Call completion is a telephony feature allowing some form of alternative interaction between parties who cannot converse directly with each other. There are several possible factors which can prevent a telephone call from connecting successfully:
There are various definitions as to what exactly constitutes call completion. Generally speaking, call completion may encompass the following services:
The rationale for providing these features is that allowing some sort of communication between parties unable to talk directly serves both the interests of the callers (allowing them to exchange information despite the inability to talk) as well as the telecommunication operators, as the service allows them to get some return for providing the resources for the call (whereas if the caller simply hangs up, the operators basically have provided the call resources with no return).
Call completion features can be implemented on a private branch exchange (PBX) (e.g. call hold or call transfer), on a dedicated server (e.g. a voicemail server) or directly on the client device (e.g. messaging applications).
There are several commercial companies which provide call completion features, as well as IETF documents specifying call completion features for open standards, such as Session Initiation Protocol (SIP).
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
Extensible Messaging and Presence Protocol is an open communication protocol designed for instant messaging (IM), presence information, and contact list maintenance. Based on XML, it enables the near-real-time exchange of structured data between two or more network entities. Designed to be extensible, the protocol offers a multitude of applications beyond traditional IM in the broader realm of message-oriented middleware, including signalling for VoIP, video, file transfer, gaming and other uses.
SIMPLE, the Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions, is an instant messaging (IM) and presence protocol suite based on Session Initiation Protocol (SIP) managed by the Internet Engineering Task Force.
In computer and telecommunications networks, presence information is a status indicator that conveys ability and willingness of a potential communication partner—for example a user—to communicate. A user's client provides presence information via a network connection to a presence service, which is stored in what constitutes his personal availability record and can be made available for distribution to other users to convey their availability for communication. Presence information has wide application in many communication services and is one of the innovations driving the popularity of instant messaging or recent implementations of voice over IP clients.
VoIP spam or SPIT is unsolicited, automatically dialed telephone calls, typically using voice over Internet Protocol (VoIP) technology.
A business telephone system is a telephone system typically used in business environments, encompassing the range of technology from the key telephone system (KTS) to the private branch exchange (PBX).
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is a standardised architectural framework for delivering IP multimedia services. Historically, mobile phones have provided voice call services over a circuit-switched-style network, rather than strictly over an IP packet-switched network. Various voice over IP technologies are available on smartphones; IMS provides a standard protocol across vendors.
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.
Gizmo5 was a voice over IP communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5. On March 4, 2011, Google announced that the service would be discontinued as of April 3, 2011.
The SIP URI scheme is a Uniform Resource Identifier (URI) scheme for the Session Initiation Protocol (SIP) multimedia communications protocol. A SIP address is a URI that addresses a specific telephone extension on a voice over IP system. Such a number could be a private branch exchange or an E.164 telephone number dialled through a specific gateway. The scheme was defined in RFC 3261.
Text over IP is a means of providing a real-time text (RTT) service that operates over IP-based networks. It complements Voice over IP (VoIP) and Video over IP.
The Media Server Control Markup Language (MSCML) is a protocol used in conjunction with the Session Initiation Protocol (SIP) to enable the delivery of advanced multimedia conferencing services over IP networks. The MSCML specification has been published by the IETF under RFC 4722, now obsoleted by the newer RFC 5022. MSCML was pioneered by the media server company Snowshore, now part of the Dialogic Corporation. MSCML built on ideas from the Netann protocol, and in turn inspired the MSML. An IETF working group called MediaCTRL have now embarked on a standardization of media server scripting languages, drawing on these earlier efforts. Voice scripting protocols like VoiceXML and CCXML are also inspiring sources, and in some cases need to be integrated with what media servers will need to operate in the real world.
Video Share is an IP Multimedia System (IMS) enabled service for mobile networks that allows users engaged in a circuit switch voice call to add a unidirectional video streaming session over the packet network during the voice call. Any of the parties on the voice call can initiate a video streaming session. There can be multiple video streaming sessions during a voice call, and each of these streaming sessions can be initiated by any of the parties on the voice call. The video source can either be the camera on the phone or a pre-recorded video clip.
Peer-to-peer SIP (P2P-SIP) is an implementation of a distributed voice over Internet Protocol (VoIP) or instant messaging communications application using a peer-to-peer (P2P) architecture in which session control between communication end points is facilitated with the Session Initiation Protocol (SIP).
In computer networking, the Message Session Relay Protocol (MSRP) is a protocol for transmitting a series of related instant messages in the context of a communications session. An application instantiates the session with the Session Description Protocol (SDP) over Session Initiation Protocol (SIP) or other rendezvous methods.
Acrobits is a privately owned software development company creating VoIP Clients for mobile platforms, based in Prague, Czech Republic.
Network address translators (NAT) are used to overcome the lack of IPv4 address availability by hiding an enterprise or even an operator's network behind one or few IP addresses. The devices behind the NAT use private IP addresses that are not routable in the public Internet. The Session Initiation Protocol (SIP) has established itself as the de facto standard for voice over IP (VoIP) communication. In order to establish a call, a caller sends a SIP message, which contains its own IP address. The callee is supposed to reply back with a SIP message destined to the IP addresses included in the received SIP message. This will obviously not work if the caller is behind a NAT and is using a private IP address.
The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) to create and control multimedia sessions with multiple participants in the IP Multimedia Subsystem (IMS). It is therefore a key element in the IMS framework.
STIR/SHAKEN, or SHAKEN/STIR, is a suite of protocols and procedures intended to combat caller ID spoofing on public telephone networks. Caller ID spoofing is used by robocallers to mask their identity or to make it appear the call is from a legitimate source, often a nearby phone number with the same area code and exchange, or from well-known agencies like the Internal Revenue Service or Ontario Provincial Police. This sort of spoofing is common for calls originating from voice-over-IP (VoIP) systems, which can be located anywhere in the world.