Building Telephony Systems with OpenSER

OpenSER是一款基于SIP协议的灵活、免费开源VoIP服务器,适用于创建、修改及终止包含互联网电话呼叫在内的多媒体会话。它专为大规模IP电信基础设施而设计,采用纯C语言编写,并针对Linux/Unix-like系统进行了架构特定优化,能够支持高达4百万用户的同时在线。该服务器还具备用户跟踪、VoIP会话设置、即时消息中继等功能。

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OpenSER is a flexible, free open-source VoIP server based on the Session Initiation Protocol (SIP), an application-layer control (or signaling) protocol for creating, modifying, and terminating sessions with one or more participants, including internet telephone calls, multimedia distribution, and multimedia conferences.

Engineered to power IP telephony infrastructures up to large scale, OpenSER is written in pure C for Linux/Unix-like systems with architecture-specific optimizations to offer high performance; it is able to handle 4 million users on a single processor server. The server keeps track of users, sets up VoIP sessions, relays instant messages, and creates space for new plug-in applications.

It can be used on systems with limited resources as well as on carrier-grade servers, scaling up to thousands of call setups per second. It is customizable, being able to feature as fast load balancer; SIP server flavors: registrar, location server, proxy server, redirect server; gateway to SMS/XMPP; or advanced VoIP application server.


http://rapidshare.com/files/113092197/1847193730.zip
openSER 完整介绍 We are starting a new era in the collaboration area. Voice and Video over IP are starting to dominate the world of telecommunications in a disruptive movement capable of changing the whole industry. The SIP (Session Initiation protocol) technology is at the center of this revolution. I believe, at present, SIP is the most used protocol for Voice and Video over IP. In the future, when people learn how to use the technology, SIP will be for voice communications what email is today for text communications. We are starting with islands of SIP communications inside VoIP providers, enterprises, and even governments. In the near future, the barriers between the islands will be broken and you will be able to communicate with anyone in anyplace without paying high fees. The only fees you will pay in the future will be the access to the data network, because with the pervasiveness of VoIP and Video over IP, everything will be simply data. I remember the frst days of Internets email in the early 90s. It took some years until everyone had an email address. The same thing could be happening in the SIP world now. Unfortunately, the SIP providers still behave as islands not, usually, allowing free inter-domain routing. With the introduction of 3G, 4G, and WiMAX, fast data communications are becoming widespread in the mobile industry. Newer phones from mainstream manufacturers are starting to support WiFI, WiMAX, and obviously 3G. SIP clients can run in these platforms changing the whole mobile communication industry in the near future. Sure, the telephone companies will try, legitimately, to protect their revenue sources, but they cannot hide for ever the SIP communication infrastructure already in place. Slowly, users will start to use SIP clients in their mobile phones hugely cutting the communication costs. The movement, even without a huge sponsor, will spread by word of mouth until it becomes pervasive.
标题基于SpringBoot+Vue的社区便民服务平台研究AI更换标题第1章引言介绍社区便民服务平台的研究背景、意义,以及基于SpringBoot+Vue技术的研究现状和创新点。1.1研究背景与意义分析社区便民服务的重要性,以及SpringBoot+Vue技术在平台建设中的优势。1.2国内外研究现状概述国内外在社区便民服务平台方面的发展现状。1.3研究方法与创新点阐述本文采用的研究方法和在SpringBoot+Vue技术应用上的创新之处。第2章相关理论介绍SpringBoot和Vue的相关理论基础,以及它们在社区便民服务平台中的应用。2.1SpringBoot技术概述解释SpringBoot的基本概念、特点及其在便民服务平台中的应用价值。2.2Vue技术概述阐述Vue的核心思想、技术特性及其在前端界面开发中的优势。2.3SpringBoot与Vue的整合应用探讨SpringBoot与Vue如何有效整合,以提升社区便民服务平台的性能。第3章平台需求分析与设计分析社区便民服务平台的需求,并基于SpringBoot+Vue技术进行平台设计。3.1需求分析明确平台需满足的功能需求和性能需求。3.2架构设计设计平台的整体架构,包括前后端分离、模块化设计等思想。3.3数据库设计根据平台需求设计合理的数据库结构,包括数据表、字段等。第4章平台实现与关键技术详细阐述基于SpringBoot+Vue的社区便民服务平台的实现过程及关键技术。4.1后端服务实现使用SpringBoot实现后端服务,包括用户管理、服务管理等核心功能。4.2前端界面实现采用Vue技术实现前端界面,提供友好的用户交互体验。4.3前后端交互技术探讨前后端数据交互的方式,如RESTful API、WebSocket等。第5章平台测试与优化对实现的社区便民服务平台进行全面测试,并针对问题进行优化。5.1测试环境与工具介绍测试
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