Session Initiation Protocol (SIP) is a signaling protocol used to initiate, maintain, and terminate real-time sessions that include voice, video, and messaging applications. SIP is used to signal and control multimedia communication sessions in Internet telephony applications for voice and video calls, private IP telephone systems, instant messaging over Internet Protocol (IP) networks, and mobile calls over LTE (VoLTE ).
SIP is only involved in the signaling operations of a multimedia communication session and is primarily used to establish and terminate voice or video calls. SIP can be used to establish two-party (unicast) or multiparty (multicast) sessions. It also allows editing of existing calls. Editing may involve changing addresses or ports, inviting multiple participants, and adding or removing media streams.
SIP has also found applications in messaging applications, such as instant messaging and event subscription and notification. SIP works in conjunction with many other protocols that specify media format and encoding and transport media once the call is established. For call setup, the body of a SIP message contains a Session Description Protocol (SDP) data unit, which specifies the media format, codec, and communication protocol of the media. .SIP dialing is a marketing term for Voice over Internet Protocol (VoIP) services offered by many Internet Telephony Service Providers (ITSPs).
The service routes telephone calls from the customer's Private Branch Exchange (PBX) telephone system to the PSTN. These services can simplify the enterprise information system infrastructure by sharing Internet access for voice and data and eliminating the cost of basic rate interface (BRI) or interface telephony circuits. primary rate (PRI). SIP trunking is a similar marketing term preferred when the service is used to simplify a telecommunications infrastructure by sharing the carrier's access circuit for voice, data, and Internet traffic while eliminating the need for PRI circuits.
CCTV-enabled cameras can initiate calls to alert the operator of events, such as the movement of objects in a protected area. SIP is used in Audio over IP for broadcast applications where it provides an interoperable way for audio interfaces from different manufacturers to establish connections Concerns about call security over the public Internet have been addressed by encrypting the SIP protocol for secure transmission.
The SIPS URI scheme is used to ensure that SIP communication is secured with Transport Layer Security (TLS). End-to-end SIP encryption is only possible if there is a direct connection between the communication terminals. While a direct connection can be established through Peer To Peer SIP or through a VPN between endpoints, most SIP communications involve multiple hops, with the first hop originating from a user agent to the user agent's ITSP.For the multi-hop case.
The remaining hops will normally not be secured with TLS and the SIP communication will not be secure. In contrast, the HTTPS protocol provides end-to-end security as it occurs with a direct connection and does not involve the notion of a hop. Media streams (audio and video), which are separate connections from the SIPS signaling stream, can be encrypted using SRTP.
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