Sip Handbook Services Technologies And Security Of Session Initiation Protocol Pdf
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- SIP Handbook: Services, Technologies, and Security of Session Initiation Protocol
- Sip Handbook Services Technologies And Security Of Session Initiation Protocol
- Session Initiation Protocol (SIP) Server Overload Control: Design and Evaluation
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SIP Handbook: Services, Technologies, and Security of Session Initiation Protocol
The Session Initiation Protocol SIP is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP works in conjunction with several other protocols that specify and carry the session media.
Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol SDP , which is carried as payload in SIP messages. SIP was designed to provide a signaling and call setup protocol for IP-based communications supporting the call processing functions and features present in the public switched telephone network PSTN with a vision of supporting new multimedia applications. It has been extended for video conferencing , streaming media distribution, instant messaging , presence information , file transfer , Internet fax and online games.
SIP is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry. SIP is only involved in the signaling operations of a media communication session and is primarily used to set up and terminate voice or video calls. 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. A SIP phone is an IP phone that implements client and server functions of a SIP user agent and provides the traditional call functions of a telephone, such as dial, answer, reject, call hold, and call transfer.
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.
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. A proxy server primarily plays the role of call routing; it sends SIP requests to another entity closer to it destination. Proxies are also useful for enforcing policy, such as for determining whether a user is allowed to make a call.
A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. SIP proxy servers that route messages to more than one destination are called forking proxies.
The forking of a SIP request establishes multiple dialogs from the single request. Thus, a call may be answered from one of multiple SIP endpoints. For identification of multiple dialogs, each dialog has an identifier with contributions from both endpoints. A redirect server is a user agent server that generates 3xx redirection responses to requests it receives, directing the client to contact an alternate set of URIs. A redirect server allows proxy servers to direct SIP session invitations to external domains.
A registrar is a SIP endpoint that provides a location service. For subsequent requests, it provides an essential means to locate possible communication peers on the network.
Multiple user agents may register for the same URI, with the result that all registered user agents receive the calls to the URI. To improve network scalability, location services may instead be located with a redirect server. Session border controllers serve as middleboxes between user agents and SIP servers for various types of functions, including network topology hiding and assistance in NAT traversal.
There are two different types of SIP messages: requests and responses. The first line of a request has a method , defining the nature of the request, and a Request-URI, indicating where the request should be sent. Requests initiate a functionality of the protocol.
They are sent by a user agent client to the server, and are answered with one or more SIP responses , which return a result code of the transaction, and generally indicate the success, failure, or other state of the transaction.
Responses are sent by the user agent server indicating the result of a received request. Several classes of responses are recognized, determined by the numerical range of result codes: . SIP defines a transaction mechanism to control the exchanges between participants and deliver messages reliably. A transaction is a state of a session, which is controlled by various timers. Client transactions send requests and server transactions respond to those requests with one or more responses.
The responses may include provisional responses with a response code in the form 1xx , and one or multiple final responses 2xx — 6xx. Transactions are further categorized as either type invite or type 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. When developing SIP software or deploying a new SIP infrastructure, it is important to test capability of servers and IP networks to handle certain call load: number of concurrent calls and number of calls per second.
SIP trunking is a similar marketing term preferred for when the service is used to simplify a telecom infrastructure by sharing the carrier access circuit for voice, data, and Internet traffic while removing the need for PRI circuits. SIP-enabled video surveillance cameras can initiate calls to alert the operator of events, such as motion of objects 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.
The U. The implementation can work in proxy server or user agent scenarios and has been used in numerous commercial and research projects.
Numerous other commercial and open-source SIP implementations exist. See List of SIP software. Services using SIP-I include voice, video telephony, fax and data. This preserves all of the detail available in the ISUP header. Concerns about the security of calls via the public internet have been addressed by encryption of the SIP protocol for secure transmission.
End-to-end encryption of SIP is only possible if there is a direct connection between communication endpoints. In contrast, the HTTPS protocol provides end-to-end security as it is done with a direct connection and does not involve the notion of hops. From Wikipedia, the free encyclopedia. Computer network protocol. Not to be confused with Standard Interchange Protocol.
Main article: List of SIP response codes. Network World. May 11, Artech House. Internet Engineering Task Force. Retrieved RFC Telecom Teracom Training Institute. Transactions on Emerging Telecommunications Technologies. March Handbook of algorithms for wireless networking and mobile computing.
CRC Press. Practical VoIP Security. VoIP User. Archived from the original on HD Voice News. EBU Technical Review. September Archived from the original PDF on Instant messaging.
List of defunct instant messaging platforms. Hidden categories: Articles with short description Short description matches Wikidata Articles with Curlie links. Namespaces Article Talk.
Sip Handbook Services Technologies And Security Of Session Initiation Protocol
The scope of this volume ranges from basic concepts to future perspectives. Divided into three sections, the book begins with a discussion of SIP in peer-to-peer networks and then goes on to examine advanced media integration, migration considerations, mobility management, and group conferencing, while also reviewing home networking and compliance issues. The middle section of the book focuses on the underlying technologies of SIP. Chapters review network architecture, vertical handoffs, NAT traversals, multipoint extensions, and other areas at the forefront of research. Finally, the text examines various security vulnerabilities and provides perspectives on secure intelligent SIP services with a future outlook on a fraud detection framework in VoIP networks. Authored by 65 experts from across the world, this text is sure to advance the field of knowledge in this ever-changing industry and provide further impetus for new areas of exploration. Account Options Sign in.
Skype for Business Online SfBO , as part of the Microsoft and Office services, follows all the security best practices and procedures such as service-level security through defense-in-depth, customer controls within the service, security hardening and operational best practices. Information Technology Depr. Protocols are shared standards or rules for transmitting data packets among telephones, PCs, devices. To operate properly, devices must receive and send data under the same standards. Various techniques have been devised to detect spam calls; some take effect even before the recipient has answered a call to disconnect.
Session Initiation Protocol (SIP) Server Overload Control: Design and Evaluation
The SIP server overload problem is interesting especially because the costs of serving or rejecting a SIP session can be similar. For this reason, the built-in SIP overload control mechanism based on generating rejection messages cannot prevent the server from entering congestion collapse under heavy load. The pushback framework can be achieved by either a rate-based feedback or a window-based feedback. The centerpiece of the feedback mechanism is the algorithm used to generate load regulation information.
Home Login My Account. Cart 0. Change Location. By author : Alan B. Description Contents Author Reviews Now in its fourth edition, the ground-breaking Artech House bestseller SIP: Understanding the Session Initiation Protocol offers you the most comprehensive and current understanding of this revolutionary protocol for call signaling and IP Telephony.
Widely adopted by service providers to enable IP telephony, instant messaging, and other data services, SIP is the signaling protocol of choice for advanced multimedia communications signaling.
The Session Initiation Protocol SIP is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol SDP , which is carried as payload in SIP messages.
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