding destinations for
ACK and BYE requests. In most cases, these requests will bypass
In most cases, ACK and BYE requests will bypass proxies and reach proxies and reach the desired party directly, keeping proxies from
the desired party directly, keeping proxies from having to make having to make forwarding decisions.
forwarding decisions.
User agent clients respond to ACK and BYE requests with unknown User agent clients respond to ACK and BYE requests with unknown
Call-ID with status code 481 (Invalid Call-ID). Call-ID with status code 481 (Invalid Call-ID).
A proxy MAY maintain call state for a period of its choosing. If a A proxy MAY maintain call state for a period of its choosing. If a
proxy still has list of destinations that it forwarded the last proxy still has list of destinations that it forwarded the last
INVITE to, it SHOULD direct ACK requests only to those downstream INVITE to, it SHOULD direct ACK requests only to those downstream
servers. It SHOULD direct BYE to only those servers that had servers. It SHOULD direct BYE to only those servers that had
previously responded with 2xx or have not yet responded to the last previously responded with 2xx or have not yet responded to the last
INVITE. INVITE. If the proxy has no call state for a particular Call-ID and
To destination, it forks the request as it would for an INVITE
If the proxy has no call state for a particular Call-ID and To request.
destination, it forwards the request to all downstream servers.
12 Security Considerations 12 Security Considerations
12.1 Confidentiality and Privacy: Encryption 12.1 Confidentiality and Privacy: Encryption
12.1.1 SIP Requests and Responses 12.1.1 End-to-End Encryption
SIP requests and responses can contain sensitive information about
the communication patterns and communication content of individuals the communication patterns and communication content of individuals
and thus should be protected against eavesdropping. The SIP message and thus should be protected against eavesdropping. The SIP message
body may also contain encryption keys for the session itself. body may also contain encryption keys for the session itself.
SIP supports three complementary forms of encryption to protect SIP supports three complementary forms of encryption to protect
privacy: privacy:
o End-to-end encryption of the SIP message body and certain o End-to-end encryption of the SIP message body and certain
sensitive header fields; sensitive header fields;
o hop-by-hop encryption to prevent eavesdropping that tracks who o hop-by-hop encryption to prevent eavesdropping that tracks who
is calling whom; is calling whom;
o hop-by-hop encryption of Via fields to hide the route a o hop-by-hop encryption of Via fields to hide the route a
request has taken. request has taken.
Not all of the SIP request can be encrypted end-to-end because header Not all of the SIP request or response can be encrypted end-to-end
fields such as To and Via need to be visible to proxies so that the because header fields such as To and Via need to be visible to
SIP request can be routed correctly. Hop-by-hop encryption encrypts proxies so that the SIP request can be routed correctly. Hop-by-hop
the entire SIP request or response on the wire (the request may encryption encrypts the entire SIP request or response on the wire
already have been end-to-end encrypted) so that packet sniffers or (the request may already have been end-to-end encrypted) so that
other eavesdroppers cannot see who is calling whom. Note that proxies packet sniffers or other eavesdroppers cannot see who is calling
can still see who is calling whom, and this information may also be whom. Note that proxies can still see who is calling whom, and this
deducible by performing a network traffic analysis, so this provides information may also be deducible by performing a network traffic
a very limited but still worthwhile degree of protection. analysis, so this provides a very limited but still worthwhile degree
of protection.
SIP Via fields are used to route a response back along the path SIP Via fields are used to route a response back along the path
taken by the request and to prevent infinite request loops. However, taken by the request and to prevent infinite request loops. However,
the information given by them may also provide useful information to the information given by them may also provide useful information to
an attacker. Section 6.21 describes how a sender can request that Via an attacker. Section 6.21 describes how a sender can request that Via
fields be encrypted by cooperating proxies without compromising the fields be encrypted by cooperating proxies without compromising the
purpose of the Via field. purpose of the Via field.
12.2 End-to-End Encryption
End-to-end encryption relies on keys shared by the two user agents End-to-end encryption relies on keys shared by the two user agents
involved in the request. Typically, the message is sent encrypted involved in the request. Typically, the message is sent encrypted
with the public key of the recipient, so that only that recipient can with the public key of the recipient, so that only that recipient can
read the message. SIP does not limit the security mechanisms that may read the message. SIP does not limit the security mechanisms that may
be used, but all implementations SHOULD support PGP-based encryption. be used, but all implementations SHOULD support PGP-based encryption.
A SIP request (or response) is end-to-end encrypted by splitting the A SIP request (or response) is end-to-end encrypted by splitting the
message to be sent into a part to be encrypted and a short header message to be sent into a part to be encrypted and a short header
that will remain in the clear. Some parts of the SIP message, namely that will remain in the clear. Some parts of the SIP message, namely
the request line, the response line and certain header fields marked the request line, the response line and certain header fields marked
with "n" in the "enc." column in Table 3 need to be read and returned with "n" in the "enc." column in Table 4 need to be read and returned
by proxies and thus MUST NOT be encrypted end-to-end. Possibly by proxies and thus MUST NOT be encrypted end-to-end. Possibly
sensitive information that needs to be made available as plaintext sensitive information that needs to be made available as plaintext
include destination address ( To) and the forwarding path ( Via) of include destination address ( To) and the forwarding path ( Via) of
the call. The Authorization header MUST remain in the clear if it the call. The Authorization header MUST remain in the clear if it
contains a digital signature as the signature is generated after contains a digital signature as the signature is generated after
encryption, but MAY be encrypted if it contains "basic" or "digest"
authentication. The From header field SHOULD normally remain in the
clear, but MAY be encrypted if required, in which case some proxies clear, but MAY be encrypted if required, in which case some proxies
MAY return a 401 (Unauthorized) status if they require a From field. MAY return a 401 (Unauthorized) status if they require a From field.
Other header fields MAY be encrypted or MAY travel in the clear as Other header fields MAY be encrypted or MAY travel in the clear as
desired by the sender. The Subject, Allow, Call-ID, and Content- desired by the sender. The Subject, Allow, Call-ID, and Content-
Type header fields will typically be encrypted. The Accept, Type header fields will typically be encrypted. The Accept,
Accept-Language, Date, Expires, Priority, Require, Cseq, and Accept-Language, Date, Expires, Priority, Require, Cseq, and
Timestamp header fields will remain in the clear. Timestamp header fields will remain in the clear.
All fields that will remain in the clear MUST precede those that will All fields that will remain in the clear MUST precede those that will
be encrypted. The message is encrypted starting with the first be encrypted. The message is encrypted starting with the first
character of the first header field that will be encrypted and character of the first header field that will be encrypted and
continuing through to the end of the message body. If no header continuing through to the end of the message body. If no header
fields are to be encrypted, encrypting starts with the second CRLF fields are to be encrypted, encrypting starts with the second CRLF
pair after the last header field, as shown below. Carriage return and pair after the last header field, as shown below. Carriage return and
line feed characters have been made visible as "$", and the encrypted line feed characters have been made visible as "$", and the encrypted
part of the message is outlined. part of the message is outlined.
INVITE watson@boston.bell-telephone.com SIP/2.0$ INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$ Via: SIP/2.0/UDP 169.130.12.5$
To: watson@bell-telephone.com (T. A. Watson)$ To: T. A. Watson <sip:watson@bell-telephone.com>$
From: a.g.bell@bell-telephone.com (A. Bell)$ From: A. Bell <sip:a.g.bell@bell-telephone.com>$
Encryption: PGP version=5.0$ Encryption: PGP version=5.0$
Content-Length: 224$ Content-Length: 224$
CSeq: 488$ CSeq: 488$
$ $
******************************************************* *******************************************************
* Call-ID: 187602141351@worcester.bell-telephone.com$ * * Call-ID: 187602141351@worcester.bell-telephone.com$ *
* Subject: Mr. Watson, come here.$ * * Subject: Mr. Watson, come here.$ *
* Content-Type: application/sdp$ * * Content-Type: application/sdp$ *
* $ * * $ *
* v=0$ * * v=0$ *
skipping to change at page 76, line 5 skipping to change at page 87, line 7
length of the encrypted body. The encrypted body is preceded by a length of the encrypted body. The encrypted body is preceded by a
blank line as a normal SIP message body would be. blank line as a normal SIP message body would be.
Upon receipt by the called user agent possessing the correct Upon receipt by the called user agent possessing the correct
decryption key, the message body as indicated by the Content-Length decryption key, the message body as indicated by the Content-Length
field is decrypted, and the now-decrypted body is appended to the field is decrypted, and the now-decrypted body is appended to the
clear-text header fields. There is no need for an additional clear-text header fields. There is no need for an additional
Content-Length header field within the encrypted body because the Content-Length header field within the encrypted body because the
length of the actual message body is unambiguous after decryption. length of the actual message body is unambiguous after decryption.
Had no SIP header fields required encryption, the message would have
been as below. Note that the encrypted body must then include a blank been as below. Note that the encrypted body must then include a blank
line (start with CRLF) to disambiguate between any possible SIP line (start with CRLF) to disambiguate between any possible SIP
header fields that might have been present and the SIP message body. header fields that might have been present and the SIP message body.
INVITE watson@boston.bell-telephone.com SIP/2.0$ INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$ Via: SIP/2.0/UDP 169.130.12.5$
To: watson@bell-telephone.com (T. A. Watson)$ To: T. A. Watson <sip:watson@bell-telephone.com>$
From: a.g.bell@bell-telephone.com (A. Bell)$ From: A. Bell <a.g.bell@bell-telephone.com>$
Encryption: PGP version=5.0$ Encryption: PGP version=5.0$
Content-Type: application/sdp$ Content-Type: application/sdp$
Content-Length: 107$ Content-Length: 107$
$ $
************************************************* *************************************************
* $ * * $ *
* v=0$ * * v=0$ *
* o=bell 53655765 2353687637 IN IP4 128.3.4.5$ * * o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
* c=IN IP4 135.180.144.94$ * * c=IN IP4 135.180.144.94$ *
* m=audio 3456 RTP/AVP 0 3 4 5$ * * m=audio 3456 RTP/AVP 0 3 4 5$ *
************************************************* *************************************************
12.2.1 Privacy of SIP Responses 12.1.2 Privacy of SIP Responses
SIP requests may be sent securely using end-to-end encryption and SIP requests may be sent securely using end-to-end encryption and
authentication to a called user agent that sends an insecure authentication to a called user agent that sends an insecure
response. This is allowed by the SIP security model, but is not a response. This is allowed by the SIP security model, but is not a
good idea. good idea. However, unless the correct behaviour is explicit, it
would not always be possible for the called user agent to infer what
However, unless the correct behaviour is explicit, it would not a reasonable behaviour was. Thus when end-to-end encryption is used
always be possible for the called user agent to infer what a by the request originator, the encryption key to be used for the
reasonable behaviour was. Thus when end-to-end encryption is used by
the request originator, the encryption key to be used for the
response SHOULD be specified in the request. If this were not done, response SHOULD be specified in the request. If this were not done,
it might be possible for the called user agent to incorrectly infer it might be possible for the called user agent to incorrectly infer
an appropriate key to use in the response. Thus, to prevent key- an appropriate key to use in the response. Thus, to prevent key-
guessing becoming an acceptable strategy, we specify that a called guessing becoming an acceptable strategy, we specify that a called
user agent receiving a request that does not specify a key to be used user agent receiving a request that does not specify a key to be used
for the response SHOULD send that response unencrypted. for the response SHOULD send that response unencrypted.
Any SIP header fields that were encrypted in a request should also be Any SIP header fields that were encrypted in a request should also be
encrypted in an encrypted response. Location response fields MAY be encrypted in an encrypted response. Location response fields MAY be
encrypted if the information they contain is sensitive, or MAY be encrypted if the information they contain is sensitive, or MAY be
left in the clear to permit proxies more scope for localized left in the clear to permit proxies more scope for localized
searches. searches.
12.2.2 Encryption by Proxies 12.1.3 Encryption by Proxies
Normally, proxies are not allowed to alter end-to-end header fields Normally, proxies are not allowed to alter end-to-end header fields
and message bodies. Proxies MAY, however, encrypt an unsigned request and message bodies. Proxies MAY, however, encrypt an unsigned request
or response with the key of the call recipient.
Proxies may need to encrypt a SIP request if the end system Proxies may need to encrypt a SIP request if the end system
cannot perform encryption or to enforce organizational cannot perform encryption or to enforce organizational
security policies. security policies.
12.2.3 Hop-by-Hop Encryption 12.1.4 Hop-by-Hop Encryption
It is RECOMMENDED that SIP requests and responses are also protected It is RECOMMENDED that SIP requests and responses are also protected
by security mechanisms at the transport and network layer. by security mechanisms at the transport and network layer.
12.2.4 Via field encryption 12.1.5 Via field encryption
When Via fields are to be hidden, a proxy that receives a request When Via fields are to be hidden, a proxy that receives a request
containing an appropriate " Hide: hop" header field (as specified in containing an appropriate " Hide: hop" header field (as specified in
section 6.21) SHOULD encrypt the header field. As only the proxy that section 6.21) SHOULD encrypt the header field. As only the proxy that
encrypts the field will decrypt it, the algorithm chosen is entirely encrypts the field will decrypt it, the algorithm chosen is entirely
up to the proxy implementor. Two methods satisfy these requirements: up to the proxy implementor. Two methods satisfy these requirements:
o The server keeps a cache of Via fields and the associated To o The server keeps a cache of Via fields and the associated To
field, and replaces the Via field with an index into the field, and replaces the Via field with an index into the
cache. On the reverse path, take the Via field from the cache cache. On the reverse path, take the Via field from the cache
skipping to change at page 77, line 49 skipping to change at page 89, line 8
timestamp and an appropriate checksum in any such message with timestamp and an appropriate checksum in any such message with
the same secret key. The checksum is needed to detect whether the same secret key. The checksum is needed to detect whether
successful decoding has occurred, and the timestamp is successful decoding has occurred, and the timestamp is
required to prevent possible response attacks and to ensure required to prevent possible response attacks and to ensure
that no two requests from the same previous hop have the same that no two requests from the same previous hop have the same
encrypted Via field. encrypted Via field.
The latter is the preferred solution, although proxy developers may The latter is the preferred solution, although proxy developers may
devise other methods that might also satisfy the requirements. devise other methods that might also satisfy the requirements.
12.3 Message Integrity and Access Control: Authentication 12.2 Message Integrity and Access Control: Authentication
An active attacker may be able to modify and replay SIP requests and An active attacker may be able to modify and replay SIP requests and
responses unless protective measures are taken. In practice, the same responses unless protective measures are taken. In practice, the same
cryptographic measures that are used to ensure the authenticity of cryptographic measures that are used to ensure the authenticity of
the SIP message also serve to authenticate the originator of the the SIP message also serve to authenticate the originator of the
message.
Transport-layer or network-layer authentication may be used for hop- Transport-layer or network-layer authentication may be used for hop-
by-hop authentication. SIP also extends the HTTP WWW-Authenticate by-hop authentication. SIP also extends the HTTP WWW-Authenticate
(Section 6.42 and Authorization (Section 6.11) header and their (Section 6.42) and Authorization (Section 6.11) header and their
Proxy- counterparts to include cryptographically strong signatures. Proxy- counterparts to include cryptographically strong signatures.
SIP also supports the HTTP "basic" authentication scheme [33] that SIP also supports the HTTP "basic" authentication scheme
offers a very rudimentary mechanism of ascertaining the identity of
the caller. that offers a very rudimentary mechanism of ascertaining the identity
of the caller.
Since SIP requests are often sent to parties with which no Since SIP requests are often sent to parties with which no
prior communication relationship has existed, we do not prior communication relationship has existed, we do not
specify authentication based on shared secrets. specify authentication based on shared secrets.
SIP requests may be authenticated using the Authorization header SIP requests may be authenticated using the Authorization header
field to include a digital signature of certain header fields, the field to include a digital signature of certain header fields, the
request method and version number and the payload, none of which are request method and version number and the payload, none of which are
modified between client and called user agent. The Authorization modified between client and called user agent. The Authorization
header field may be used in requests to end-to-end authenticate the header field may be used in requests to end-to-end authenticate the
skipping to change at page 78, line 45 skipping to change at page 90, line 10
following the Authorization header field have been included in the following the Authorization header field have been included in the
signature. To sign a request, a client removes all of the SIP header signature. To sign a request, a client removes all of the SIP header
from before where the Authorization field will be added. It then from before where the Authorization field will be added. It then
prepends the request method (in upper case) followed by the SIP prepends the request method (in upper case) followed by the SIP
version number field (in upper case) directly to the start of the version number field (in upper case) directly to the start of the
message with no whitespace, CR or LF characters inserted. This message with no whitespace, CR or LF characters inserted. This
extended message is what is signed. extended message is what is signed.
For example, if the SIP request is to be: For example, if the SIP request is to be:
INVITE watson@boston.bell-telephone.com SIP/2.0 INVITE sip:watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5 Via: SIP/2.0/UDP 169.130.12.5
Authorization: PGP version=5.0, signature=... Authorization: PGP version=5.0, signature=...
From: a.g.bell@bell-telephone.com (A. Bell) From: A. Bell <sip:a.g.bell@bell-telephone.com>
To: watson@bell-telephone.com (T. A. Watson) To: T. A. Watson <sip:watson@bell-telephone.com>
Call-ID: 187602141351@worcester.bell-telephone.com Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here. Subject: Mr. Watson, come here.
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: ... Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5 o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94 c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5 m=audio 3456 RTP/AVP 0 3 4 5
Then the data block that is signed is: Then the data block that is signed is:
INVITESIP/2.0From: a.g.bell@bell-telephone.com (A. Bell) INVITESIP/2.0From: A. Bell <sip:a.g.bell@bell-telephone.com>
To: watson@bell-telephone.com (T. A. Watson) To: T. A. Watson <sip:watson@bell-telephone.com>
Call-ID: 187602141351@worcester.bell-telephone.com Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here. Subject: Mr. Watson, come here.
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: ... Content-Length: ...
v=0 v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5 o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94 c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5 m=audio 3456 RTP/AVP 0 3 4 5
skipping to change at page 79, line 35 skipping to change at page 91, line 7
before the digital signature is generated. On receipt, the digital before the digital signature is generated. On receipt, the digital
signature is checked before decryption. signature is checked before decryption.
A client MAY require that a server sign its response by including a A client MAY require that a server sign its response by including a
Require: org.ietf.sip.signed-response request header field. The Require: org.ietf.sip.signed-response request header field. The
client indicates the desired authentication method via the WWW- client indicates the desired authentication method via the WWW-
Authenticate header. Authenticate header.
The correct behaviour in handling unauthenticated responses to a The correct behaviour in handling unauthenticated responses to a
request that requires authenticated responses is described in section request that requires authenticated responses is described in section
12.3.1. 12.2.1.
12.3.1 Trusting responses 12.2.1 Trusting responses
It may be possible for an eavesdropper to listen to requests and to It may be possible for an eavesdropper to listen to requests and to
inject unauthenticated responses that would terminate, redirect or inject unauthenticated responses that would terminate, redirect or
otherwise interfere with a call. (Even encrypted requests contain otherwise interfere with a call. (Even encrypted requests contain
enough information to fake a response.) enough information to fake a response.)
Client should be particularly careful with 3xx redirection responses. Client should be particularly careful with 3xx redirection responses.
Thus a client receiving, for example, a 301 (Moved Permanently) which Thus a client receiving, for example, a 301 (Moved Permanently) which
was not authenticated when the public key of the called user agent is was not authenticated when the public key of the called user agent is
known to the client, and authentication was requested in the request known to the client, and authentication was requested in the request
SHOULD be treated as suspicious. The correct behaviour in such a case SHOULD be treated as suspicious. The correct behaviour in such a case
would be for the called-user to form a dated response containing the would be for the called-user to form a dated response containing the
Location field to be used, to sign it, and give this signed stub Location field to be used, to sign it, and give this signed stub
response to the proxy that will provide the redirection. Thus the response to the proxy that will provide the redirection. Thus the
response can be authenticated correctly. There may be circumstances
where such unauthenticated responses are unavoidable, but a client
SHOULD NOT automatically redirect such a request to the new location SHOULD NOT automatically redirect such a request to the new location
without alerting the user to the authentication failure before doing without alerting the user to the authentication failure before doing
so. so.
Another problem might be responses such as 6xx failure responses Another problem might be responses such as 6xx failure responses
which would simply terminate a search, or "4xx" and "5xx" response which would simply terminate a search, or "4xx" and "5xx" response
failures. failures.
If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as
valid, as they will not terminate a search. However, 6xx responses valid, as they will not terminate a search. However, 6xx responses
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called user agent, and so there is no simple way to detect these called user agent, and so there is no simple way to detect these
rogue responses. This problem is best prevented by using hop-by-hop rogue responses. This problem is best prevented by using hop-by-hop
encryption of the SIP request, which removes any additional problems encryption of the SIP request, which removes any additional problems
that UDP might have over TCP. that UDP might have over TCP.
These attacks are prevented by having the client require response These attacks are prevented by having the client require response
authentication and dropping unauthenticated responses. A server user authentication and dropping unauthenticated responses. A server user
agent that cannot perform response authentication responds using the agent that cannot perform response authentication responds using the
normal Require response of 420 (Bad Extension). normal Require response of 420 (Bad Extension).
12.4 Callee Privacy 12.3 Callee Privacy
User location and SIP-initiated calls may violate a callee's privacy. User location and SIP-initiated calls may violate a callee's privacy.
An implementation SHOULD be able to restrict, on a per-user basis, An implementation SHOULD be able to restrict, on a per-user basis,
what kind of location and availability information is given out to what kind of location and availability information is given out to
certain classes of callers. certain classes of callers.
12.5 Known Security Problems 12.4 Known Security Problems
With either TCP or UDP, a denial of service attack exists by a rogue With either TCP or UDP, a denial of service attack exists by a rogue
proxy sending 6xx responses. Although a client SHOULD choose to proxy sending 6xx responses. Although a client SHOULD choose to
ignore such responses if it requested authentication, a proxy cannot ignore such responses if it requested authentication, a proxy cannot
do so. It is obliged to forward the 6xx response back to the client. do so. It is obliged to forward the 6xx response back to the client.
The client can then ignore the response, but if it repeats the The client can then ignore the response, but if it repeats the
request it will probably reach the same rogue proxy again, and the request it will probably reach the same rogue proxy again, and the
process will repeat. process will repeat.
13 SIP Security Using PGP 13 SIP Security Using PGP
13.1 PGP Authentication Scheme 13.1 PGP Authentication Scheme
The "pgp" authentication scheme is based on the model that the client The "pgp" authentication scheme is based on the model that the client
must authenticate itself with a request signed with the client's
private key. The server can then ascertain the origin of the request
if it has access to the public key, preferably signed by a trusted if it has access to the public key, preferably signed by a trusted
third party. third party.
13.1.1 The WWW-Authenticate Response Header 13.1.1 The WWW-Authenticate Response Header
WWW-Authenticate = "WWW-Authenticate" ":" "pgp" pgp-challenge WWW-Authenticate = "WWW-Authenticate" ":" "pgp" pgp-challenge
pgp-challenge = 1# ( realm | pgp-version | pgp-algorithm ) pgp-challenge = 1# ( realm | pgp-version | pgp-algorithm )
realm = "realm" "=" realm-value realm = "realm" "=" realm-value
realm-value = quoted-string realm-value = quoted-string
pgp-version = "version" "=" digit *( "." digit ) *letter pgp-version = "version" "=" digit *( "." digit ) *letter
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13.1.2 The Authorization Request Header 13.1.2 The Authorization Request Header
The client is expected to retry the request, passing an Authorization The client is expected to retry the request, passing an Authorization
header line, which is defined as follows. header line, which is defined as follows.
Authorization ___ "Authorization" ":" "pgp" pgp-response Authorization ___ "Authorization" ":" "pgp" pgp-response
pgp-response ___ 1# (realm | pgp-version | pgp-signature | signed-by) pgp-response ___ 1# (realm | pgp-version | pgp-signature | signed-by)
pgp-signature ___ "signature" "=" quoted-string pgp-signature ___ "signature" "=" quoted-string
signed-by ___ "signed-by" "=" URI signed-by ___ "signed-by" "=" URI
The signature MUST correspond to the From header of the request
unless the signed-by parameter is provided. unless the signed-by parameter is provided.
pgp-signature: The PGP ASCII-armored signature, as it appears between pgp-signature: The PGP ASCII-armored signature, as it appears between
the "BEGIN PGP MESSAGE" and "END PGP MESSAGE" delimiters, the "BEGIN PGP MESSAGE" and "END PGP MESSAGE" delimiters,
without the version indication. The signature is included without the version indication. The signature is included
without any linebreaks. without any linebreaks.
The signature is computed across the request method, request version The signature is computed across the request method, request version
and header fields following the Authorization header and the message and header fields following the Authorization header and the message
body, in the same order as they appear in the message. The request body, in the same order as they appear in the message. The request
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Example: Example:
Authorization: pgp version="5.0", Authorization: pgp version="5.0",
realm="Your Startrek identity, please", realm="Your Startrek identity, please",
signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
=aIrx" =aIrx"
13.2 PGP Encryption Scheme 13.2 PGP Encryption Scheme
The PGP encryption scheme uses the following syntax:
Encryption ___ "Encryption" ":" "pgp" pgp-eparams Encryption ___ "Encryption" ":" "pgp" pgp-eparams
pgp-eparams ___ 1# ( pgp-version | pgp-encoding ) pgp-eparams ___ 1# ( pgp-version | pgp-encoding )
pgp-encoding ___ "encoding" "=" "ascii" | token pgp-encoding ___ "encoding" "=" "ascii" | token
encoding: Describes the encoding or "armor" used by PGP. The value encoding: Describes the encoding or "armor" used by PGP. The value
"ascii" refers to the standard PGP ASCII armor, without the "ascii" refers to the standard PGP ASCII armor, without the
lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and
without the version identifier. By default, the encrypted part without the version identifier. By default, the encrypted part
is included as binary. is included as binary.
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14 Examples 14 Examples
14.1 Registration 14.1 Registration
A user at host saturn.bell-tel.com registers on start-up, via A user at host saturn.bell-tel.com registers on start-up, via
multicast, with the local SIP server named sip.bell-tel.com the multicast, with the local SIP server named sip.bell-tel.com the
example, the user agent on saturn expects to receive SIP requests on example, the user agent on saturn expects to receive SIP requests on
UDP port 3890. UDP port 3890.
C->S: REGISTER sip:@sip.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP 128.16.64.19
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com To: sip:watson@bell-tel.com
Location: sip:saturn.bell-tel.com:3890;transport=udp Location: sip:saturn.bell-tel.com:3890;transport=udp
Call-ID: 123@saturn.bell-tel.com
Expires: 7200 Expires: 7200
CSeq: 1 REGISTER CSeq: 1 REGISTER
The registration expires after two hours. Any future invitations for The registration expires after two hours. Any future invitations for
watson@bell-tel.com arriving at sip.bell-tel.com will now be watson@bell-tel.com arriving at sip.bell-tel.com will now be
redirected to watson@saturn.bell-tel.com , UDP port 3890. redirected to watson@saturn.bell-tel.com , UDP port 3890.
If Watson wants to be reached elsewhere, say, an on-line service he If Watson wants to be reached elsewhere, say, an on-line service he
uses while traveling, he updates his reservation after first uses while traveling, he updates his reservation after first
cancelling any existing locations: cancelling any existing locations:
C->S: REGISTER sip:@bell-tel.com SIP/2.0 C->S: REGISTER sip:@bell-tel.com SIP/2.0
Via: SIP/2.0/UDP 128.16.64.19
From: sip:watson@bell-tel.com From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com To: sip:watson@bell-tel.com
Call-ID: 1234@saturn.bell-tel.com
Expire: 0 Expire: 0
Location: * Location: *
C->S: REGISTER sip:@bell-tel.com SIP/2.0 C->S: REGISTER sip:@bell-tel.com SIP/2.0
Via: SIP/2.0/UDP 128.16.64.19
From: sip:watson@bell-tel.com From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com To: sip:watson@bell-tel.com
Call-ID: 1235@saturn.bell-tel.com
Location: sip:tawatson@example.com Location: sip:tawatson@example.com
Now, the server will forward any request for Watson to the server at Now, the server will forward any request for Watson to the server at
example.com , using the Request-URI tawatson@example.com example.com , using the Request-URI tawatson@example.com
It is possible to use third-party registration. Here, the secretary It is possible to use third-party registration. Here, the secretary
jon.diligent registers his boss: jon.diligent registers his boss:
C->S: REGISTER sip:@bell-tel.com SIP/2.0 C->S: REGISTER sip:@bell-tel.com SIP/2.0
Via: SIP/2.0/UDP 128.16.64.19
From: sip:jon.diligent@bell-tel.com From: sip:jon.diligent@bell-tel.com
To: sip:watson@bell-tel.com To: sip:watson@bell-tel.com
Location: sip:tawatson@example.com Location: sip:tawatson@example.com
Call-ID: 1236@saturn.bell-tel.com
The request could be send to either the registrar at bell-tel.com or The request could be send to either the registrar at bell-tel.com or
the server at example.com example.com would proxy the request to the the server at example.com example.com would proxy the request to the
address indicated in the Request-URI. Then, Max-Forwards header address indicated in the Request-URI. Then, Max-Forwards header
could be used to restrict the registration to that server. could be used to restrict the registration to that server.
14.2 Invitation to Multicast Conference 14.2 Invitation to Multicast Conference
The first example invites schooler@vlsi.cs.caltech.edu to a multicast The first example invites schooler@vlsi.cs.caltech.edu to a multicast
session. All examples use the Session Description Protocol (SDP) (RFC session. All examples use the Session Description Protocol (SDP) (RFC
2327 [5]) as the session description format.
14.2.1 Request 14.2.1 Request
C->S: INVITE sip:schooler@vlsi.cs.caltech.edu SIP/2.0 C->S: INVITE sip:schooler@vlsi.cs.caltech.edu SIP/2.0
Via: SIP/2.0/UDP 239.128.16.254 ;ttl=16 Via: SIP/2.0/UDP 131.215.131.131;maddr=239.128.16.254;ttl=16
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: Mark Handley <sip:mjh@isi.edu> From: Mark Handley <sip:mjh@isi.edu>
To: Eve Schooler <sip:schooler@caltech.edu> To: Eve Schooler <sip:schooler@caltech.edu>
Subject: SIP will be discussed, too Subject: SIP will be discussed, too
Call-ID: 19971205T234505.56.78@oregon.isi.edu Call-ID: 42100bb8-1dd2-11b2-8d70-c91e31477491@oregon.isi.edu
Content-Type: application/sdp Content-Type: application/sdp
CSeq: 4711 INVITE CSeq: 4711 INVITE
Content-Length: 187 Content-Length: 187
v=0 v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5 o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio s=Mbone Audio
i=Discussion of Mbone Engineering Issues i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127 c=IN IP4 224.2.0.1/127
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invited; the message is currently being routed to invited; the message is currently being routed to
schooler@vlsi.cs.caltech.edu schooler@vlsi.cs.caltech.edu
In this case, the session description is using the Session In this case, the session description is using the Session
Description Protocol (SDP), as stated in the Content-Type header. Description Protocol (SDP), as stated in the Content-Type header.
The header is terminated by an empty line and is followed by a The header is terminated by an empty line and is followed by a
message body containing the session description. message body containing the session description.
14.2.2 Response 14.2.2 Response
The called user agent, directly or indirectly through proxy servers, The called user agent, directly or indirectly through proxy servers,
indicates that it is alerting ("ringing") the called party: indicates that it is alerting ("ringing") the called party:
S->C: SIP/2.0 180 Ringing S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 239.128.16.254 ;ttl=16 ;branch=17 Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348;
;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP north.east.isi.edu
To: Eve Schooler <sip:schooler@caltech.edu> To: Eve Schooler <sip:schooler@caltech.edu>
From: Mark Handley <sip:mjh@isi.edu> From: Mark Handley <sip:mjh@isi.edu>
Call-ID: 19971205T234505.56.78@north.east.isi.edu Call-ID: 42100bb8-1dd2-11b2-8d70-c91e31477491@north.east.isi.edu
Location: sip:es@jove.cs.caltech.edu Location: sip:es@jove.cs.caltech.edu
CSeq: 4711 INVITE CSeq: 4711 INVITE
A sample response to the invitation is given below. The first line of A sample response to the invitation is given below. The first line of
the response states the SIP version number, that it is a 200 (OK) the response states the SIP version number, that it is a 200 (OK)
response, which means the request was successful. The Via headers response, which means the request was successful. The Via headers
are taken from the request, and entries are removed hop by hop as the are taken from the request, and entries are removed hop by hop as the
response retraces the path of the request. A new authentication field response 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 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 taken directly from the original request, along with the remaining
fields of the request message. The original sense of From field is fields of the request message. The original sense of From field is
preserved (i.e., it is the session initiator). preserved (i.e., it is the session initiator).
In addition, the Location header gives details of the host where the In addition, the Location header gives details of the host where the
user was located, or alternatively the relevant proxy contact point user was located, or alternatively the relevant proxy contact point
which should be reachable from the caller's host. which should be reachable from the caller's host.
S->C: SIP/2.0 200 OK S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP 239.128.16.254 16 ;branch=17
Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348 Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
maddr=239.128.16.254 16;ttl=16
Via: SIP/2.0/UDP north.east.isi.edu Via: SIP/2.0/UDP north.east.isi.edu
From: sip:mjh@isi.edu From: sip:mjh@isi.edu
To: sip:schooler@cs.caltech.edu To: sip:schooler@cs.caltech.edu
Call-ID: 19971205T234505.56.78@north.east.isi.edu Call-ID: 42100bb8-1dd2-11b2-8d70-c91e31477491@oregon.isi.edu
Location: sip:es@jove.cs.caltech.edu Location: sip:es@jove.cs.caltech.edu
CSeq: 4711 INVITE CSeq: 4711 INVITE
The caller confirms the invitation by sending a request to the The caller confirms the invitation by sending a request to the
location named in the Location header: location named in the Location header:
C->S: ACK sip:es@jove.cs.caltech.edu SIP/2.0 C->S: ACK sip:es@jove.cs.caltech.edu SIP/2.0
From: sip:mjh@isi.edu From: sip:mjh@isi.edu
To: sip:schooler@cs.caltech.edu To: sip:schooler@cs.caltech.edu
Call-ID: 19971205T234505.56.78@oregon.isi.edu Call-ID: 42100bb8-1dd2-11b2-8d70-c91e31477491@oregon.isi.edu
CSeq: 4711 ACK CSeq: 4711 ACK
14.3 Two-party Call 14.3 Two-party Call
For two-party Internet phone calls, the response must contain a For two-party Internet phone calls, the response must contain a
description of where to send the data. In the example below, Bell description of where to send the data. In the example below, Bell
calls Watson. Bell indicates that he can receive RTP audio codings 0
(PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).
C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0 C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5 Via: SIP/2.0/UDP 169.130.12.5
From: A. Bell <sip:a.g.bell@bell-tel.com> From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com> To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com Call-ID: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
Subject: Mr. Watson, come here. Subject: Mr. Watson, come here.
CSeq: 17 INVITE CSeq: 17 INVITE
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: ... Content-Length: ...
v=0 v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5 o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94 c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5 m=audio 3456 RTP/AVP 0 3