SIP is a text-based protocol similar to HTTP. SIP can reduce the development time of applications, especially advanced applications. Because SIP based on the IP protocol utilizes IP networks, fixed network operators will gradually realize the profound significance of SIP technology for them. 1. Introduction What is SIP SIP (Session IniTIaTIon Protocol) is an application layer signaling control protocol. Used to create, modify, and release one or more participants' sessions. These sessions can be like Internet multimedia conferences, IP phones or multimedia distribution. Participants in a session can communicate via multicast (mulTIcast), mesh unicast (unicast), or a mixture of both. SIP is a text-based protocol similar to HTTP. SIP can reduce the development time of applications, especially advanced applications. Because SIP based on the IP protocol utilizes IP networks, fixed network operators will gradually realize the profound significance of SIP technology for them. With SIP, service providers can choose standard components at will. Regardless of media content and number of participants, users can find and contact each other. SIP negotiates the session so that all parties can agree on and modify the session functions. It can even add, delete or transfer users. SIP is neither a session description protocol nor a conference control function. In order to describe the load and characteristics of the message content, SIP uses the Internet's Session Description Protocol (SDP) to describe the characteristics of terminal devices. SIP itself does not provide quality of service (QoS), it interoperates with the Resource Reservation Protocol (RSVP) responsible for voice quality. It also collaborates with several other protocols, including Lightweight Directory Access Protocol (LDAP) for location, Remote Authentication Dial-In User Service (RADIUS) for authentication, and RTP for real-time transmission. An important feature of SIP is that it does not define the type of session to be established, but only defines how the session should be managed. With this flexibility, it means that SIP can be used in many applications and services, including interactive games, music and video on demand, as well as voice, video and Web conferencing. SIP messages are text-based, so they are easy to read and debug. The programming of the new service is simpler and more intuitive for designers. SIP reuses MIME type descriptions like an email client, so session-related applications can be started automatically. SIP reuses several existing mature Internet services and protocols, such as DNS, RTP, and RSVP. There is no need to introduce new services to support the SIP infrastructure because many parts of the infrastructure are already in place or ready to use. The expansion of SIP is easy to define and can be added by service providers in new applications without damaging the network. Old SIP-based devices in the network will not hinder new SIP-based services. For example, if the old SIP implementation does not support the method / header used by the new SIP application, it will be ignored. SIP is independent of the transport layer. Therefore, the underlying transmission can be IP using ATM. SIP uses User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) to flexibly connect users independent of the underlying infrastructure. SIP supports multi-device function adjustment and negotiation. If the service or session starts video and voice, you can still transmit voice to devices that do not support video, or you can use other device functions, such as one-way video streaming. Communication providers and their partners and users are increasingly eager for a new generation of IP-based services. Now with SIP (The Session IniTIation Protocol), it is a matter of urgency. SIP was an idea born in a computer science laboratory less than a decade ago. It is the first protocol suitable for a variety of media content to achieve multi-user sessions, and has now become a specification of the Internet Engineering Task Force (IETF). Today, more and more operators, CLEC (competitive local operators) and ITSP (IP telephone service providers) are providing SIP-based services such as local and long-distance telephone technology, online information and instant messaging, IP Centrex / Hosted PBX, voice message, push-to-talk (push to talk), multimedia conference, etc. Independent software vendors (ISVs) are developing new development tools to build SIP-based applications and SIP software for carrier networks. Network equipment vendors (NEVs) are developing hardware that supports SIP signaling and services. Now, many IP phones, user agents, network proxy servers, VOIP gateways, media servers and application servers are using SIP. SIP has evolved from similar authoritative protocols-such as the Web Hypertext Transfer Protocol (HTTP) format protocol and the Simple Mail Transfer Protocol (SMTP) email protocol-and developed into a powerful new standard. However, although SIP uses its own unique user agent and server, it does not work in isolation. SIP supports the provision of converged multimedia services and works in conjunction with many existing protocols responsible for identity verification, location information, voice quality, etc. This white paper gives a general introduction to SIP and its role. It also introduces the process of SIP from laboratory development to market orientation. This white paper explains what services SIP provides and what development initiatives are being implemented. It also introduces in detail the important characteristics of SIP different from various protocols and explains how to establish a SIP session. SIP is more flexible, extensible, and open. It has inspired the power of the Internet and fixed and mobile IP networks to launch a new generation of services. SIP can complete network messages on multiple PCs and phones, simulating the Internet to establish a session. Unlike the long-standing International Telecommunication Union (ITU) SS7 standard (used for call establishment) and the ITU H.323 video protocol combination standard, SIP works independently on the underlying network transmission protocol and media. It specifies how one or more participants ’terminal devices can establish, modify, and terminate connections, regardless of whether they are voice, video, data, or Web-based content. SIP is much better than some existing protocols, such as the Media Gateway Control Protocol (MGCP) that converts PSTN audio signals into IP packets. Because MGCP is a closed pure voice standard, it is more complicated to enhance it through signaling functions, sometimes causing messages to be destroyed or discarded, which prevents providers from adding new services. With SIP, programmers can add a small amount of new information to the message without affecting the connection. For example, SIP service providers can create new media that includes voice, video, and chat content. If MGCP, H.323 or SS7 standards are used, the provider must wait for a new version of the protocol that can support this new media. If SIP is used, although gateways and devices may not recognize the media, companies with branches on the two continents can achieve media transmission. Moreover, because the SIP message construction method is similar to HTTP, developers can more easily and conveniently use common programming languages ​​(such as Java) to create applications. For operators who have been waiting for years to use SS7 and Advanced Intelligent Network (AIN) to deploy call waiting, calling number identification, and other services, if they use SIP, they can now deploy advanced communication services in just a few months. This scalability has achieved significant success in more and more SIP-based services. Vonage is a service provider for users and small business users. It uses SIP to provide users with more than 20,000 digital local, long distance and voice mail lines. Deltathree provides Internet telephony technology products, services and infrastructure to service providers. It provides a PC-to-phone solution based on SIP, enabling PC users to call any phone in the world. Denwa Communications wholesales voice services worldwide. It uses SIP to provide PC-to-PC and phone-to-PC caller number recognition, voice mail, and teleconferencing, unified communications, customer management, self-configuration, and Web-based personalized services. Some authoritative persons expect that the relationship between SIP and IP will develop into a relationship similar to SMTP and HTTP to the Internet, but some people say that it may mark the end of AIN. So far, the 3G community has selected SIP as the session control mechanism for next-generation mobile networks. Microsoft has selected SIP as its real-time communication strategy and has deployed it in Microsoft XP, Pocket PC and MSN Messenger. Microsoft also announced that the next version of CE dot net will use the SIP-based VoIP application interface layer and promises to provide SIP-based voice and video calls to user PCs. In addition, MCI is using SIP to deploy advanced telephony services to IP communications users. The user will be able to inform the caller if he is available and the preferred method of communication, such as e-mail, phone or instant messaging. Using online information, users can also instantly establish chat sessions and hold audio conferences. Using SIP will continuously realize various functions. SIP session composition A SIP session uses up to four main components: SIP user agent, SIP registration server, SIP proxy server, and SIP redirect server. These systems complete the SIP session by transmitting messages that include the SDP protocol (used to define the content and characteristics of the message). The following is an overview of each SIP component and its role in this process. SIP user agent SIP User Agents (UA) are end-user devices, such as mobile phones, multimedia handheld devices, PCs, PDAs, etc. used to create and manage SIP sessions. The user agent client issues a message. The user agent server responds to the message. SIP registration server The SIP registration server is a database that contains the locations of all user agents in the domain. In SIP communication, these servers retrieve the participant's IP address and other related information and send it to the SIP proxy server. SIP proxy server The SIP proxy server accepts the SIP UA session request and queries the SIP registration server to obtain the address information of the recipient UA. Then, it forwards the session invitation information directly to the recipient UA (if it is in the same domain) or the proxy server (if the UA is in another domain). SIP redirect server The SIP redirect server allows the SIP proxy server to direct SIP session invitation information to external domains. The SIP redirect server can be on the same hardware as the SIP registration server and SIP proxy server. The following scenarios illustrate how SIP components can coordinate to establish SIP sessions between UAs in the same domain and different domains: Establish a SIP session in the same domain The following diagram illustrates the process of establishing a SIP session between two users who subscribe to the same ISP and use the same domain. User A uses a SIP phone. User B has a PC running a soft client program that supports voice and video. After power-on, both users registered their availability and IP address on the SIP proxy server in the ISP network. User A initiates this call and tells the SIP proxy server to contact user B. Then, the SIP proxy server sends a request to the SIP registration server to request the IP address of user B, and receives the IP address of user B. The SIP proxy server forwards the invitation information (using SDP) for the communication between user A and user B, including the media to be used by user A. User B informs the SIP proxy server that it can accept User A's invitation and is ready to receive messages. The SIP proxy server communicates this message to user A, thereby establishing a SIP session. Then, the user creates a point-to-point RTP connection to achieve interactive communication between users. 2. Query where B 3. SIP address of response B 4. Call 5. Response 6. Response 7. Multimedia channel has been established Establish SIP sessions in different domains The difference between this scenario and the first scenario is as follows. When user A invites user B who is using a multimedia handheld device for a SIP session, the SIP proxy server in domain A recognizes that user B is not in the same domain. Then, the SIP proxy server queries user B's IP address on the SIP redirect server. The SIP redirect server can be in domain A, domain B, or both domain A and domain B. The SIP redirect server feeds back the contact information of user B to the SIP proxy server, which then forwards the SIP session invitation information to the SIP proxy server in domain B. The SIP proxy server in domain B sends user A's invitation information to user B. User B then forwards the information for accepting the invitation along the same path through which the invitation information passes. 2. Ask the user in B 3. Response 4. Call the SIP proxy of domain B 5. Query where B 6. User B's address 7. Agent call 8. Response 9. Response 10. Response 11. Multimedia channel has been established Seamless, flexible, and scalable: looking forward to the future of SIP SIP can connect users using any IP network (wired LAN and WAN, public Internet backbone, mobile 2.5G, 3G, and Wi-Fi) and any IP device (telephone, PC, PDA, mobile handheld device) New lucrative business opportunities have improved the way companies and users communicate. SIP-based applications (such as VOIP, multimedia conferencing, push-to-talk (push-to-talk), location services, online information, and IM), even if used alone, will provide service providers, ISVs, network equipment vendors, and developers Many new business opportunities. However, the fundamental value of SIP lies in its ability to combine these functions to form a variety of larger-scale seamless communication services. Using SIP, service providers and their partners can customize and provide SIP-based combined services, enabling users to use services such as conference, Web control, online information, IM, etc. in a single communication session. In fact, service providers can create a flexible application portfolio that meets the needs of multiple end users, instead of installing and supporting a single decentralized application that relies on limited specific functions or types of terminal devices. By merging IP-based communication services under a single, open standard SIP application architecture, service providers can greatly reduce the cost of designing and deploying new and innovative IP-based managed services for users. It is a powerful impetus for SIP scalability to promote the development of this industry and market, and is the hope of all of us. Easy Electronic Technology Co.,Ltd , https://www.nbpcelectronicgroup.com
1. Call User B 
1. Call User B