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Voice over IP (also called VoIP, IP Telephony, and Internet telephony) is the 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.
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 private building-wide LAN.
Technical details
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 severe problems dealing with latency, especially if satellite circuits are involved. They are faced with the problem of 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 on consumer internet links with less than 256 kbit/s bandwidth.
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 users' computers as super node servers to route other people's calls.
Keeping packet latency acceptable can also be a problem, simply due to transmission distances.
Protocols
After a long and at times heated debate about what protocol is best suited to replace today's PSTN telephony infrastructure, the industry has now settled on the Session Initiation Protocol (SIP), an IETF standard. While Voice over IP (VoIP) systems deployed in the Enterprise as a replacement of aging PBX systems still remain proprietary, carriers and ISPs have started to invest heavily in their next generation infrastructure based on SIP. A notable exception on the Enterprise IP PBX side is sipX - The Open Source SIP PBX for Linux from SIPfoundry, a fully SIP compliant implementation running on Linux and available for free.
Signaling protocols:
- H.323
- defined by the ITU-T
- Session Initiation Protocol (SIP)
- defined by the IETF, newer than H.323, is becoming the dominant protocol
- Megaco (a.k.a. H.248) and MGCP
- both media gateway control protocols
- Skinny Client Control Protocol
- proprietary protocol from Cisco
- MiNET
- proprietary protocol from Mitel
- IAX
- the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server and associated client software
- Skype
- a proprietary peer-to-peer protocol used in the Skype application
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.
Advantages
- Wider range of features and facilities. In addition to the basic end-to-end voice conversation, more information about and control over 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 information about whether others (e.g. friends or colleagues) 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 additional charges for usage unless the destination is outside the IP network. However, this must be offset by the higher costs of telephony equipment. Services from VoIP providers are not always cheaper than from conventional telephone service providers.
- 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 potentially 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.
Drawbacks
VOIP technology still has several shortcomings that lead some to believe that it's not ready for widespread deployment. However, most industry analysts predict that 2005 is the Year of Inflection, which is the year in which more IP PBX ports will ship than legacy digital PBX ports.
Reliability
Traditional telephones are powered by the phone lines, which in the event of a power failure will be kept live by back-up generators or batteries located at the 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 expensive uninterruptible power supply or generator must be installed on the premises.
Additionally, broadband connections often 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. 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.
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.
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, 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 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 capabilities built into the system. e911 capabilities do vary by vendor.
Integration into global telephone number system
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.
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 three or four 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. Today with commercial services such as Vonage and AT&T CallVantage, 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.
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, and many people are giving up landlines and using mobiles exclusively. In this situation, 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 (thereby enabling mobile VoIP phones).
Adoption
Mass-market telephony
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. Companies in the U.S., such as 1TouchTone, Broadvoice, Comcast, Verizon, Vonage, VoicePulse, Voipex, Packet8, Lingo, and SunRocket, use IP to 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, or be of any 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.
Corporate and telco use
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 (Public Switched Telephone Network) 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.
See also
Networks
- Aevoe
- A company offering a complete serverless hardware VoIP solution without servers or subscription charges for residential or SOHO use. Aevoe products employ SNAP technology so users can keep their regular phone numbers for VoIP calling.
- Babble
- A UK-based VoIP network.
- BroadVoice
- A U.S.-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 U.S.-based VoIP Provider with interconnections with FWD
- MetroTel
- A U.S.-based VoIP Provider with 800 and local numbers
- MyCyberphone
- A U.S.-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
- TelTel
- The largest SIP community
- TelSIP
- A European-based VoIP network providing the only SIP solution that traverses firewalls and proxies.
- ViaTalk
- A U.S.-based VoIP phone service
- Voipex
- : A U.S.-based VoIP phone service
- Vonage
- A U.S.-based VoIP phone service provider
- Jajah
- A freeware VoIP client with free videotelephony, chat, text messaging, voicemailbox and is compatible to SIP, Skype, Gizmo and IAX/H.323
Software
- Asterisk PBX
- The popular Linux-based open source software PBX switch.
- GameComm Roger Wilco, Teamspeak, and 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.
- 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.
- sipX - The Open Source SIP PBX for Linux
- The popular open source SIP PBX, native SIP call control, many features, Web management, and fully standards-compliant
- 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
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