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Security Considerations. Normative References. Informative References. Intellectual Property Statement.
Authors' Addresses. Full Copyright Statement. They represent an example minimum set of functionality. It is the hope of the authors that this document will be useful for SIP implementers, designers, and protocol researchers alike and will help further the goal of a standard implementation of RFC [ 1 ].
These flows represent carefully checked and working group reviewed scenarios of the most basic examples as a companion to the specifications. General Assumptions A number of architecture, network, and protocol assumptions underlie the call flows in this document.
Note that these assumptions are not requirements. They are outlined in this section so that they may be taken into consideration and to aid in the understanding of the call flow examples. Some Proxy Servers in these call flows insert Record-Route headers into requests to ensure that they are in the signaling path for future message exchanges.
Legend for Message Flows Dashed lines represent signaling messages that are mandatory to the call scenario. The arrow indicates the direction of message flow. Messages with parentheses around their name represent optional messages.
Messages are identified in the Figures as F1, F2, etc. This references the message details in the list that follows the Figure.
SIP Protocol Assumptions This document does not prescribe the flows precisely as they are shown, but rather the flows illustrate the principles for best practice. They are best practices usages orderings, syntax, selection of features for the purpose, handling of error of SIP methods, headers and parameters.
IMPORTANT: The exact flows here must not be copied as is by an implementer due to specific incorrect characteristics that were introduced into the document for convenience and are listed below. For simplicity in reading and editing the document, there are a number of differences between some of the examples and actual SIP messages. Call-IDs are often repeated, and CSeq counts often begin at 1. Header fields are usually shown in the same order.
Usually only the minimum required header field set is shown, others that would normally be present such as Accept, Supported, Allow, etc are not shown. The request includes the user's contact list.
The SIP server provides a challenge to Bob. The SIP server validates the user's credentials. The response includes the user's current contact list in Contact headers.
The format of the authentication shown is HTTP digest. It is assumed that Bob has not previously registered with this Server. Bob's request includes an updated contact list. Since the user already has authenticated with the server, the user supplies authentication credentials with the request and is not challenged by the server.
SIP can be used to establish two-party unicast or multiparty multicast sessions. It also allows modification of existing calls. The modification can involve changing addresses or ports , inviting more participants, and adding or deleting media streams. SIP has also found applications in messaging applications, such as instant messaging, and event subscription and notification. SIP works in conjunction with several other protocols that specify the media format and coding and that carry the media once the call is set up.
For call setup, the body of a SIP message contains a Session Description Protocol SDP data unit, which specifies the media format, codec and media communication protocol.
The syntax of the URI follows the general standard syntax also used in Web services and e-mail. If secure transmission is required, the scheme sips is used. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format. Port is commonly used for non-encrypted signaling traffic whereas port is typically used for traffic encrypted with Transport Layer Security TLS.
SIP-based telephony networks often implement call processing features of Signaling System 7 SS7 , for which special SIP protocol extensions exist, although the two protocols themselves are very different. SS7 is a centralized protocol, characterized by a complex central network architecture and dumb endpoints traditional telephone handsets.
SIP is a client-server protocol of equipotent peers. SIP features are implemented in the communicating endpoints, while the traditional SS7 architecture is in use only between switching centers. Each user agent UA performs the function of a user agent client UAC when it is requesting a service function, and that of a user agent server UAS when responding to a request. However, for network operational reasons, for provisioning public services to users, and for directory services, SIP defines several specific types of network server elements.
Each of these service elements also communicates within the client-server model implemented in user agent clients and servers. User agents have client and server components. Unlike other network protocols that fix the roles of client and server, e. As vendors increasingly implement SIP as a standard telephony platform, the distinction between hardware-based and software-based SIP phones is blurred and SIP elements are implemented in the basic firmware functions of many IP-capable communications devices such as smartphones.
In SIP, as in HTTP, the user agent may identify itself using a message header field User-Agent , containing a text description of the software, hardware, or the product name.
The user agent field is sent in request messages, which means that the receiving SIP server can evaluate this information to perform device-specific configuration or feature activation. Operators of SIP network elements sometimes store this information in customer account portals,  where it can be useful in diagnosing SIP compatibility problems or in the display of service status.
Proxy server[ edit ] A proxy server is a network server with UAC and UAS components that functions as an intermediary entity for the purpose of performing requests on behalf of other network elements.