Summary of sip Port

Source: Internet
Author: User

Through the study of the SIP protocol, we have learned a variety of applications of the SIP protocol. Here we will describe the content of the SIP protocol port. First, we will introduce the concept of SIP to help you review it. The Session Initiation Protocol (sip) is a hierarchical structure Protocol. Its behavior is described based on a set of equal and independent processing stages. Each stage is loosely coupled. the sip protocol port adopts the C/S structure and uses the message mechanism to establish a session. the sip protocol port is very suitable for real-time communication applications in the Internet. Its Design Concept and protocol structure fully comply with the characteristics and requirements of NGN and will become the mainstream development direction of multimedia communication systems in the future. the core idea of the sip protocol port is to invite new members to join an existing session or create a new session. Each member can use multicast) and unicast) or a combination of the two.

In a VoIP system, the sip protocol port is widely used. In addition to the sip protocol port, the VoIP protocol mainly includes H.323 and Media Gateway Control Protocol MGCP. MGCP can connect a large number of IP Phone Gateways to a whole with interoperability, which is especially suitable for configuring large-scale application systems. However, it is too complicated for small application systems. h.323 is designed for circuit switching networks, and the protocol is relatively complex. The VoIP service using H.323 has high requirements on terminal devices. sip is simple and easy to integrate with other services, with obvious advantages.

What is the role of the sip protocol port? The sip port is generally used to generate, modify, and end sessions between one or more participants. these sessions include Internet Multimedia conferences, Internet or any IP Network) telephone calls and multimedia Publishing. the members of a session can communicate with each other through multicast or unicast networks.

Functional entities and workflow of the sip protocol port

The sip protocol port adopts the C \ S mode and uses the message mechanism to establish and manage sessions. based on logical functions, the SIP system can be divided into four functional entities: SIP User proxy, SIP proxy server, SIP redirection server, and SIP registration server. These constitute the basic model of the sip protocol port.

Sip protocol port user agent sip ua): it is also called a SIP terminal. It is the end user in the SIP System and is defined as an application in rfc3261. based on the roles they play in sessions, they can be divided into user proxy client (UAC) and user Proxy Server (UAS. the former is used to initiate a call request, and the latter is used to contact the user when a SIP request is received and return a response on behalf of the user.

Sip Proxy Server: An intermediate element. It is both a client and a Server that can send a call request to the Server at the next hop. in addition to the routing capability, the SIP proxy server can also integrate functions such as firewall and radiusAAA.

Sip Redirect Server: it is a Server that plans the SIP call path. After obtaining the next hop address, immediately tell the preceding user, let the user send a request directly to the next hop address and exit the control of the call.

Sip Register Server: Used to log on to UAS. All UAS instances must be logged on to a specific logon Server, so that UAC can find them through the server. the registration service does not determine the requested identity authentication. in SIP, authorization and authentication can be achieved through context-related requests established in the request/response mode, or through a more underlying approach.

The bottom layer of the sip protocol port is syntax and encoding. its Encoding is defined using the enhanced Backus-Nayr format syntax BNF. used to initiate a session. It can be used to create, modify, and end multimedia session processes attended by multiple participants. participants can communicate with each other through multicast, unicast, or network connection. to provide multimedia services, different standards and protocols are also required. For example, RTP is required to ensure media transmission, RSVP ensures voice quality, and RADIUS Authentication Users.

Contact Us

The content source of this page is from Internet, which doesn't represent Alibaba Cloud's opinion; products and services mentioned on that page don't have any relationship with Alibaba Cloud. If the content of the page makes you feel confusing, please write us an email, we will handle the problem within 5 days after receiving your email.

If you find any instances of plagiarism from the community, please send an email to: info-contact@alibabacloud.com and provide relevant evidence. A staff member will contact you within 5 working days.

A Free Trial That Lets You Build Big!

Start building with 50+ products and up to 12 months usage for Elastic Compute Service

  • Sales Support

    1 on 1 presale consultation

  • After-Sales Support

    24/7 Technical Support 6 Free Tickets per Quarter Faster Response

  • Alibaba Cloud offers highly flexible support services tailored to meet your exact needs.