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58-480: RTAudio was designed for real-time two-way Voice over IP ( VoIP ) applications. Some of the target applications include games, audio conferencing, and wireless applications over IP. RTAudio is the preferred Microsoft Real-Time audio codec, and is the default voice codec for Microsoft ’s Unified Communications platforms. The RTAudio encoder is capable of encoding single-channel (mono), 16 bit per sample audio signals. The encoder can be configured to operate either in

116-838: A cordless phone . Traditional PSTN phones can be used as VoIP phones with analog telephone adapters (ATA). A VoIP phone or application may have many features an analog phone doesn't support, such as e-mail-like IDs for contacts that may be easier to remember than names or phone numbers, or easy sharing of contact lists among multiple accounts. Generally the features of VoIP phones follow those of Skype and other PC-based phone services, which have richer feature sets but may experience latency-related problems, because they rely on mainstream operating systems' IP and audio support. As mainstream operating systems became better at voice applications with appropriate quality of service (QoS) guarantees, and 5G handoff ( IEEE 802.21 etc.) becomes available from wireless carriers, tablets and smartphones became

174-514: A 3G handset or USB wireless broadband adapter, the IP address has no relationship with any physical location known to the telephony service provider, since a mobile user could be anywhere in a region with network coverage, even roaming via another cellular company. At the VoIP level, a phone or gateway may identify itself by its account credentials with a Session Initiation Protocol (SIP) registrar. In such cases,

232-646: A VoIP signalling protocol stack, such as for the Session Initiation Protocol (SIP), H.323 , Skinny Client Control Protocol (Cisco), and/or Skype , is needed. For media streams, the Real-time Transport Protocol (RTP) is used in most VoIP systems. For voice and media encoding, a variety of codecs are available, such as for audio: G.711 , GSM , iLBC , Speex , G.729 , G.722 , G.722.2 (AMR-WB), other audio codecs , and for video H.263 , H.263+ , H.264 . User interface software controls

290-402: A computer or mobile device), will connect to the VoIP service remotely. These connections typically take place over public internet links, such as local fixed WAN breakout or mobile carrier service. In the case of a private VoIP system, the primary telephony system itself is located within the private infrastructure of the end-user organization. Usually, the system will be deployed on-premises at

348-487: A few and must be used in concert. These functions include: VoIP protocols include: Mass-market VoIP services use existing broadband Internet access , by which subscribers place and receive telephone calls in much the same manner as they would via the PSTN. Full-service VoIP phone companies provide inbound and outbound service with direct inbound dialing . Many offer unlimited domestic calling and sometimes international calls for

406-555: A first-come, first-served basis. Fixed delays cannot be controlled as they are caused by the physical distance the packets travel. They are especially problematic when satellite circuits are involved because of the long distance to a geostationary satellite and back; delays of 400–600 ms are typical. Latency can be minimized by marking voice packets as being delay-sensitive with QoS methods such as DiffServ . Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Excessive load on

464-482: A flat monthly subscription fee. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available. A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways: It is increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks as a backhaul to connect switching centers and to interconnect with other telephony network providers; this

522-453: A framework for consolidation of all modern communications technologies using a single unified communications system. Voice over IP has been implemented with proprietary protocols and protocols based on open standards in applications such as VoIP phones, mobile applications, and web-based communications . A variety of functions are needed to implement VoIP communication. Some protocols perform multiple functions, while others perform only

580-470: A given network path due to competition from other users for the same transmission links. VoIP receivers accommodate this variation by storing incoming packets briefly in a playout buffer , deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it. The added delay is thus a compromise between excessive latency and excessive dropout , i.e. momentary audio interruptions. Although jitter

638-448: A link can cause congestion and associated queueing delays and packet loss . This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion. But VoIP usually uses UDP not TCP because recovering from congestion through retransmission usually entails too much latency. So QoS mechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting them ahead of any queued bulk traffic on

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696-530: A media gateway (aka IP Business Gateway) and connects the digital media stream, so as to complete the path for voice and data. Gateways include interfaces for connecting to standard PSTN networks. Ethernet interfaces are also included in the modern systems which are specially designed to link calls that are passed via VoIP. E.164 is a global numbering standard for both the PSTN and public land mobile network (PLMN). Most VoIP implementations support E.164 to allow calls to be routed to and from VoIP subscribers and

754-439: A service provider or telecommunications carrier hosting the telephone system as a software solution within their own infrastructure. Typically this will be one or more data centers with geographic relevance to the end-user(s) of the system. This infrastructure is external to the user of the system and is deployed and maintained by the service provider. Endpoints, such as VoIP telephones or softphone applications (apps running on

812-567: A site within the direct control of the organization. This can provide numerous benefits in terms of QoS control (see below ), cost scalability, and ensuring privacy and security of communications traffic. However, the responsibility for ensuring that the VoIP system remains performant and resilient is predominantly vested in the end-user organization. This is not the case with a Hosted VoIP solution. Private VoIP systems can be physical hardware PBX appliances, converged with other infrastructure, or they can be deployed as software applications. Generally,

870-413: A standard RJ11 interface that can accommodate a standard analog telephone. Another type of gateway device acts as a simple cellular base station. Regular mobile phones can connect to this device, and make VoIP calls. While a license is required to run a cellular base station in most countries, these can be useful on ships , or in remote areas where a low-powered gateway transmitting on unused frequencies

928-441: A subscriber to select a new telephone carrier without requiring a new number to be issued. Typically, it is the responsibility of the former carrier to "map" the old number to the undisclosed number assigned by the new carrier. This is achieved by maintaining a database of numbers. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier. Multiple porting references must be maintained even if

986-448: A variety of other applications. DSL modems typically provide Ethernet connections to local equipment, but inside they may actually be Asynchronous Transfer Mode (ATM) modems. They use ATM Adaptation Layer 5 (AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission, reassembling them back into Ethernet frames at the receiving end. Using a separate virtual circuit identifier (VCI) for voice over IP has

1044-519: Is a method and group of technologies for voice calls for the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet . The broader terms Internet telephony , broadband telephony , and broadband phone service specifically refer to the provisioning of voice and other communications services ( fax , SMS , voice messaging ) over the Internet, rather than via

1102-470: Is a random variable, it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Motivated by the central limit theorem , jitter can be modeled as a Gaussian random variable . This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above

1160-399: Is characterized by several metrics that may be monitored by network elements and by the user agent hardware or software. Such metrics include network packet loss , packet jitter , packet latency (delay), post-dial delay, and echo. The metrics are determined by VoIP performance testing and monitoring. A VoIP media gateway controller (aka Class 5 Softswitch) works in cooperation with

1218-708: Is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market. Skype , which originally marketed itself as a service among friends, has begun to cater to businesses, providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge. In general, the provision of VoIP telephony systems to organizational or individual users can be divided into two primary delivery methods: private or on-premises solutions, or externally hosted solutions delivered by third-party providers. On-premises delivery methods are more akin to

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1276-595: Is generally uncommon for those private connectivity methods to be provided by Hosted or Cloud VoIP providers. Communication on the IP network is perceived as less reliable in contrast to the circuit-switched public telephone network because it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental quality of service (QoS) guarantees. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to data loss in

1334-414: Is just one of them. STUN or any other NAT traversal mechanism is not required when the two SIP phones connecting are routable from each other and no firewall exists in between. DHCP client software simplifies connection of a device to an IP network. The software automatically configures the network and VoIP service parameters. The overall hardware may look like a telephone or mobile phone. A VoIP phone has

1392-491: Is no longer necessary to carry both a desktop phone and a cell phone. Maintenance becomes simpler as there are fewer devices to oversee. VoIP solutions aimed at businesses have evolved into unified communications services that treat all communications—phone calls, faxes, voice mail, e-mail, web conferences, and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of service providers are operating in this space: one set

1450-471: Is often referred to as IP backhaul . Smartphones may have SIP clients built into the firmware or available as an application download. Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new Private branch exchange (PBX) lines installed internationally were VoIP. For example, in

1508-519: Is packetized and transmission occurs as IP packets over a packet-switched network . They transport media streams using special media delivery protocols that encode audio and video with audio codecs and video codecs . Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on narrowband and compressed speech , while others support high-fidelity stereo codecs. The most widely used speech coding standards in VoIP are based on

1566-444: Is received by a center the location is automatically determined from its databases and displayed on the operator console. In IP telephony, no such direct link between location and communications end point exists. Even a provider having wired infrastructure, such as a DSL provider, may know only the approximate location of the device, based on the IP address allocated to the network router and

1624-512: Is required in this case to enable routing of SIP packets from one network to other. STUN is used in some of the sip phones to enable the SIP/RTP packets to cross boundaries of two different IP networks. A packet becomes unroutable between two sip elements if one of the networks uses private IP address range and other is in public IP address range. Stun is a mechanism to enable this border traversal. There are alternate mechanisms for traversal of NAT, STUN

1682-412: Is to reduce the maximum transmission time by reducing the maximum transmission unit . But since every packet must contain protocol headers, this increases relative header overhead on every link traversed. The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Packet delay variation results from changes in queuing delay along

1740-458: The E.164 number to URI mapping (ENUM) service in IMS and SIP. Echo can also be an issue for PSTN integration. Common causes of echo include impedance mismatches in analog circuitry and an acoustic path from the receive to transmit signal at the receiving end. Local number portability (LNP) and mobile number portability (MNP) also impact VoIP business. Number portability is a service that allows

1798-460: The Internet . This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN). Digital IP-based telephone service uses control protocols such as the Session Initiation Protocol (SIP), Skinny Client Control Protocol (SCCP) or various other proprietary protocols. VoIP phones can be simple software-based softphones or purpose-built hardware devices that appear much like an ordinary telephone or

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1856-510: The Internet telephony service provider (ITSP) knows only that a particular user's equipment is active. Service providers often provide emergency response services by agreement with the user who registers a physical location and agrees that, if an emergency number is called from the IP device, emergency services are provided to that address only. VoIP phone A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as

1914-861: The linear predictive coding (LPC) and modified discrete cosine transform (MDCT) compression methods. Popular codecs include the MDCT-based AAC-LD (used in FaceTime ), the LPC/MDCT-based Opus (used in WhatsApp ), the LPC-based SILK (used in Skype ), μ-law and A-law versions of G.711 , G.722 , and an open source voice codec known as iLBC , a codec that uses only 8 kbit/s each way called G.729 . Early providers of voice-over-IP services used business models and offered technical solutions that mirrored

1972-400: The public switched telephone network (PSTN), also known as plain old telephone service (POTS). The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network , the digital information

2030-640: The Narrow Band mode (8 kHz sampling rate) or the Wide Band mode (16 kHz sampling rate). The RTAudio decoder has a built-in jitter control module as well as an error concealment module. RTAudio is a proprietary codec. Like RTVideo , this protocol can also be licensed from Microsoft. This article about software created, produced or developed by Microsoft is a stub . You can help Misplaced Pages by expanding it . Voice over IP Voice over Internet Protocol ( VoIP ), also called IP telephony ,

2088-543: The PSTN/PLMN. VoIP implementations can also allow other identification techniques to be used. For example, Skype allows subscribers to choose Skype names (usernames) whereas SIP implementations can use Uniform Resource Identifier (URIs) similar to email addresses . Often VoIP implementations employ methods of translating non-E.164 identifiers to E.164 numbers and vice versa, such as the Skype-In service provided by Skype and

2146-967: The United States, the Social Security Administration is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network. VoIP allows both voice and data communications to be run over a single network, which can significantly reduce infrastructure costs. The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as personal computers . Rather than closed architectures, these devices rely on standard interfaces. VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi network, so that it

2204-426: The architecture of the legacy telephone network. Second-generation providers, such as Skype , built closed networks for private user bases, offering the benefit of free calls and convenience while potentially charging for access to other communication networks, such as the PSTN. This limited the freedom of users to mix-and-match third-party hardware and software. Third-generation providers, such as Google Talk , adopted

2262-446: The classic PBX deployment model for connecting an office to local PSTN networks. While many use cases still remain for private or on-premises VoIP systems, the wider market has been gradually shifting toward Cloud or Hosted VoIP solutions. Hosted systems are also generally better suited to smaller or personal use VoIP deployments, where a private system may not be viable for these scenarios. Hosted or Cloud VoIP solutions involve

2320-399: The concept of federated VoIP . These solutions typically allow dynamic interconnection between users in any two domains of the Internet, when a user wishes to place a call. In addition to VoIP phones , VoIP is also available on many personal computers and other Internet access devices. Calls and SMS text messages may be sent via Wi-Fi or the carrier's mobile data network. VoIP provides

2378-531: The dominant interfaces. iPhone , Android and the QNX OS used in 2012-and-later BlackBerry phones are widely capable of VoIP performance. Besides wireless, they also typically support USB, but not Ethernet or Power over Ethernet interfaces. The smartphone became the dominant VoIP phone because it works both indoors and outdoors, and shifts base stations/protocols easily. It achieves this by accepting higher access costs and call clarity, and other factors personal to

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2436-436: The enterprise markets because of LCR options, VoIP needs to provide a certain level of reliability when handling calls. A telephone connected to a land line has a direct relationship between a telephone number and a physical location, which is maintained by the telephone company and available to emergency responders via the national emergency response service centers in form of emergency subscriber lists. When an emergency call

2494-438: The following hardware components There are several Wi-Fi enabled mobile phones and PDAs that have pre-installed SIP client software, or are capable of running IP telephony clients, including most smartphones . Analog telephone adapters provide an interface for traditional analog telephones to a voice-over-IP network. They connect to the Internet or local area network using an Ethernet port and have jacks that provide

2552-404: The jitter buffer. VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP summary report or one of the other signaling protocol extensions. VoIP metrics reports are intended to support real-time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and

2610-508: The known service address. Some ISPs do not track the automatic assignment of IP addresses to customer equipment. IP communication provides for device mobility. For example, a residential broadband connection may be used as a link to a virtual private network of a corporate entity, in which case the IP address being used for customer communications may belong to the enterprise, not the residential ISP. Such off-premises extensions may appear as part of an upstream IP PBX. On mobile devices, e.g.,

2668-525: The latter two options will be in the form of a separate virtualized appliance. However, in some scenarios, these systems are deployed on bare metal infrastructure or IoT devices. With some solutions, such as 3CX, companies can attempt to blend the benefits of hosted and private on-premises systems by implementing their own private solution but within an external environment. Examples can include data center collocation services, public cloud, or private cloud locations. For on-premises systems, local endpoints within

2726-498: The mean will arrive too late to be useful. In practice, the variance in latency of many Internet paths is dominated by a small number (often one) of relatively slow and congested bottleneck links . Most Internet backbone links are now so fast (e.g. 10 Gbit/s) that their delays are dominated by the transmission medium (e.g. optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant. A number of protocols have been defined to support

2784-475: The network root prefix to determine how to route a call. Instead, they must now determine the actual network of every number before routing the call. Therefore, VoIP solutions also need to handle MNP when routing a voice call. In countries without a central database, like the UK, it may be necessary to query the mobile network about which home network a mobile phone number belongs to. As the popularity of VoIP increases in

2842-405: The number is routed to a mobile phone number on a traditional mobile carrier. LCR is based on checking the destination of each telephone call as it is made, and then sending the call via the network that will cost the customer the least. This rating is subject to some debate given the complexity of call routing created by number portability. With MNP in place, LCR providers can no longer rely on using

2900-478: The operation of the hardware components, and may respond to user actions with messages to a display screen. To enable the VoIP communications, the SIP/RTP packets should be utilised and STUN client would be the key component for VoIP communications with management of the SIP/RTP packets. A Session Traversal Utilities for NAT (STUN) client is used on some SIP -based VoIP phones as firewalls on network interface sometimes block SIP/RTP packets. Some special mechanism

2958-748: The potential to reduce latency on shared connections. ATM's potential for latency reduction is greatest on slow links because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM VCs. The latest generations of DSL, VDSL and VDSL2 , carry Ethernet without intermediate ATM/AAL5 layers, and they generally support IEEE 802.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic. ATM has substantial header overhead: 5/53 = 9.4%, roughly twice

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3016-463: The presence of congestion than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically. Therefore, VoIP implementations may face problems with latency , packet loss, and jitter . By default, network routers handle traffic on

3074-689: The reporting of quality of service (QoS) and quality of experience (QoE) for VoIP calls. These include RTP Control Protocol (RTCP) extended reports, SIP RTCP summary reports, H.460.9 Annex B (for H.323 ), H.248 .30 and MGCP extensions. The RTCP extended report VoIP metrics block specified by RFC   3611 is generated by an VoIP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal/noise/echo level, mean opinion scores (MOS) and R factors and configuration information related to

3132-497: The same link, even when the link is congested by bulk traffic. VoIP endpoints usually have to wait for the completion of transmission of previous packets before new data may be sent. Although it is possible to preempt (abort) a less important packet in mid-transmission, this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets. An alternative to preemption on slower links, such as dialup and digital subscriber line (DSL),

3190-530: The same location typically connect directly over the LAN . For remote and external endpoints, available connectivity options mirror those of Hosted or Cloud VoIP solutions. However, VoIP traffic to and from the on-premises systems can often also be sent over secure private links. Examples include personal VPN, site-to-site VPN , private networks such as MPLS and SD-WAN, or via private SBCs (Session Border Controllers). While exceptions and private peering options do exist, it

3248-522: The subscriber returns to the original carrier. The Federal Communications Commission (FCC) mandates carrier compliance with these consumer-protection stipulations. In November 2007, the FCC in the United States released an order extending number portability obligations to interconnected VoIP providers and carriers that support VoIP providers. A voice call originating in the VoIP environment also faces least-cost routing (LCR) challenges to reach its destination if

3306-443: The total header overhead of a 1500 byte Ethernet frame. This "ATM tax" is incurred by every DSL user whether or not they take advantage of multiple virtual circuits – and few can. Several protocols are used in the data link layer and physical layer for quality-of-service mechanisms that help VoIP applications work well even in the presence of network congestion . Some examples include: The quality of voice transmission

3364-474: The user. The PoE/USB VoIP phone was thus relegated to the role of a transitional device, except in traditional business office, where it is still widely used as a desk phone. A VoIP telephone consist of the hardware and software components. The software requires standard networking components such as a TCP/IP network stack, client implementation for DHCP , and the Domain Name System (DNS). In addition,

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