Voice over IP
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Voice over IP (also called VoIP, IP Telephony, and Internet telephony) refers to technology that enables routing of voice conversations over the Internet or any other IP network. The voice data flows over a general-purpose packet-switched network, instead of the traditional dedicated, circuit-switched voice transmission lines.
This arrangement has several advantages over traditional telephony:
- Wider range of features and facilities. In addition to the basic end-to-end voice conversation, more information and control about each call can easily be provided. This includes sending and receiving messages or data files in parallel with the voice conversation, audio conferencing, managing address books and passing presence information about whether friends/colleague are available online to interested parties.
- Freer innovation. Innovation progresses at market rates rather than the slow pace of the multilateral International Telecommunications Union (ITU) committee process, resulting in more new advanced features.
- Lower per-call costs. Once the capital costs of terminals and/or computers and the operating costs of a data network connection are in place, there are no additioanl charges for usage unless the destination is outside the IP network. However, this must be offset by the higher costs of telephony equipment. Prices from VoIP providers are not always cheaper than other carriers.
- Higher quality voice is an option where higher bandwidth voice encoding can be selected to improve end-to-end quality. However, often high compression techniques are used to save bandwidth and potential result in slightly poorer quality than traditional telephony circuits.
- Lower infrastructure costs. VoIP reduces the traditional scheme—two separate wiring systems, one for voice and one for network—to a single connection.
- "Future proof" hardware. Functionality is software (protocol) based, allowing for changes in software coding without requiring a "forklift" or component upgrade.
However, there are several drawbacks which are being addressed as the technology matures:
- Occasional drop-out of voice, where IP packets are lost or dropped in the network between end users. This is more noticeable in highly congested networks and/or where there is long distances and/or interworking between end points. Avoidance of this problem will require introduction of priority schemes for voice traffic, using Quality of Service mechanisms. These have been developed for IP Version 6, but rarely implemented.
- Single point of calling. Typically a computer is used as the terminal for VoIP calls, although telephone adaptors and/or VoIP telephones are commercially available. Unlike a standard POTS (Plain Old Telephone System) phone it is not possible to share a single line with 3 or 4 telephones. In many homes, these ring in parallel, and any may be used to answer and complete the call. This is typically not possible with VoIP where individual terminals are called, although new schemes with VoIP compatible cordless phones and routers with VoIP capability have been introduced.
- Lack of global telephone number range. Whilst the standard 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.
- Emergency Calls are usually barred from VoIP phones because their location cannot be determined. There have been examples of deaths due to home telephone services that had no emergency call service available.
- Need for Electrical Power. VoIP telephones (or computers) are typically powered by mains electricity and thus would not be useable during any power outage. Standard POTS lines are powered remotely from the local telephone exchange which usually has both battery and generator backup to retain service in these conditions.
Protocols used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols.
Voice over IP traffic may be deployed on any IP network, including ones lacking an internet connection, for instance on a building-wide LAN without an internet connection.
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Corporate and telco use of VoIP
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. IP telephony is commonly used to route traffic starting and ending at conventional PSTN (Public Switched Telephone Network) telephones. VoIP is widely employed by carriers, especially for international telephone calls. 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. Companies can acquire their own gateways to eliminate third-party costs, which is worthwhile in some situations.
Implementation challenges
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 (engineering) and data integrity.
A central challenge for VoIP implementers is restructuring streams of received IP packets, which can come in any order and have packets missing, to ensure that the ensuing audio stream maintains a proper time consistency. To help with this, the network provider can ensure that there is enough end-to-end bandwidth to guarantee low-latency, high quality voice. This is trivial in private networks, but very difficult with less than 256 kbit/s bandwidth without a fragmentation mechanism.
Another main challenge is routing VoIP traffic to traverse 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 user's computers as super node servers to route other people's calls.
Keeping packet latency acceptable on satellite circuits can also be a problem, simply due to transmission distances.
Drawbacks
VOIP technology still has several shortcomings which currently prevent its widespread use.
Emergency calls
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. Moreover, in the event that the caller is unable to give an address, emergency services may be unable to respond. Following the lead of mobile phone carriers, several VOIP carriers are already implementing a technical work-around. The United States government has set a deadline, requiring VOIP carriers to implement e911.
This is a different situation with IPBX systems, where these corporate systems often have full e911 capabilites built into the system. e911 capabilies do very by vendor.
Robustness and redundancy
Electrical power is notoriously unreliable. Most people can remember a power outage lasting for longer than 24 hours. Traditional telephones are powered by the phone lines, which are powered by back-up generators in the event of a power outage. Household VOIP hardware uses broadband modems and other equipment powered by household electricity. In order to use VOIP during a power outage, an expensive Uninterruptible power supply or generator must be installed on the premises.
Additionally, broadband connections often have less than desirable reliability. Traditional phone service will likely remain a necessary redundancy until IP technology can match the track record of electrically-switched phones. Providing this level of quality control for a technology as sophisticated as IP might make Internet connections too expensive for the typical consumer.
Mobile phones
Telcos and consumers have invested billions of dollars in mobile phone equipment. In developed countries, mobile phones have achieved nearly complete market penetration. Many people are giving up landlines and using mobiles exclusively. It is questionable whether there would be significant demand for VOIP among consumers, at least until public wireless IP networking has similar geographical coverage to cellular networks (and thereby enabling mobile VoIP phones).
VoIP protocols
In the overwhelming majority of implementations, RTP is used to transmit VoIP traffic ("media"). The notable exception is Inter-Asterisk eXchange protocol IAX which carries both signaling and voice data over a UDP stream, which eases firewall and NAT traversal.
Signaling protocols:
- Session Initiation Protocol (SIP)
- an IETF newcomer gaining popularity
- H.323
- the ITU's widely deployed and continually updated VoIP protocol carrying billions of minutes of traffic each month
- Skinny Client Control Protocol
- proprietary protocol from Cisco
- Megaco (a.k.a. H.248) and MGCP
- both media gateway control protocols
- MiNET
- proprietary protocol from Mitel
- IAX
- the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server and associated client software
Several different speech codecs can be used for stream audio compression. Commonly used codecs for VoIP traffic include G.711 and G.729, both ITU-T-specified codecs.
Mass-market telephony over broadband Internet access
A new development 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. Companies in the US, such as 1TouchTone, Broadvoice, Comcast, Verizon, Vonage, VoicePulse, Voipex, Packet8, Lingo and SunRocket, use IP to offer unlimited calling to the US, 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 US 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 US emergency services number 911, may not automatically be routed to the nearest local emergency dispatch center, or be of any use for subscribers outside the US.
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 a hitch, but in other cases they won't 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.
There is also a free service called Free World Dialup (FWD), that permits users to make free telephone calls to other FWD users, although has only limited connections to and from the public switched telephone network.
See also
Networks
- Babble
- A UK-based VoIP network.
- BroadVoice
- A US-based VoIP network that supplies VoIP adapters, or allows customers to use their own SIP devices.
- DOW Networks
- VoIP Network Provider, Call Center Solutions, IP PBX, connecting toll free numbers to VoIP UIFN and ITFS and Hosted Predictive Dialer ASP.
- Free IP Call
- The Home to Free IP Call, SIP and VoIP Networks Provider.
- Free World Dialup
- A free SIP-based VoIP network.
- Gateshare
- A US Based VoIP Provider with interconnections with FWD
- MyCyberphone
- A US-based VoIP phone service
- MyWebCalls
- A UK-based VoIP phone service using SIP and also supporting the Asterisk PBX
- Sipgate
- A German based VoIP phone service provider with connections to all UK telephone exchanges, and interworking with all leading VoIP providers.
- SIPphone
- A free SIP-based VoIP network.
- Skype
- A proprietary freeware VoIP system which uses a messenger-like client.
- Teleo
- A VoIP network using a P2P model
- TelSIP
- A European-based VoIP network providing the only SIP solution that traverses firewalls and proxies.
- Voipex
- : A US-based VoIP phone service
Software
- Asterisk PBX
- The popular Linux-based open source software PBX switch.
- GameComm Roger Wilco, Teamspeak, & Ventrilo
- Voice communication programs popular in online gaming
- GnomeMeeting
- The popular Linux-based open source softphone, supports H.323 and soon SIP.
- IP Multimedia Subsystem
- architectural model (with several SIP extensions), used by the traditional telecommunications industry to develop systems to replace the current circuit switched network with a NGN network.
- LignUp Corporation
- A powerful, web services based VoIP platform that makes telephony development as easy as web development.
- PhoneGaim
- A free VoIP system based on GAIM and SIP.
- ReSIProcate
- A robust and feature rich open source SIP stack.
- SIMPLE
- An instant messaging protocol based on SIP.
- TelTel
- A Instant Voice software which is combined IM-like features and telephony functions.
- Tivi
- A SIP VoIP client softphone.
- YATE
- A free software VoIP telephony engine (VoIP server and client for H.323,IAX,SIP)
Other
- TestYourVoIP
- A free VoIP quality test website that just requires a Java enabled web browser.
Related concepts
External links
- Info on VoIP (http://www.fcc.gov/voip/) by FCC | Consumer & Governmental Affairs Bureau
- Internet Telephony Magazine (http://www.tmcnet.com/voip/) online resource for learning about VoIP
- VoIP Tutorial (http://www.tutorial-reports.com/internet/telephony/voip/) Includes information on VoIP Technology, Software and Hardware Requirements, Setting up a VoIP and Using PSTN lines.
- Cisco Systems CallManager (http://www.cisco.com/en/US/products/sw/voicesw/ps556/index.html)da:VoIP
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