The transport layer is responsible for the actual transmission of
requests and responses over network transports. This includes
determination of the connection to use for a request or response in
the case of connection-oriented transports.
The transport layer is responsible for managing persistent
connections for transport protocols like TCP and SCTP, or TLS over
those, including ones opened to the transport layer. This includes
connections opened by the client or server transports, so that
connections are shared between client and server transport functions.
These connections are indexed by the tuple formed from the address,
port, and transport protocol at the far end of the connection. When
a connection is opened by the transport layer, this index is set to
the destination IP, port and transport. When the connection is
accepted by the transport layer, this index is set to the source IP
address, port number, and transport. Note that, because the source
port is often ephemeral, but it cannot be known whether it is
ephemeral or selected through procedures in , connections accepted
by the transport layer will frequently not be reused. The result is
that two proxies in a "peering" relationship using a connection-
oriented transport frequently will have two connections in use, one
for transactions initiated in each direction.
It is RECOMMENDED that connections be kept open for some
implementation-defined duration after the last message was sent or
received over that connection. This duration SHOULD at least equal
the longest amount of time the element would need in order to bring a
transaction from instantiation to the terminated state. This is to
make it likely that transactions are completed over the same
connection on which they are initiated (for example, request,
response, and in the case of INVITE, ACK for non-2xx responses).
This usually means at least 64*T1 (see Section 22.214.171.124 for a
definition of T1). However, it could be larger in an element that
has a TU using a large value for timer C (bullet 11 of Section 16.6),
All SIP elements MUST implement UDP and TCP. SIP elements MAY
implement other protocols.
Making TCP mandatory for the UA is a substantial change from RFC
2543. It has arisen out of the need to handle larger messages,
which MUST use TCP, as discussed below. Thus, even if an element
never sends large messages, it may receive one and needs to be
able to handle them.
18.1.1 Sending Requests
The client side of the transport layer is responsible for sending the
request and receiving responses. The user of the transport layer
passes the client transport the request, an IP address, port,
transport, and possibly TTL for multicast destinations.
If a request is within 200 bytes of the path MTU, or if it is larger
than 1300 bytes and the path MTU is unknown, the request MUST be sent
using an RFC 2914  congestion controlled transport protocol, such
as TCP. If this causes a change in the transport protocol from the
one indicated in the top Via, the value in the top Via MUST be
changed. This prevents fragmentation of messages over UDP and
provides congestion control for larger messages. However,
implementations MUST be able to handle messages up to the maximum
datagram packet size. For UDP, this size is 65,535 bytes, including
IP and UDP headers.
The 200 byte "buffer" between the message size and the MTU
accommodates the fact that the response in SIP can be larger than
the request. This happens due to the addition of Record-Route
header field values to the responses to INVITE, for example. With
the extra buffer, the response can be about 170 bytes larger than
the request, and still not be fragmented on IPv4 (about 30 bytes
is consumed by IP/UDP, assuming no IPSec). 1300 is chosen when
path MTU is not known, based on the assumption of a 1500 byte
If an element sends a request over TCP because of these message size
constraints, and that request would have otherwise been sent over
UDP, if the attempt to establish the connection generates either an
ICMP Protocol Not Supported, or results in a TCP reset, the element
SHOULD retry the request, using UDP. This is only to provide
backwards compatibility with RFC 2543 compliant implementations that
do not support TCP. It is anticipated that this behavior will be
deprecated in a future revision of this specification.
A client that sends a request to a multicast address MUST add the
"maddr" parameter to its Via header field value containing the
destination multicast address, and for IPv4, SHOULD add the "ttl"
parameter with a value of 1. Usage of IPv6 multicast is not defined
in this specification, and will be a subject of future
standardization when the need arises.
These rules result in a purposeful limitation of multicast in SIP.
Its primary function is to provide a "single-hop-discovery-like"
service, delivering a request to a group of homogeneous servers,
where it is only required to process the response from any one of
them. This functionality is most useful for registrations. In fact,
based on the transaction processing rules in Section 17.1.3, the
client transaction will accept the first response, and view any
others as retransmissions because they all contain the same Via
Before a request is sent, the client transport MUST insert a value of
the "sent-by" field into the Via header field. This field contains
an IP address or host name, and port. The usage of an FQDN is
RECOMMENDED. This field is used for sending responses under certain
conditions, described below. If the port is absent, the default
value depends on the transport. It is 5060 for UDP, TCP and SCTP,
5061 for TLS.
For reliable transports, the response is normally sent on the
connection on which the request was received. Therefore, the client
transport MUST be prepared to receive the response on the same
connection used to send the request. Under error conditions, the
server may attempt to open a new connection to send the response. To
handle this case, the transport layer MUST also be prepared to
receive an incoming connection on the source IP address from which
the request was sent and port number in the "sent-by" field. It also
MUST be prepared to receive incoming connections on any address and
port that would be selected by a server based on the procedures
described in Section 5 of .
For unreliable unicast transports, the client transport MUST be
prepared to receive responses on the source IP address from which the
request is sent (as responses are sent back to the source address)
and the port number in the "sent-by" field. Furthermore, as with
reliable transports, in certain cases the response will be sent
elsewhere. The client MUST be prepared to receive responses on any
address and port that would be selected by a server based on the
procedures described in Section 5 of .
For multicast, the client transport MUST be prepared to receive
responses on the same multicast group and port to which the request
is sent (that is, it needs to be a member of the multicast group it
sent the request to.)
If a request is destined to an IP address, port, and transport to
which an existing connection is open, it is RECOMMENDED that this
connection be used to send the request, but another connection MAY be
opened and used.
If a request is sent using multicast, it is sent to the group
address, port, and TTL provided by the transport user. If a request
is sent using unicast unreliable transports, it is sent to the IP
address and port provided by the transport user.
18.1.2 Receiving Responses
When a response is received, the client transport examines the top
Via header field value. If the value of the "sent-by" parameter in
that header field value does not correspond to a value that the
client transport is configured to insert into requests, the response
MUST be silently discarded.
If there are any client transactions in existence, the clienttransport uses the matching procedures of Section 17.1.3 to attemptto match the response to an existing transaction. If there is amatch, the response MUST be passed to that transaction. Otherwise,the response MUST be passed to the core (whether it be statelessproxy, stateful proxy, or UA) for further processing. Handling ofthese "stray" responses is dependent on the core (a proxy willforward them, while a UA will discard, for example).
The client transport uses the matching procedures of Section
17.1.3 to attempt to match the response to an existing
transaction. If there is a match, the response MUST be passed to
that transaction. Otherwise, any element other than a stateless
proxy MUST silently discard the response.
18.2.1 Receiving Requests
A server SHOULD be prepared to receive requests on any IP address,
port and transport combination that can be the result of a DNS lookup
on a SIP or SIPS URI  that is handed out for the purposes of
communicating with that server. In this context, "handing out"
includes placing a URI in a Contact header field in a REGISTER
request or a redirect response, or in a Record-Route header field in
a request or response. A URI can also be "handed out" by placing it
on a web page or business card. It is also RECOMMENDED that a server
listen for requests on the default SIP ports (5060 for TCP and UDP,
5061 for TLS over TCP) on all public interfaces. The typical
exception would be private networks, or when multiple server
instances are running on the same host. For any port and interface
that a server listens on for UDP, it MUST listen on that same port
and interface for TCP. This is because a message may need to be sent
using TCP, rather than UDP, if it is too large. As a result, the
converse is not true. A server need not listen for UDP on a
particular address and port just because it is listening on that same
address and port for TCP. There may, of course, be other reasons why
a server needs to listen for UDP on a particular address and port.
When the server transport receives a request over any transport, it
MUST examine the value of the "sent-by" parameter in the top Via
header field value. If the host portion of the "sent-by" parameter
contains a domain name, or if it contains an IP address that differs
from the packet source address, the server MUST add a "received"
parameter to that Via header field value. This parameter MUST
contain the source address from which the packet was received. This
is to assist the server transport layer in sending the response,
since it must be sent to the source IP address from which the request
Consider a request received by the server transport which looks like,
INVITE sip:bob@Biloxi.com SIP/2.0
Via: SIP/2.0/UDP bobspc.biloxi.com:5060
The request is received with a source IP address of 192.0.2.4.
Before passing the request up, the transport adds a "received"
parameter, so that the request would look like, in part:
INVITE sip:bob@Biloxi.com SIP/2.0
Via: SIP/2.0/UDP bobspc.biloxi.com:5060;received=192.0.2.4
Next, the server transport attempts to match the request to a servertransaction. It does so using the matching rules described inSection 17.2.3. If a matching server transaction is found, therequest is passed to that transaction for processing. If no match isfound, the request is passed to the core, which may decide toconstruct a new server transaction for that request. Note that whena UAS core sends a 2xx response to INVITE, the server transaction isdestroyed. This means that when the ACK arrives, there will be nomatching server transaction, and based on this rule, the ACK ispassed to the UAS core, where it is processed.
Next, the server transport attempts to match the request to a
server transaction. It does so using the matching rules described
in Section 17.2.3. If a matching server transaction is found, the
request is passed to that transaction for processing. If no match
is found, the request is passed to the core, which may decide to
construct a new server transaction for that request.
18.2.2 Sending Responses
The server transport uses the value of the top Via header field in
order to determine where to send a response. It MUST follow the
o If the "sent-protocol" is a reliable transport protocol such as
TCP or SCTP, or TLS over those, the response MUST be sent using
the existing connection to the source of the original request
that created the transaction, if that connection is still open.
This requires the server transport to maintain an association
between server transactions and transport connections. If that
connection is no longer open, the server SHOULD open a
connection to the IP address in the "received" parameter, if
present, using the port in the "sent-by" value, or the default
port for that transport, if no port is specified. If that
connection attempt fails, the server SHOULD use the procedures
in  for servers in order to determine the IP address and
port to open the connection and send the response to.
o Otherwise, if the Via header field value contains a "maddr"
parameter, the response MUST be forwarded to the address listed
there, using the port indicated in "sent-by", or port 5060 if
none is present. If the address is a multicast address, the
response SHOULD be sent using the TTL indicated in the "ttl"
parameter, or with a TTL of 1 if that parameter is not present.
o Otherwise (for unreliable unicast transports), if the top Via
has a "received" parameter, the response MUST be sent to the
address in the "received" parameter, using the port indicated
in the "sent-by" value, or using port 5060 if none is specified
explicitly. If this fails, for example, elicits an ICMP "port
unreachable" response, the procedures of Section 5 of 
SHOULD be used to determine where to send the response.
o Otherwise, if it is not receiver-tagged, the response MUST be
sent to the address indicated by the "sent-by" value, using the
procedures in Section 5 of .
In the case of message-oriented transports (such as UDP), if the
message has a Content-Length header field, the message body is
assumed to contain that many bytes. If there are additional bytes in
the transport packet beyond the end of the body, they MUST be
discarded. If the transport packet ends before the end of the
message body, this is considered an error. If the message is a
response, it MUST be discarded. If the message is a request, the
element SHOULD generate a 400 (Bad Request) response. If the message
has no Content-Length header field, the message body is assumed to
end at the end of the transport packet.
In the case of stream-oriented transports such as TCP, the Content-
Length header field indicates the size of the body. The Content-
Length header field MUST be used with stream oriented transports.
18.4 Error Handling
Error handling is independent of whether the message was a request or
If the transport user asks for a message to be sent over an
unreliable transport, and the result is an ICMP error, the behavior
depends on the type of ICMP error. Host, network, port or protocol
unreachable errors, or parameter problem errors SHOULD cause the
transport layer to inform the transport user of a failure in sending.
Source quench and TTL exceeded ICMP errors SHOULD be ignored.
If the transport user asks for a request to be sent over a reliable
transport, and the result is a connection failure, the transport
layer SHOULD inform the transport user of a failure in sending.
19 Common Message Components
There are certain components of SIP messages that appear in various
places within SIP messages (and sometimes, outside of them) that
merit separate discussion.
19.1 SIP and SIPS Uniform Resource Indicators
A SIP or SIPS URI identifies a communications resource. Like all
URIs, SIP and SIPS URIs may be placed in web pages, email messages,
or printed literature. They contain sufficient information to
initiate and maintain a communication session with the resource.
Examples of communications resources include the following:
o a user of an online service
o an appearance on a multi-line phone
o a mailbox on a messaging system
o a PSTN number at a gateway service
o a group (such as "sales" or "helpdesk") in an organization
A SIPS URI specifies that the resource be contacted securely. Thismeans, in particular, that TLS is to be used between the UAC and thedomain that owns the URI. From there, secure communications are usedto reach the user, where the specific security mechanism depends onthe policy of the domain.
A SIPS URI specifies that the resource be contacted securely.
This means, in particular, that TLS is to be used on each hop
between the UAC and the resource identified by the target SIPS
Any resource described by a SIP URI can be
"upgraded" to a SIPS URI by just changing the scheme, if it is
desired to communicate with that resource securely.
19.1.1 SIP and SIPS URI Components
The "sip:" and "sips:" schemes follow the guidelines in RFC 2396 .
They use a form similar to the mailto URL, allowing the specification
of SIP request-header fields and the SIP message-body. This makes it
possible to specify the subject, media type, or urgency of sessions
initiated by using a URI on a web page or in an email message. The
formal syntax for a SIP or SIPS URI is presented in Section 25. Its
general form, in the case of a SIP URI, is:
The format for a SIPS URI is the same, except that the scheme is
"sips" instead of sip. These tokens, and some of the tokens in their
expansions, have the following meanings:
user: The identifier of a particular resource at the host being
addressed. The term "host" in this context frequently refers
to a domain. The "userinfo" of a URI consists of this user
field, the password field, and the @ sign following them. The
userinfo part of a URI is optional and MAY be absent when the
destination host does not have a notion of users or when the
host itself is the resource being identified. If the @ sign is
present in a SIP or SIPS URI, the user field MUST NOT be empty.
If the host being addressed can process telephone numbers, for
instance, an Internet telephony gateway, a telephone-
subscriber field defined in RFC 2806  MAY be used to
populate the user field. There are special escaping rules for
encoding telephone-subscriber fields in SIP and SIPS URIs
described in Section 19.1.2.
password: A password associated with the user. While the SIP and
SIPS URI syntax allows this field to be present, its use is NOT
RECOMMENDED, because the passing of authentication information
in clear text (such as URIs) has proven to be a security risk
in almost every case where it has been used. For instance,
transporting a PIN number in this field exposes the PIN.
Note that the password field is just an extension of the user
portion. Implementations not wishing to give special
significance to the password portion of the field MAY simply
treat "user:password" as a single string.
host: The host providing the SIP resource. The host part contains
either a fully-qualified domain name or numeric IPv4 or IPv6
address. Using the fully-qualified domain name form is
RECOMMENDED whenever possible.
port: The port number where the request is to be sent.
URI parameters: Parameters affecting a request constructed from
URI parameters are added after the hostport component and are
separated by semi-colons.
URI parameters take the form:
parameter-name "=" parameter-value
Even though an arbitrary number of URI parameters may be
included in a URI, any given parameter-name MUST NOT appear
more than once.
This extensible mechanism includes the transport, maddr, ttl,
user, method and lr parameters.
The transport parameter determines the transport mechanism to
be used for sending SIP messages, as specified in . SIP can
use any network transport protocol. Parameter names are
defined for UDP (RFC 768 ), TCP (RFC 761 ), and SCTP
(RFC 2960 ). For a SIPS URI, the transport parameter MUST
indicate a reliable transport.
The maddr parameter indicates the server address to be
contacted for this user, overriding any address derived from
the host field. When an maddr parameter is present, the port
and transport components of the URI apply to the address
indicated in the maddr parameter value.  describes the
proper interpretation of the transport, maddr, and hostport in
order to obtain the destination address, port, and transport
for sending a request.
The maddr field has been used as a simple form of loose source
routing. It allows a URI to specify a proxy that must be
traversed en-route to the destination. Continuing to use the
maddr parameter this way is strongly discouraged (the
mechanisms that enable it are deprecated). Implementations
should instead use the Route mechanism described in this
document, establishing a pre-existing route set if necessary
(see Section 126.96.36.199). This provides a full URI to describe
the node to be traversed.
The ttl parameter determines the time-to-live value of the UDP
multicast packet and MUST only be used if maddr is a multicast
address and the transport protocol is UDP. For example, to
specify a call to firstname.lastname@example.org using multicast to
188.8.131.52 with a ttl of 15, the following URI would be
The set of valid telephone-subscriber strings is a subset of
valid user strings. The user URI parameter exists to
distinguish telephone numbers from user names that happen to
look like telephone numbers. If the user string contains a
telephone number formatted as a telephone-subscriber, the user
parameter value "phone" SHOULD be present. Even without this
parameter, recipients of SIP and SIPS URIs MAY interpret the
pre-@ part as a telephone number if local restrictions on the
name space for user name allow it.
The method of the SIP request constructed from the URI can be
specified with the method parameter.
The lr parameter, when present, indicates that the element
responsible for this resource implements the routing mechanisms
specified in this document. This parameter will be used in the
URIs proxies place into Record-Route header field values, and
may appear in the URIs in a pre-existing route set.
This parameter is used to achieve backwards compatibility with
systems implementing the strict-routing mechanisms of RFC 2543
and the rfc2543bis drafts up to bis-05. An element preparing
to send a request based on a URI not containing this parameter
can assume the receiving element implements strict-routing and
reformat the message to preserve the information in the
Since the uri-parameter mechanism is extensible, SIP elements
MUST silently ignore any uri-parameters that they do not
Headers: Header fields to be included in a request constructed
from the URI.
Headers fields in the SIP request can be specified with the "?"
mechanism within a URI. The header names and values are
encoded in ampersand separated hname = hvalue pairs. The
special hname "body" indicates that the associated hvalue is
the message-body of the SIP request.
Table 1 summarizes the use of SIP and SIPS URI components based on
the context in which the URI appears. The external column describes
URIs appearing anywhere outside of a SIP message, for instance on a
web page or business card. Entries marked "m" are mandatory, those
marked "o" are optional, and those marked "-" are not allowed.
Elements processing URIs SHOULD ignore any disallowed components if
they are present. The second column indicates the default value of
an optional element if it is not present. "--" indicates that the
element is either not optional, or has no default value.
URIs in Contact header fields have different restrictions depending
on the context in which the header field appears. One set applies to
messages that establish and maintain dialogs (INVITE and its 200 (OK)
response). The other applies to registration and redirection
messages (REGISTER, its 200 (OK) response, and 3xx class responses to
19.1.2 Character Escaping Requirements
default Req.-URI To From Contact R-R/Route external
user -- o o o o o o
password -- o o o o o o
host -- m m m m m m
port (1) o - - o o o
user-param ip o o o o o o
method INVITE - - - - - o
maddr-param -- o - - o o o
ttl-param 1 o - - o - o
transp.-param (2) o - - o o o
lr-param -- o - - - o o
other-param -- o o o o o o
headers -- - - - o - o
(1): The default port value is transport and scheme dependent. The
default is 5060 for sip: using UDP, TCP, or SCTP. The default is
5061 for sip: using TLS over TCP and sips: over TCP.
(2): The default transport is scheme dependent. For sip:, it is UDP.
For sips:, it is TCP.
Table 1: Use and default values of URI components for SIP header
field values, Request-URI and references
SIP follows the requirements and guidelines of RFC 2396  when
defining the set of characters that must be escaped in a SIP URI, and
uses its ""%" HEX HEX" mechanism for escaping. From RFC 2396 :
The set of characters actually reserved within any given URI
component is defined by that component. In general, a character
is reserved if the semantics of the URI changes if the character
is replaced with its escaped US-ASCII encoding . Excluded US-
ASCII characters (RFC 2396 ), such as space and control
characters and characters used as URI delimiters, also MUST be
escaped. URIs MUST NOT contain unescaped space and control
For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
For example, "@" is not in the set of characters in the user
component, so the user "j@s0n" must have at least the @ sign encoded,
as in "j%40s0n".
Expanding the hname and hvalue tokens in Section 25 show that all URI
reserved characters in header field names and values MUST be escaped.
The telephone-subscriber subset of the user component has special
escaping considerations. The set of characters not reserved in the
RFC 2806  description of telephone-subscriber contains a number of
characters in various syntax elements that need to be escaped when
used in SIP URIs. Any characters occurring in a telephone-subscriber
that do not appear in an expansion of the BNF for the user rule MUST
Note that character escaping is not allowed in the host component of
a SIP or SIPS URI (the % character is not valid in its expansion).
This is likely to change in the future as requirements for
Internationalized Domain Names are finalized. Current
implementations MUST NOT attempt to improve robustness by treating
received escaped characters in the host component as literally
equivalent to their unescaped counterpart. The behavior required to
meet the requirements of IDN may be significantly different.
19.1.3 Example SIP and SIPS URIs
The last sample URI above has a user field value of
"alice;day=tuesday". The escaping rules defined above allow a
semicolon to appear unescaped in this field. For the purposes of
this protocol, the field is opaque. The structure of that value is
only useful to the SIP element responsible for the resource.
19.1.4 URI Comparison
Some operations in this specification require determining whether two
SIP or SIPS URIs are equivalent. In this specification, registrars
need to compare bindings in Contact URIs in REGISTER requests (see
Section 10.3.). SIP and SIPS URIs are compared for equality
according to the following rules:
o A SIP and SIPS URI are never equivalent.
o Comparison of the userinfo of SIP and SIPS URIs is case-
sensitive. This includes userinfo containing passwords or
formatted as telephone-subscribers. Comparison of all other
components of the URI is case-insensitive unless explicitly
o The ordering of parameters and header fields is not significant
in comparing SIP and SIPS URIs.
o Characters other than those in the "reserved" set (see RFC 2396
) are equivalent to their ""%" HEX HEX" encoding.
o An IP address that is the result of a DNS lookup of a host name
does not match that host name.
o For two URIs to be equal, the user, password, host, and portcomponents must match.
o For two URIs to be equal, the user, password, host, and port
components must match. If the host component contains a textual
representation of IP addresses, then the representation of those
IP addresses may vary. If so, the host components are considered
to match if the different textual representations yield the same
binary IP address.
A URI omitting the user component will not match a URI that
includes one. A URI omitting the password component will not
match a URI that includes one.
A URI omitting any component with a default value will not
match a URI explicitly containing that component with its
default value. For instance, a URI omitting the optional port
component will not match a URI explicitly declaring port 5060.
The same is true for the transport-parameter, ttl-parameter,
user-parameter, and method components.
Defining sip:user@host to not be equivalent to
sip:user@host:5060 is a change from RFC 2543. When deriving
addresses from URIs, equivalent addresses are expected from
equivalent URIs. The URI sip:user@host:5060 will always
resolve to port 5060. The URI sip:user@host may resolve to
other ports through the DNS SRV mechanisms detailed in .
o URI uri-parameter components are compared as follows:
- Any uri-parameter appearing in both URIs must match.
- A user, ttl, or method uri-parameter appearing in only one
URI never matches, even if it contains the default value.
- A URI that includes an maddr parameter will not match a URI
that contains no maddr parameter.
- All other uri-parameters appearing in only one URI are
ignored when comparing the URIs.
o URI header components are never ignored. Any present header
component MUST be present in both URIs and match for the URIs
to match. The matching rules are defined for each header field
in Section 20.
The URIs within each of the following sets are equivalent:
The following URIs are equivalent because the underlying binary
representation of the IP addresses are the same although their
textual representations vary:
The URIs within each of the following sets are not equivalent:
SIP:ALICE@AtLanTa.CoM;Transport=udp (different usernames)
sip:email@example.com (can resolve to different ports)
sip:firstname.lastname@example.org (can resolve to different transports)
sip:email@example.com (can resolve to different port and transports)
sip:firstname.lastname@example.org (different header component)
sip:email@example.com (even though that's what
sip:firstname.lastname@example.org phone21.boxesbybob.com resolves to)
Note that equality is not transitive:
o sip:email@example.com and sip:firstname.lastname@example.org;security=on are
o sip:email@example.com and sip:firstname.lastname@example.org;security=off
o sip:email@example.com;security=on and
sip:firstname.lastname@example.org;security=off are not equivalent
19.1.5 Forming Requests from a URI
An implementation needs to take care when forming requests directly
from a URI. URIs from business cards, web pages, and even from
sources inside the protocol such as registered contacts may contain
inappropriate header fields or body parts.
An implementation MUST include any provided transport, maddr, ttl, or
user parameter in the Request-URI of the formed request. If the URI
contains a method parameter, its value MUST be used as the method of
the request. The method parameter MUST NOT be placed in the
Request-URI. Unknown URI parameters MUST be placed in the message's
An implementation SHOULD treat the presence of any headers or body
parts in the URI as a desire to include them in the message, and
choose to honor the request on a per-component basis.
An implementation SHOULD NOT honor these obviously dangerous header
fields: From, Call-ID, CSeq, Via, and Record-Route.
An implementation SHOULD NOT honor any requested Route header field
values in order to not be used as an unwitting agent in malicious
An implementation SHOULD NOT honor requests to include header fields
that may cause it to falsely advertise its location or capabilities.
These include: Accept, Accept-Encoding, Accept-Language, Allow,
Contact (in its dialog usage), Organization, Supported, and User-
An implementation SHOULD verify the accuracy of any requested
descriptive header fields, including: Content-Disposition, Content-
Encoding, Content-Language, Content-Length, Content-Type, Date,
Mime-Version, and Timestamp.
If the request formed from constructing a message from a given URI is
not a valid SIP request, the URI is invalid. An implementation MUST
NOT proceed with transmitting the request. It should instead pursue
the course of action due an invalid URI in the context it occurs.
The constructed request can be invalid in many ways. These
include, but are not limited to, syntax error in header fields,
invalid combinations of URI parameters, or an incorrect
description of the message body.
Sending a request formed from a given URI may require capabilities
unavailable to the implementation. The URI might indicate use of an
unimplemented transport or extension, for example. An implementation
SHOULD refuse to send these requests rather than modifying them to
match their capabilities. An implementation MUST NOT send a request
requiring an extension that it does not support.
For example, such a request can be formed through the presence of
a Require header parameter or a method URI parameter with an
unknown or explicitly unsupported value.
19.1.6 Relating SIP URIs and tel URLs
When a tel URL (RFC 2806 ) is converted to a SIP or SIPS URI, the
entire telephone-subscriber portion of the tel URL, including any
parameters, is placed into the userinfo part of the SIP or SIPS URI.
Thus, tel:+358-555-1234567;postd=pp22 becomes
In general, equivalent "tel" URLs converted to SIP or SIPS URIs in
this fashion may not produce equivalent SIP or SIPS URIs. The
userinfo of SIP and SIPS URIs are compared as a case-sensitive
string. Variance in case-insensitive portions of tel URLs and
reordering of tel URL parameters does not affect tel URL equivalence,
but does affect the equivalence of SIP URIs formed from them.
are equivalent, while
are equivalent, while
To mitigate this problem, elements constructing telephone-subscriber
fields to place in the userinfo part of a SIP or SIPS URI SHOULD fold
any case-insensitive portion of telephone-subscriber to lower case,
and order the telephone-subscriber parameters lexically by parameter
name, excepting isdn-subaddress and post-dial, which occur first and
in that order. (All components of a tel URL except for future-
extension parameters are defined to be compared case-insensitive.)
Following this suggestion, both
19.2 Option Tags
Option tags are unique identifiers used to designate new options
(extensions) in SIP. These tags are used in Require (Section 20.32),
Proxy-Require (Section 20.29), Supported (Section 20.37) and
Unsupported (Section 20.40) header fields. Note that these options
appear as parameters in those header fields in an option-tag = token
form (see Section 25 for the definition of token).
Option tags are defined in standards track RFCs. This is a change
from past practice, and is instituted to ensure continuing multi-
vendor interoperability (see discussion in Section 20.32 and Section20.37). An IANA registry of option tags is used to ensure easy
The "tag" parameter is used in the To and From header fields of SIP
messages. It serves as a general mechanism to identify a dialog,
which is the combination of the Call-ID along with two tags, one from
each participant in the dialog. When a UA sends a request outside of
a dialog, it contains a From tag only, providing "half" of the dialog
ID. The dialog is completed from the response(s), each of which
contributes the second half in the To header field. The forking of
SIP requests means that multiple dialogs can be established from a
single request. This also explains the need for the two-sided dialog
identifier; without a contribution from the recipients, the
originator could not disambiguate the multiple dialogs established
from a single request.
When a tag is generated by a UA for insertion into a request or
response, it MUST be globally unique and cryptographically random
with at least 32 bits of randomness. A property of this selection
requirement is that a UA will place a different tag into the From
header of an INVITE than it would place into the To header of the
response to the same INVITE. This is needed in order for a UA to
invite itself to a session, a common case for "hairpinning" of calls
in PSTN gateways. Similarly, two INVITEs for different calls will
have different From tags, and two responses for different calls will
have different To tags.
Besides the requirement for global uniqueness, the algorithm for
generating a tag is implementation-specific. Tags are helpful in
fault tolerant systems, where a dialog is to be recovered on an
alternate server after a failure. A UAS can select the tag in such a
way that a backup can recognize a request as part of a dialog on the
failed server, and therefore determine that it should attempt to
recover the dialog and any other state associated with it.