4], 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 18.104.22.168 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), for example. 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. 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 Ethernet MTU. 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 branch identifier. 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.
4] 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 came. Consider a request received by the server transport which looks like, in part: 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 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. Note that when a UAS core sends a 2xx response to INVITE, the server transaction is destroyed. This means that when the ACK arrives, there will be no matching server transaction, and based on this rule, the ACK is passed to the UAS core, where it is processed. 4] 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.
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: sip:user:password@host:port;uri-parameters?headers 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 the URI. 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 22.214.171.124). 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 126.96.36.199 with a ttl of 15, the following URI would be used: sip:email@example.com;maddr=188.8.131.52;ttl=15 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 Request-URI. Since the uri-parameter mechanism is extensible, SIP elements MUST silently ignore any uri-parameters that they do not understand. 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 any method).
dialog reg./redir. Contact/ 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 characters. For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. 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 be escaped. 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. 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 defined otherwise. 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 port components must match. 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: sip:%firstname.lastname@example.org;transport=TCP sip:alice@AtLanTa.CoM;Transport=tcp sip:email@example.com sip:firstname.lastname@example.org;newparam=5 sip:email@example.com;security=on sip:biloxi.com;transport=tcp;method=REGISTER?to=sip:bob%40biloxi.com sip:biloxi.com;method=REGISTER;transport=tcp?to=sip:bob%40biloxi.com sip:firstname.lastname@example.org?subject=project%20x&priority=urgent sip:email@example.com?priority=urgent&subject=project%20x The URIs within each of the following sets are not equivalent: SIP:ALICE@AtLanTa.CoM;Transport=udp (different usernames) sip:alice@AtLanTa.CoM;Transport=UDP sip:firstname.lastname@example.org (can resolve to different ports) sip:email@example.com:5060 sip:firstname.lastname@example.org (can resolve to different transports) sip:email@example.com;transport=udp sip:firstname.lastname@example.org (can resolve to different port and transports) sip:email@example.com:6000;transport=tcp sip:firstname.lastname@example.org (different header component) sip:email@example.com?Subject=next%20meeting sip:firstname.lastname@example.org (even though that's what sip:email@example.com phone21.boxesbybob.com resolves to) Note that equality is not transitive: o sip:firstname.lastname@example.org and sip:email@example.com;security=on are equivalent o sip:firstname.lastname@example.org and sip:email@example.com;security=off are equivalent
o sip:firstname.lastname@example.org;security=on and sip:email@example.com;security=off are not equivalent
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. 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 sip:+358-555-1234567;firstname.lastname@example.org;user=phone or sips:+358-555-1234567;email@example.com;user=phone not sip:+firstname.lastname@example.org;postd=pp22;user=phone or sips:+email@example.com;postd=pp22;user=phone 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. For example, tel:+358-555-1234567;postd=pp22 tel:+358-555-1234567;POSTD=PP22 are equivalent, while sip:+358-555-1234567;firstname.lastname@example.org;user=phone sip:+358-555-1234567;POSTD=PP22@foo.com;user=phone
are not. Likewise, tel:+358-555-1234567;postd=pp22;isub=1411 tel:+358-555-1234567;isub=1411;postd=pp22 are equivalent, while sip:+358-555-1234567;postd=pp22;email@example.com;user=phone sip:+358-555-1234567;isub=1411;firstname.lastname@example.org;user=phone are not. 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 tel:+358-555-1234567;postd=pp22 tel:+358-555-1234567;POSTD=PP22 become sip:+358-555-1234567;email@example.com;user=phone and both tel:+358-555-1234567;tsp=a.b;phone-context=5 tel:+358-555-1234567;phone-context=5;tsp=a.b become sip:+358-555-1234567;phone-context=5;firstname.lastname@example.org;user=phone 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 Section 20.37). An IANA registry of option tags is used to ensure easy reference.