SIP protocol: description

Session Initiation Protocol (SIP) is a protocol for signaling and managing multimedia sessions. The most common applications in Internet telephony are for voice and video calls, and for instant messaging over IP networks (Internet Protocol).

It defines the messages that are sent between the endpoints and govern the creation, termination, and other significant elements of the call. The SIP protocol described above can be used to create, modify, and terminate sessions consisting of one or more media streams. It is an application-level protocol. Designed to be independent of the main transport layer. In other words, it is a text-based protocol that includes many HTTP (Hypertext Transfer) and Simple Mail Transfer Protocol (SMTP) elements.

SIP-protocol - what is it?

The SIP works in conjunction with several other application layer protocols that identify and transmit multimedia sessions. The identification and negotiation of media data is achieved in conjunction with the Session Description Protocol (SDP). To transfer multimedia streams - voice, video - it usually uses a real-time transport protocol (RTP) or Secure mode (SRTP). For secure message transmission, SIP can be encrypted using Transport Layer Security (TLS).

History of development

The SIP protocol was originally developed by a team of specialists in 1996. It was standardized in RFC 2543 in 1999 (SIP 1.0). In November 2000, it was adopted as a signaling protocol for 3 GPPs and a permanent element of the IP architecture of the Multimedia Subsystem (IMS) for streaming multimedia services based on IP in cellular communication systems. The latest version (SIP 2.0) in the RFC 3261 specification was released in June 2002. With certain extensions and refinements, it is also used nowadays.

Despite the fact that the original SIP-protocol was developed based on voice services. Today, it supports a wide range of applications, including video conferencing, streaming media, instant messaging, file and fax transmission over IP and online games.

SIP protocol - description and operations

The Session Initiation Protocol is independent of the underlying transport protocol. It operates on the basis of Transmission Control Protocol (TCP), user Datagram Protocol (UDP) or flow control protocol (SCTP). It can be used both for data transfer between two parties (unicast) and for a multicast session.

It has design elements similar to the HTTP transaction request model. Each such operation consists of a client request that calls a particular method or function on the server, and at least one response. The SIP protocol reuses most of the header fields, encoding rules and HTTP status codes, providing a readable text format.

Each Session Initiation Protocol network resource-the user agent or the voicemail mailbox-is recognized using a resource allocation identifier (URI) that operates on the basis of a common standard syntax that is also used in web services and e-mail. The URI scheme used for SIP is a logical chain: user name: password @ host: port.

Security policy

If you want to securely transfer data, the schema dictates that each of the network elements that the request is routed to the target domain must be provided with Transport Layer Security (TLS). The last step from the proxy server to the target domain is to operate in accordance with the local security settings. TLS protects against intruders who try to intercept data at the time of their sending. But it does not provide real security to the end and can not prevent the tracking and theft of information. How can the SIP protocol, whose ports should be securely connected, works with other network services?

It works in conjunction with several other protocols and only participates in signaling the communication session. SIP clients typically use TCP or UDP with port numbers 5060 or 5061 to connect to SIP servers and other SIP endpoints. Port 5060 is typically used for unencrypted signaling traffic, while port 5061 is closely "friends" with Transport Layer Security (TLS).

What is it used for?

To most accurately answer the question "SIP-protocol - what is this?", It is necessary to understand what it is used for. It is used usually in setting up and transferring voice or video calls. It allows you to modify existing calls. The modification can include changing addresses or ports, inviting more participants to the conversation, adding or deleting multimedia streams. SIP also found application in messaging applications, as well as in subscription and event notification services.

A set of SIP rules related to the Internet Engineering Task Force (IETF) defines the instruction for such applications. Voice and video streams in applications are transferred to another application protocol in real-time Transport Protocol (RTP). Parameters-port numbers, protocols, codecs-for these media streams are defined and negotiated using the Session Initiation Protocol (SDP), which moves in the body of the Session Initiation Protocol (for example, the SIP T protocol).

The main prospect for the development of the protocol is that it should in the future ensure the installation of signaling and calling for IP communications on the basis that can support a superset of call processing functions and options present in the public switched telephone network (PSTN). It does not by itself define them. More precisely, it regulates only call and alarm settings. All actions that are directed at performing such telephone operations (i.e. dialing, response ringbacktones or busy signal) are performed by proxy servers and user agents. Their implementation and terminology are different in different countries of the world, but they operate on the same principle.

Value in telephone communication

SIP-enabled telephone networks can also implement many of the more advanced call handling features present in Signaling System 7 (SS7). Although both these protocols are very different. SS7 is a centralized protocol. It is characterized by a complex central network architecture and "blunt" endpoints (traditional telephones). SIP is a client-server protocol. However, most devices with Session Initiation Protocol support can perform both the client and server roles. In general, the initiator of the session is the client, and the call recipient performs the server function. Thus, SIP functions are implemented in communicating endpoints, contrary to the traditional SS7 capabilities that are implemented on the network.

SIP is fundamentally different in that this technology is developing in the IT field, and not in the telecommunications industry. The SIP protocol is standardized and defined primarily by the IETF, while others (eg H.323) are traditionally associated with the International Telecommunication Union (ITU).

Network Elements

SIP defines user agents, as well as several types of network elements of the server. Two SIP endpoints can interact without any intermediate infrastructure. Nevertheless, this approach often turns out to be impractical for state communication, which needs a directory service to find available nodes on the network. The register's SIP protocol can not provide this functionality.

User agent

The SIP user agent (UA) is a logical network of endpoints. They are used to create or receive messages and thus control the SIP session. The SIP-UA can act as a User Agent (UAC) client, which sends SIP requests, as well as its server (UAS), which receives requests and returns a SIP response. Such account control and UAS is only performed during the SIP transaction.

Telephony

SIP-telephony, in fact, is an IP-telephony, which implements the client and server functions of the user SIP-agent. In addition, it provides traditional phone call options - dialing, replying, rejecting, holding / dropping and forwarding a call.

SIP phones can be implemented as a hardware device or as a softphone. As manufacturers increasingly use this protocol as a standard telephony platform (in recent years - through 4G), the difference between the hardware and software basics of SIP phones remains blurred. In addition, Session Initiation Protocol elements are now implemented in the basic functions of the firmware of many IP-compatible devices. Examples are many devices from Nokia and BlackBerry, and SIP-protocol on Android is now an indispensable service.

In SIP, as in HTTP, the user agent can identify itself using a User-Agent header field message containing a textual description of the software / hardware / product names. The user agent field is sent in the request messages. This means that the receiving SIP server can see this information. Session Initiation Protocol network elements can sometimes store this information. And this can be useful in diagnosing compatibility problems.