What is VoIP?

VOIP - Voice Over Internet Protocol

Internet Voice, also known as Voice over Internet Protocol (VoIP), is a technology that allows you to make telephone calls using a broadband Internet connection instead of a regular (or analog) phone line. Some services using VoIP may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some services only work over your computer or a special VoIP phone, other services allow you to use a traditional phone through an adaptor.

Today we can see a real revolution in communication world: everybody begins to use PCs and Internet for job and free time to communicate each other, to exchange data (like images, sounds, documents) and, sometimes, to talk each other using applications like Netmeeting or Internet Phone. Particularly starts to diffusing a common idea that could be the future and that can allow real-time vocal communication: VoIP.

What is VoIP?

VoIP stands for Voice over Internet Protocol. As the term says VoIP tries to let go voice (mainly human) through IP packets and, in definitive through Internet. VoIP can use accelerating hardware to achieve this purpose and can also be used in a PC environment.

How does it work?

Many years ago we discovered that sending a signal to a remote destination could have be done also in a digital fashion: before sending it we have to digitalize it with an ADC (analog to digital converter), transmit it, and at the end transform it again in analog format with DAC (digital to analog converter) to use it.

VoIP works like that, digitalizing voice in data packets, sending them and reconverting them in voice at destination.

Digital format can be better controlled: we can compress it, route it, convert it to a new better format, and so on; also we saw that digital signal is more noise tolerant than the analog one (see GSM vs TACS).

TCP/IP networks are made of IP packets containing a header (to control communication) and a payload to transport data: VoIP use it to go across the network and come to destination.

What is the advantages using VoIP rather PSTN?

When you are using PSTN line, you typically pay for time used to a PSTN line manager company: more time you stay at phone and more you'll pay. In addition you couldn't talk with other that one person at a time.

In opposite with VoIP mechanism you can talk all the time with every person you want (the needed is that other person is also connected to Internet at the same time), as far as you want (money independent) and, in addition, you can talk with many people at the same time.

If you're still not persuaded you can consider that, at the same time, you can exchange data with people are you talking with, sending images, graphs and videos.

网络上 VoIP 的定义
IP电话(简称VoIP，源自英语Voice over Internet Protocol；又名宽带电话或网络电话)是一种透过因特网或其他使用IP技术的网络，来实现新型的电话通讯。过去IP电话主要应用在大型公司的内联网内，以让技术管理人员可以同时以一个网络提供数据及语音服务，简化管理之余，更可提高生产力。随着因特网日渐普及，以及跨境通讯数量大幅飙升，IP电话亦被应用在长途电话业务上。由于世界各主要大城市的通信公司竞争日剧，以及各国电信相关法令松绑，IP电话也开始应用于固网通信，其低通话成本、低建设成本、易扩充性及日渐优良化的通话质量等主要特点，被目前国际电信企业看成是传统电信业务的潜在有力竞争者。

IP电话的技术

IP电话通过把语音信号经过数字处理、压缩编码，在网络上传输，然后再解压、把数字信号还原成声音，让通话对方听到。

话音从源端到达目的端的基本过程是：
声电转换：通过压电陶瓷等类似装置将声波变换为电信号
量化采样：将模拟电信号按照某种采样方法（比如脉冲编码调制，即PCM）转换成数字信号
封包：将一定时长的数字化之后的语音信号组合为一帧，随后，按照国际电联（ITU-T）的标准，这些话音帧被封装到一个RTP（即实时传输协议，Realtime Transport Protocol）报文中，并被进一步封装到UDP报文和IP报文中。
传输：IP报文在IP网络由源端传递到目的端
去抖动：去除因封包在网络中传输速度不均匀所造成的抖动音
拆包
电声转换

一个完整的、可以大规模商用运营的IP电话系统包括如下一些技术(暂不完全)：
寻址
话音编码
回声消除和回声抑制
传输
IP报文时延控制功能
去抖动
IP报文的去抖动(de-jitter)功能

话音编码方案

目前世界多个标准组织和工业实体提出了很多话音编码方案。其中包括国际电信联盟的G.711(速率64kbps)，G.723.1(速率5.3kbps或者6.3kbps)，G.729A(速率8kbps)编码方案。微软、Intel等业界巨头也有自己的编码方案。

VoIP 控制协议

目前常用的协议如 H.323、SIP、MEGACO 和 MGCP。

H.323 是一种 ITU-T 标准，最初用于局域网（LAN）上的多媒体会议，后来扩展至覆盖 VoIP 。该标准既包括了点对点通信也包括了多点会议。 H.323 定义了四种逻辑组成部分：终端、网关、关守及多点控制单元（MCU）。终端、网关和 MCU 均被视为终端点。

会话发起协议（SIP）是建立 VOIP 连接的 IETF 标准。 SIP 是一种应用层控制协议，用于和一个或多个参与者创建、修改和终止会话。 SIP 的结构与 HTTP （客户－服务器协议）相似。客户机发出请求，并发送给服务器，服务器处理这些请求后给客户机发送一个响应。该请求与响应形成一次事务。

媒体网关控制协议（MGCP）是由思科和 Telcordia 提议的 VoIP 协议，它定义了呼叫控制单元（呼叫代理或媒体网关）与电话网关之间的通信服务。 MGCP 属于控制协议，允许中心控制台监测 IP 电话和网关事件，并通知它们发送内容至指定地址。在 MGCP 结构中，智能呼叫控制置于网关外部并由呼叫控制单元（呼叫代理）来处理。同时呼叫控制单元互相保持同步，发送一致的命令给网关。

媒体网关控制协议（Megaco）是 IETF 和 ITU-T （ITU-T H.248 建议）共同努力的结果。 Megaco/H.248 是一种用于控制物理上分开的多媒体网关的协议单元的协议，从而可以从媒体转化中分离呼叫控制。 Megaco/H.248 说明了用于转换电路交换语音到基于包的通信流量的媒体网关（MG）和用于规定这种流量的服务逻辑的媒介网关控制器之间的联系。 Megaco/H.248 通知媒体网关将来自于数据包或单元数据网络之外的数据流连接到数据包或单元数据流上，如实时传输协议（RTP）。从 VoIP 结构和网关控制的关系来看， Megaco/H.248 与 MGCP 在本质上相当相似，但是 Megaco/H.248 支持更广泛的网络，如 ATM 。
　

VoIP (Voice over Internet Protocol) is simply the transmission of voice traffic over IP-based networks.

The Internet Protocol (IP) was originally designed for data networking. The success of IP in becoming a world standard for data networking has led to its adaption to voice networking.

秋水轩由秋长水倾情制作 QQ：234626762 群：8184701