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Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, Diffserv).
The principal cause of packet loss is congestion, which can be controlled by congestion management and avoidance. Carrier VoIP networks avoid congestion by means of traffic engineering.
Variation in delay is called Jitter. The effects of jitter can be mitigated by storing voice packets in a buffer (called a play-out buffer) upon arrival, before playing them out. This avoids a condition known as buffer underrun, in which the playout process runs out of voice data to play because the next voice packet has not yet arrived, but increases delay by the length of the buffer.Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of the transmit and receive signal at the receiving end.
VoIP technology does not necessarily require broadband Internet access, but this usually supports better quality of service. A sizable percentage of homes today are connected to the Internet through DSL, which requires a traditional phone line. Having to pay for VoIP in addition to both a basic phone line and broadband Internet access reduces the potential benefits of VoIP.
Most telephone companies now offer DSL service without the phone (often called "naked DSL" or "dry loop DSL"), thus saving subscribers money when they switch to VoIP. VoIP can also be used with Cable Internet instead of DSL, potentially eliminating the need for a traditional phone line entirely.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 dictating the use of an uninterruptible power supply or generator to ensure availability during power outages.
Early adopters of VoIP may also be users of other phone equipment, such as PBX and cordless phone bases, that rely on power not provided by the telephone company. Even with local power still available, the broadband carrier itself may experience outages as well. While the PSTN has been matured over decades and is typically extremely reliable, most broadband networks are less than 10 years old, and even the best are still subject to intermittent outages. Furthermore, consumer network technologies such as cable and DSL often are not subject to the same restoration service levels as the PSTN or business technologies such as T-1 connection.
Some broadband connections may have less than desirable quality. 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. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on.
While 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 provider specific short codes.
With hardware VoIP solution 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. Software based VoIP services require the use of a computer, so they are limited to single point of calling, though handsets are now available, allowing them to be used without a PC. Some services 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.
The majority of consumer VoIP solutions do not support encryption yet. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. There are several open source solutions that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented audio codecs that are not easily available for open source applications, however such security through obscurity has not proven effective in the long run in other fields. Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely available at a consumer level. The existing secure standard SRTP is available on consumer devices from some manufacturers like Sipura/Linksys for Analog Telephone Adapters (ATAs) and Gizmo Project for softphones (PCs/laptops emulating a phone).
The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec encryption to the digitized voice stream.
As of April 2006, the beta testing of Zfone, a 'security wrapper' for certain VoIP systems by the inventor of PGP, is notable, as a means by which strong security may be added to certain otherwise less secure VoIP systems. The softphone Skype claims to use strong encryption by default, although it is not clear which encryption standards it uses.
VoIP has become a major provider of phone services to due to not having a fixed or mobile phone or high overseas roaming charges choose instead to use VoIP services to make their phone calls. Pre-Paid phone cards can be used either from a normal phone or from Internet Cafes that have phone services. The undeveloped markets are usually markets where Pre-Paid cards are used; however in cities with high tourist or immigrant communities they are also common.
Caller ID support among VoIP providers varies. When calling a PSTN number from some VoIP providers, Caller ID isn't supported, and the target person will not know who is calling. The number shows up as 'Unknown' or '000-012-3456'.
In a few cases, VoIP providers may allow a caller to spoof the Caller ID information, making it appear that they are calling from a different number.
A major development well under way 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. Full phone service VoIP phone companies provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited calling to the U.S., and some to Canada or selected countries in Europe or Asia as well, for a flat monthly fee.
These services take a wide variety of forms which can be more or less similar to traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet connection and an existing telephone jack in order to provide service nearly indistinguishable from POTS on all the other jacks in the residence. This type of service, which is fixed to one location, is generally offered by broadband Internet providers such as cable companies and telephone companies as a cheaper flat-rate traditional phone service. Often the phrase "VoIP" is not used in selling these services, but instead the industry has marketed the phrase "Internet Phone" or "Digital Phone" which is aimed at typical phone users who are not necessarily tech-savvy. Typically, the provider touts the advantage of being able to keep one's existing phone number. Examples of this type of service in the United States include Time Warner and Comcast's Digital Phone, Verizon VoiceWing, and AT&T CallVantage.
At the other extreme are services like Gizmo Project and Skype which rely on a software client on the computer in order to place a call over the network, where one user ID can be used on many different computers or in different locations on a laptop. In the middle lie services like Vonage or BroadVoice which also provide a telephone adapter for connecting to the broadband connection similar to the services offered by broadband providers (and in some cases also allow direct connections of SIP phones) but which are aimed at a more tech-savvy user and allow portability from location to location. One advantage of these two types of services is the ability to make and receive calls as one would at home, anywhere in the world, at no extra cost. No additional charges are incurred, as call diversion via the PSTN would, and the called party does not have to pay for the call. For example, if a subscriber with a home phone number in a U.S. area code calls someone else in his home area code, it will be treated as a local call regardless of where that person is in the world. Often the user may also select a phone number with any desired area code; this is generally done to minimize the phone tariffs of those who frequently call.
For some users, the broadband phone complements, rather than replaces, a PSTN line, due to a number of inconveniences compared to traditional services. VoIP requires a broadband Internet connection and, if a telephone adapter is used, a power adapter is usually needed. In the case of a power failure, VoIP services will generally not function. Additionally, a call to 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. This is particularly true for users who select a number with an area code outside their area.
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 (IMS) 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.
Click-to-call is a service which lets users click a button and immediately speak with a customer service representative. The call can either be carried over VoIP, or the customer may request an immediate call back by entering their phone number. One significant benefit to click-to-call providers is that it allows companies to monitor when online visitors change from the website to a phone sales channel.
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 legacy PSTN services.
In the U.S., the Federal Communications Commission now requires all VoIP operators who do not 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. The Telecommunications Act of 2005 proposes adding more traditional PSTN regulations, such as local number portability and universal service fees. Other future legal issues are likely to include laws against wiretapping and network neutrality.
Some Latin American countries, fearful for their state owned telephone services, have imposed restrictions on the use of VoIP, including in Panama where VoIP is taxed. In Ethiopia, where a totalitarian government is monopolizing telecommunication service, it is a criminal offence to offer services using VoIP. The country has installed firewalls to prevent international calls being made using VoIP. These measures were taken after a popularity in VoIP reduced the income generated by the state owned telecommunication company.
In the European Union, the treatment of VoIP service providers is a decision for each Member State's national telecoms regulator, which must use competition law theory to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). A general distinction is usually made between VoIP services that function over managed networks (via broadband connections) and VoIP services that function over unmanaged networks (essentially, the Internet).
VoIP services that function over managed networks are often considered to be a viable substitute for PSTN telephone services (despite the problems of power outages and lack of geographical information); as a result, major operators that provide these services (in practice, incumbent operators) may find themselves bound by obligations of price control or accounting separation.
VoIP services that function over unmanaged networks are often considered to be too poor in quality to be a viable substitute for PSTN services; as a result, they may be provided without any specific obligations, even if a service provider has "significant market power".
The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power (e.g. the obligation to offer access to emergency calls), and it is impossible to say definitively whether VoIP service providers of either type are bound by them. A review of the EU Directive is under way and should be complete by 2007.
In India, it is legal to use VoIP. But it is illegal to have VoIP gateways inside India. This effectively means, people who have PCs can use it to make a VoIP call to any number. But if the remote side is a normal phone, the gateway that converts VoIP call to POTS call should not be inside India.
The two major competing protocols for VoIP are SIP and H.323. Initially H.323 was the most popular protocol, though its popularity has decreased in the "local loop" due to its poor traversal of NAT and firewalls. For this reason as domestic VoIP services have been developed, SIP has been far more widely adopted. However in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being terminated over VoIP. So really SIP is a useful tool for the "local loop" and H.323 is like the "fiber backbone". With the most recent changes introduced for H.323, however, it is now possible for H.323 devices to easily and consistently traverse NAT and firewall devices, opening up the possibility that H.323 may again be looked upon more favorably in cases where such devices encumbered its use previously.
Where VoIP travels through multiple providers' Soft Switches the concept of Full Media Proxy and signalling proxy are important. In H.323 the data is made up of 3 streams of data: 1) H.225.0 Call Signalling 2) H.245 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600 ms) and packet loss will be high. However in signalling proxy mode where only the signalling flows through the provider the delay will be reduced to a more user friendly 120-150 ms. These proxy concepts could lead the way to true global providers.
One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically to send a G.723.1 5.6 kbit/s compressed audio path will require 18 kbit/s of bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth optimisation techniques used such as silence suppression and header compression. This can typically save 35% on bandwidth used. But the really interesting technology comes from VoIP off shoots such as TDMoIP which take advantage of the concept of bundling conversations that are heading to the same destination and wrapping them up inside the same packets. These can offer near toll quality audio in a 6-7 kbit/s data stream.
Most standards-based solutions use either the H.323 or Session Initiation Protocol (SIP) protocols. A number of proprietary designs also exist. These typically support features such as call waiting, conference calling, and call transfer.
Transport protocols:
| SRTP | Existing secure transport protocol | |
| RTP | Unsecure transport protocol | |
| ZRTP | New secure transport protocol proposal | |
| Signaling protocols: | ||
| Session Initiation Protocol (SIP) | defined by the IETF, newer than H.323 | |
| H.323 | defined by the ITU-T | |
| 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 | |
| CorNet-IP | proprietary protocol from Siemens | |
| IAX | the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server, Yate and associated client software | |
| Skype | a proprietary peer-to-peer protocol used in the Skype application | |
| Jajah | a proprietary peer-to-peer protocol used in the Jajah SIP and IAX compatible webphone | |
| Jingle | open peer-to-peer protocol based on XMPP (Jabber) and being harmonised with the 'substantially equivalent' Google Talk protocol. | |
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Until August 31
