eefocus术语辞典
首页 |
中文解释:IP电话、宽频电话、网络电话，IP上的话音
英文缩写:VoIP
英文来历:Voice Over IP
使用包链接路由在因特网上发送常规电话话音的技术。VoIP并不只是简单的IP上的话音，而且能适应双向视频会议和共享的应用。
VoIP是指将模拟的声音讯号经过压缩与封包之后，以数据封包的形式在IP 网络的环境进行语音讯号的传输，通俗来说也就是互联网电话、网络电话或者简称IP电话的意思。VoIP的基本原理是：...
使用包链接路由在因特网上发送常规电话话音的技术。VoIP并不只是简单的IP上的话音，而且能适应双向视频会议和共享的应用。
通常，采用VOIP的电话服务业务花费比利用传统资源的少很多， 节省下来开支的主要是因为采用单一网络传输声音与数据， 特别是它可以利用用户目前现有设备利用不充分的带宽而不会产生额外的花费。有一点必须注意的是你的计算机网络连接过程中的最大上行流不会像标准电信公司提供的服务那样而是成为你的决定性障碍。 用IP电话打电话时（包括国际电话）一般被认为是免费的，但是它包括一个互联网服务的费用，通过这个服务使用IP电话一般不会包括任何额外的费用，所以用户就会认为它是免费的。那有许多提供这种方便“免费”服务的供应商,例如; Google Talk, Skype and TheGlobe.
VoIP使得许多使用传统通话网络复杂的事变得简单易用：
不管你在什么地方，来电都可以通过网络选择路径转移到你的IP电话上。所以只有你可以和网络连接，就算是携带着电话旅游，也能收到呼入电话
VoIP technology still has a few shortcomings that have led some to believe that it is not ready for widespread deployment. However, many industry analysts predicted that 2005 was the "Year of Inflection," where more IP PBX ports shipped than legacy digital PBX ports.
Because IP does not provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, VoIP implementations may face problems dealing with latency (especially if satellite circuits are involved), and jitter. They are faced with the problem of restructuring streams of received IP packets, which can come in any order and have packets delayed or missing, to ensure that the ensuing audio stream maintains a proper time consistency.
Another main challenge is routing VoIP traffic to traverse certain firewalls and NAT. Intermediary devices called Session Border Controllers (SBC) are often used to achieve this, though some proprietary systems such as Skype traverse firewall and NAT without a SBC by using users' computers as super node servers to route other people's calls.
Keeping packet latency acceptable can also be a problem, due to network routing time (buffering, switching) and transmission distances (more relevant under satellite links).
Conventional telephones are connected directly to telephone company phone lines, which in the event of a power failure are kept functioning by back-up generators or batteries located at the telephone exchange. However, household VoIP hardware uses broadband modems and other equipment powered by household electricity, which may be subject to outages. In order to use VoIP during a power outage, an uninterruptible power supply or a generator must be installed on the premises. It should be noted that many early adopters of VoIP are also users of other phone equipment such as PBX and cordless phone bases that also rely on power not provided by the telephone company.
Some broadband connections may have less than desirable reliability. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there is long distances and/or interworking between end points.
The nature of IP makes it difficult to geographically locate network users. Emergency calls, therefore, can not easily be routed to a nearby call center, and are impossible on some VoIP systems. Moreover, in the event that the caller is unable to give an address, emergency services may be unable to locate them in any other way. Following the lead of mobile phone carriers, several VoIP carriers are already implementing a technical work-around. The United States government had set a deadline, requiring VoIP carriers to implement e911, however, the deadline is being appealed by several of the leading VoIP companies. This is a different situation with IPBX systems, where these corporate systems often have full e911 capabilities built into the system. A simple solution to this problem is to store the local emergency numbers on speed dial which is usually even faster than having to be transferred by the 911 operator.
Whilst the traditional Plain Old Telephone System (POTS) and mobile phone networks share a common global standard (E.164) which allocates and identifies any specific telephone line, there is no widely adopted similar standard for VoIP networks. Some allocate an E.164 number which can be used for VoIP as well as incoming/external calls. However, there are often different, incompatible schemes when calling between VoIP providers which use short codes that are provider specific.
With commercial services such as Vonage, it is possible to connect the VoIP router into the existing central phone box in the house and have VoIP at every phone already connected. Other services, such as Skype & PeerMe require the use of a computer, so they are limited to single point of calling. Some services, such as BroadVoice provide the ability to connect WiFi SIP phones so that service can be extended throughout the premises, and off-site to any location with an open hotspot..
Telcos and consumers have invested billions of dollars in mobile phone equipment. In developed countries, mobile phones have achieved nearly complete market penetration, and many people are giving up landlines and using mobiles exclusively. Given this situation, it is not entirely clear whether there would be a significant higher demand for VoIP among consumers until either a) public or community wireless networks have similar geographical coverage to cellular networks (thereby enabling mobile VoIP phones, so called WiFi phones) or b) VoIP is implemented over legacy 3G networks. However, "dual mode" handsets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular.
A major development starting in 2004 has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. This requires an analog telephone adapter (ATA) to connect a telephone to the broadband Internet connection. Full phone service VoIP phone companies provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited calling to the U.S., and sometimes to Canada or to selected countries in Europe or Asia, for a flat monthly fee. One advantage of this is the ability to make and receive calls as one would at home, anywhere in the world, at no extra cost. As calls go via IP, this does not incur charges as call diversion does via the PSTN, and the called party does not have to pay for the call.
For example, somebody may call someone on a number with a U.S. area code, but one could be in London, and if someone were to call another number with that area code, it would be treated as a local call, regardless of where that person is in the world. However, the broadband phone is likely to complement, rather than replace a PSTN line, as it still needs a power supply, while calling the U.S. emergency services number 911, may not automatically be routed to the nearest local emergency dispatch center, and would be of no use for subscribers outside the U.S.
Another challenge for these services is the proper handling of outgoing calls from fax machines, TiVo/ReplayTV boxes, satellite television receivers, alarm systems, conventional modems or FAXmodems, and other similar devices that depend on access to a voice-grade telephone line for some or all of their functionality. At present, these types of calls sometimes go through without any problems, but in other cases they will not go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower bits per second rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional voice-grade telephone line would be available in almost all homes in North America and Western-Europe. The TestYourVoIP website offers a free service to test the quality of or diagnose an Internet connection by placing simulated VoIP calls from any Java-enabled Web browser, or from any phone or VoIP device capable of calling the PSTN network.
Although few office environments and even fewer homes use a pure VoIP infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect switching stations, converting voice signals to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes.
Corporate customer telephone support often use IP telephony exclusively to take advantage of the data abstraction. The benefit of using this technology is the need for only one class of circuit connection and better bandwidth use. Companies can acquire their own gateways to eliminate third-party costs, which is worthwhile in some situations.
VoIP is widely employed by carriers, especially for international telephone calls. It is commonly used to route traffic starting and ending at conventional PSTN telephones.
Many telecommunications companies are looking at the IP Multimedia Subsystem which will merge Internet technologies with the mobile world, using a pure VoIP infrastructure. It will enable them to upgrade their existing systems while embracing Internet technologies such as the Web, email, instant messaging, presence, and video conferencing. It will also allow existing VoIP systems to interface with the conventional PSTN and mobile phones.
Electronic Numbering (Enum) uses standard phone numbers (E.164), but allows connections entirely over the Internet. If the other party uses Enum, the only expense is the Internet connection.
Amateur radio has adopted VOIP by linking repeaters and users with Echolink, IRLP, Dstar and EQSO. Echolink and IRLP are programs/systems based upon the Speak Freely VOIP open source software. By using VOIP Amateur Radio operators are able to create large repeater networks with repeaters all over the world where operators can access the system with actual ham radios.
Ham Radio operators using radios are able to tune to repeaters with VOIP capabilities and use DTMF buttons to command the repeater to connect to various other repeaters.
As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, governments are becoming more interested in regulating VoIP in a manner similar to PSTN services.
In the US, the FCC now requires all VoIP operators who don't support Enhanced 911 to attach a sticker warning that traditional 911 services aren't available. The FCC recently required VoIP operators to support CALEA wiretap functionality [1]. The Telecommunications Act of 2005 proposes adding more traditional PSTN regulations, such as local number portability and universal service fees.
Most standards-based solutions use either the H.323 or Session Initiation Protocol (SIP) protocols. A number of proprietary designs also exist.
信令协议:
Several different 语音编解码 can be used for stream audio compression. Commonly used codecs for VoIP traffic include G.711, G.723.1 and G.729, all ITU-T-specified.
将声音数据数字化后在 TCP / IP 网络上传输的技术。一般用来表示 Internet 电话等的实现方法