SIP's most popular communication protocols are mature

Introduction

Communication providers and their partners and users are increasingly eager for a new generation of IP-based services. Now we have the SIP Session Initiation Protocol. SIP was born less than a decade ago in the computer science laboratory.

It is the first protocol suitable for multi-user sessions in various media content. Now it has become the specification of the Internet Engineering Task Group (IETF.

Today, more and more carriers, CLEC competitors, and ITSPIP 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 messaging, push-to-talk, and Multimedia conferences. Independent software vendors (ISVs) are developing new development tools to build SIP-based applications and SIP software for carrier networks. Network equipment suppliers (NEV) are developing hardware that supports SIP signaling and services. Currently, many IP phones, user agents, network proxy servers, VOIP gateways, media servers, and application servers are using SIP.

SIP uses similar authoritative protocols, such as Web Hypertext Transfer Protocol (HTTP) formatting and Simple Mail Transfer Protocol (SMTP) email protocols-evolved and evolved into a new and powerful standard. However, although SIP uses its own unique user proxy and server, it does not work in an integrated manner. SIP supports integrated multimedia services and works with a large number of existing protocols for identity authentication, location information, and voice quality.

This White Paper provides a general introduction to SIP and its functions. It also introduces the process of SIP from lab development to market-oriented. This White Paper describes what services are provided by SIP and what development programs are being implemented. It also details the important features of different SIP protocols and describes how to establish a SIP session.

Next-generation services

SIP is flexible, scalable, and open. It inspires the power of a new generation of Services launched by the Internet and fixed and mobile IP networks. SIP can complete network messages on multiple PCs and phones and simulate Internet sessions.

Unlike the long-standing International Telecommunication Union (ITU) SS7 standard for call establishment) and the ITU H.323 Video protocol combination standard, SIP works independently of the underlying network transmission protocol and media. It specifies how terminal devices of one or more participants can establish, modify, and interrupt connections, regardless of 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 the PSTN audio signal into IP packets ). Because MGCP is a closed voice standard, it is complicated to enhance it through the signaling function. Sometimes, the message is damaged or discarded, which prevents the provider from adding new services. With SIP, programmers can add a small amount of new information to messages without affecting connections.

For example, a SIP service provider can create a new media that contains voice, video, and chat content. If the MGCP, H.323, or SS7 standard is used, the provider must wait for a new version of the protocol that supports this new media. If you use SIP, although the gateway and device may not be able to recognize the media, companies with branches on two continents can achieve media transmission.

In addition, because the SIP Message construction method is similar to HTTP, developers can easily use a common programming language such as Java to create applications. For carriers that want to use SS7 and advanced intelligent network (AIN) to deploy call wait, caller ID Recognition, and other services after several years, if you use SIP, deployment of advanced communication services takes only a few months.

This scalability has already 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 over 20,000 digital city calls, long calls, and voice mail lines. Deltathree provides Internet telephone technology products, services and infrastructure for service providers. It provides a SIP-based PC-to-phone solution that enables PC users to call any phone number in the world. Denwa Communications wholesale voice services worldwide. It uses SIP to provide caller identification, voice mail, teleconference, Unified Communication, customer management, self-configuration, and Web-based personalized services from PC to PC and telephone to PC.

Some authority predict that the relationship between SIP and IP will develop into a relationship similar to SMTP and HTTP and Internet, but some people say that it may mark the end of AIN. So far, the 3G community has selected SIP as the next-generation mobile network session control mechanism. Microsoft has selected SIP as its real-time communication policy and deployed it in Microsoft XP, Pocket PC, and MSN Messenger. Microsoft also announced that the next version of CE.net will use the SIP-based VoIP application interface layer and promise to provide SIP-based voice and video calls to users' PCs.

In addition, MCI is using SIP to deploy advanced telephone technical services to IP users. The user will be able to notify the caller whether they are free and the preferred communication method, such as email, phone or instant message. With online information, users can also instantly establish chat sessions and hold audio meetings. Using SIP will continuously implement various functions.

Historical Review

SIP appeared in the middle of 1990s and originated from research by Henning Schulzrinne, associate professor of computer science at Columbia University, and its research team. Professor Schulzrinne not only proposed a real-time data transmission protocol (RTP) over the Internet, but also developed a standard proposal for real-time stream transmission protocol (RTSP, it is used to control the stream transmission of audio and video content on the Web.

Schulzrinne originally planned to write multi-party multimedia session control (MMUSIC) standards. In October 1996, he submitted a draft to IETF, which included the important content of SIP. Shulzrinne deleted irrelevant content related to media content in the new standards submitted on April 9, 1999. Subsequently, IETF released the first SIP specification, RFC 2543. Although some vendors have expressed concern that the H.323 and MGCP protocols may greatly compromise their investment in the SIP service, IETF continued to do so and released the SIP specification RFC 2001 in 3261.

The release of RFC 3261 marks the establishment of the SIP Foundation. Since then, several additional versions of RFC have been released to enrich content in security, identity authentication, and other fields. For example, RFC 3262 specifies the reliability of the temporary response. RFC 3263 establishes the rules for locating the SIP proxy server. RFC 3264 provides a proposal/response model, and RFC 3265 identifies specific event notifications.

As early as 2001, suppliers began to launch SIP-based services. Today, people are enthusiastic about the agreement. Organizations such as Sun Microsystems's Java Community Process are using a common Java programming language to define application programming interfaces (APIS) so that developers can build SIP components and applications for service providers and enterprises. Most importantly, more and more competitors are using promising new services to enter the SIP market. SIP is becoming one of the most important protocols since HTTP and SMTP.

Advantages of SIP: Web-like scalable open communication

With SIP, service providers can select standard components at will to quickly control new technologies. Regardless of the number of media content and participants, users can find and contact each other. SIP negotiates sessions so that all participants can agree on and modify session functions. It can even add, delete, or transfer users.

However, SIP is not omnipotent. It is neither a Session Description Protocol nor a conference control function. To describe the load conditions and features of message content, SIP uses the Internet Session Description Protocol (SDP) to describe the characteristics of terminal devices. SIP itself does not provide quality of service (QoS). It is interoperable with the resource retention setting Protocol (RSVP) responsible for voice quality. It also cooperates with several other protocols, including Lightweight Directory Access Protocol (LDAP) for locating and remote identity authentication for Identity Authentication Dial-In User Service (RADIUS) and RTP and other Protocols responsible for real-time transmission.

SIP defines the following basic communication requirements:

1. User positioning service
2. Session Creation
3. Session participant Management
4. Limited identification of features

An important feature of SIP is that it does not define the type of session to be established, but only defines how to manage sessions. With this flexibility, it means that SIP can be used in a wide range of applications and services, including interactive games, music and video on demand, as well as voice, video, and Web conferencing.

Below are some of the outstanding features of SIP in the new signaling protocol.

SIP messages are text-based and therefore easy to read and debug. New Service programming is simpler and more intuitive for designers.

The MIME type description is reused as the e-mail client, so session-related applications can be started automatically.

SIP reuse 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 of the infrastructure is in place or ready to use.

The expansion of SIP is easy to define. It can be added by the service provider to a new application without damaging the network. The old SIP-based devices in the network will not impede new services based on the SIP. 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 transmission layer. Therefore, the underlying transmission can use an atm ip address. 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 the video and voice, you can still transmit the voice to a device that does not support the video, or use other device functions, such as one-way video stream transmission.

SIP session Composition

SIP sessions use up to four main components: SIP User Agent, SIP registration server, SIP proxy server, and SIP redirection server. These systems complete the SIP session by transmitting messages that include the content and features of the SDP protocol used to define the message. The following describes the various SIP components and their functions in this process.

The SIP User Agent (UA) is an end user device, such as a mobile phone, multimedia handheld device, PC, and PDA used to create and manage SIP sessions. The user agent client sends a message. The user proxy server responds to messages.

The SIP registration server is a database that contains all user proxies in the domain. In SIP Communication, these servers retrieve the IP address and other information of the participant and send it to the SIP proxy server.

The SIP Proxy Server accepts the sip ua Session Request and queries the SIP registration server to obtain the UA address information of the recipient. It then forwards the session invitation information directly to the recipient's UA if it is in the same domain) or the proxy server if the UA is in another domain ).

The SIP redirection server allows the SIP proxy server to direct the SIP session invitation information to an external domain. The SIP redirection server can be on the same hardware as the SIP registration server and the SIP proxy server.

The following scenarios describe how to coordinate SIP components to establish a SIP session between UA in the same domain and different domains:

Establish a SIP session in the same domain

Describes how to create a SIP session between two users in the same domain by subscribing to the same ISP. User A uses the SIP Phone. User B has a PC running soft client programs that support voice and video. After power-on, both users have registered their idle status 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 provide the IP address of user B and receive the IP address of user B. The SIP Proxy Server forwards user A's invitation to communicate with user B using SDP), including the media to be used by user. User B notifies the SIP proxy server to accept the invitation from user A and is ready to receive the message. The SIP proxy server sends the message to user A to establish a sip session. Then, the user creates a point-to-point RTP connection to implement interactive communication between users.

1. Call the user
2. Where is the query?
3. Response to the SIP address call
5. Response
6. Response
7. The multimedia channel has been created

Establish a SIP session 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, to perform a sip Session, the SIP proxy server in Domain A identifies that user B is not in the same domain. Then, the SIP Proxy Server queries the IP address of user B on the SIP redirection server. The SIP redirection server can be in Domain A, Domain B, or Domain A and Domain B. The SIP redirect server sends user B's contact information 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 that accepts the invitation along the same path of the invitation information.

Seamless, flexible, and scalable: Future of SIP

SIP can be connected to any IP network, including wired LAN and WAN, public Internet backbone network, mobile 2.5G, 3G, and Wi-Fi) and any IP device phone, PC, PDA, mobile handheld device) users, and thus a large number of lucrative new business opportunities, improved the communication between enterprises and users. SIP-based applications such as VOIP, Multimedia conferences, push-to-talk Key calls), locating services, online information, and IM) even if they are used separately, it also provides many new business opportunities for service providers, ISVs, network equipment providers, and developers. However, the fundamental value of SIP is that it can combine these functions to form a variety of large-scale seamless communication services.

With SIP, service providers and their partners can customize and provide SIP-based combined services so that users can use conference, Web control, online information, IM and other services in a single communication session. In fact, a service provider can create a flexible combination of applications that meet the needs of multiple end users, rather than installing and supporting a single distributed application that relies on the limited functionality or types of terminal devices.

By combining IP-based communication services under a single and open standard SIP application architecture, service providers can greatly reduce the cost of designing and deploying new IP-based hosting services for users. It is the powerful motive force for SIP scalability to promote the development of the industry and the market, and the hope of all of us.