Youth Carnival
Transactions
Example: User1’s UAC uses anInvite Client Transaction to send the initial INVITE (1) message. If no response is received after a timer controlled wait period the UAC may chose to terminate the transaction or retransmit the INVITE. Once a response is received, User1 is confident the INVITE was delivered reliably. User1’s UAC must then acknowledge the response. On delivery of the ACK (2) both sides of the transaction are complete. In this case, a dialog may have been established.
SIP makes use of transactions to control the exchanges between participants and deliver messages reliably. The transactions maintain an internal state and make use of timers. Client Transactions send requests and Server Transactions respond to those requests with one-or-more responses. The responses may include zero-or-more Provisional (1xx) responses and one-or-more final (2xx-6xx) responses.
Transactions are further categorized as either Invite or Non-Invite. Invite transactions differ in that they can establish a long-running conversation, referred to as a Dialog in SIP, and so include an acknowledgment (ACK) of any non-failing final response (e.g. 200 OK).
Because of these transactional mechanisms, SIP can make use of un-reliable transports such as User Datagram Protocol (UDP).
Instant messaging and presence
The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence information. MSRP (Message Session Relay Protocol) allows instant message sessions and file transfer.
Conformance testing
TTCN-3 test specification language is used for the purposes of specifying conformance tests for SIP implementations. SIP test suite is developed by a Specialist Task Force at ETSI (STF 196).The SIP developer community meets regularly at the SIP Forum SIPit events to test interoperability and test implementations of new RFCs.
Applications
A SIP connection is a marketing term for voice over Internet Protocol (VoIP) services offered by many Internet telephony service providers (ITSPs). The service provides routing of telephone calls from a clients private branch exchange (PBX) telephone system to the public switched telephone network (PSTN). Such services may simplify corporate information system infrastructure by sharing Internet accessfor voice and data, and removing the cost for Basic Rate Interface (BRI) or Primary Rate Interface (PRI) telephone circuits.
Many VoIP phone companies allow customers to use their own SIP devices, such as SIP-capable telephone sets, or softphones.
SIP-enabled video surveillance cameras can make calls to alert the owner or operator that an event has occurred; for example, to notify that motion has been detected out-of-hours in a protected area.
SIP is used in audio over IP for broadcasting applications where it provides an interoperable means for audio interfaces from different manufacturers to make connections with one another.
SIP-ISUP interworking
SIP-I, or the Session Initiation Protocol with encapsulated ISUP, is a protocol used to create, modify, and terminate communication sessions based on ISUP using SIP and IP networks. Services using SIP-I include voice, video telephony, fax and data. SIP-I and SIP-T[22] are two protocols with similar features, notably to allow ISUP messages to be transported over SIP networks. This preserves all of the detail available in the ISUP header, which is important as there are many country-specific variants of ISUP that have been implemented over the last 30 years, and it is not always possible to express all of the same detail using a native SIP message. SIP-I was defined by the ITU-T, whereas SIP-T was defined via the IETFRFC route.
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Useful information will be) especially for young boys))
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