This article needs additional citations for verification .(December 2006) |
Voice over Internet Protocol (VoIP) recording is a subset of telephone recording or voice logging, first used by call centers and now being used by all types of businesses. There are many reasons for recording voice over IP call traffic such as: reducing company vulnerability to lawsuits by maintaining recorded evidence, complying with telephone call recording laws, increasing security, employee training and performance reviews, enhancing employee control and alignment, verifying data, sharing data as well as customer satisfaction and enhancing call center agent morale. [1]
By definition, Voice over IP is audio converted into digital packets and then converted to IP packets. [2] VoIP recording is accomplished either by sniffing the network or by having the packets duplicated and directed to the recorder—passive recording or active recording, respectively. [3] [4]
VoIP is usually implemented as a cost-saving measure over POTS (Plain old telephone systems). The same holds true now for VoIP recording. Most recording vendors are able to record the various standards of VoIP such as G.711, G.729a/b and G.723 and software-only solutions as compared to the intensive hardware and software associated with legacy PBX recording. [6]
Today, most of the VoIP vendors are offering VoIP recording methods specific to their VoIP call and communications management servers. These vendors are offering what is referred to as active VoIP recording where the recording vendor's solution becomes an "active" participant within the call for recording purposes. This approach offers some benefits over the long established method of sniffing (Passive) recording in environments where the handsets to be recorded are off site or in remote locations, or in situations where the network routing would mean that a passive solution would be overcomplicated. It also greatly simplifies recording internal calls, as it no longer necessary to duplicate the audio streamed between two handsets to the voice recorder as the telephony system will automatically manage this in any solution.[ citation needed ]
Disadvantages of "active" call recordings can include overheads on the PBX, the need for agent interaction and changes to the quality of the call. "Passive" call-recording software works by using packet filter technology to listen for VoIP calls on the LAN on a monitored port. The RTP stream is then captured and converted to a WAV file for storage and retrieval. [7]
VoIP calls can be recorded via streaming audio recording applications. Most call centers and other organizations required to record calls would more often use a recording system offered by the softphone or IP PBX. Streaming audio recorders can be useful for home-based recording. [8] [9]
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).
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls, the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.
A softswitch is a call-switching node in a telecommunications network, based not on the specialized switching hardware of the traditional telephone exchange, but implemented in software running on a general-purpose computing platform. Like its traditional counterparts it connects telephone calls between subscribers or other switching systems across a telecommunication network. Often a softswitch is implemented to switch calls using voice over IP (VoIP) technologies, but hybrid systems exist.
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints, such as customary telephone sets, destinations on the public switched telephone network (PSTN), and devices or services on voice over Internet Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing.
A business telephone system is a multiline telephone system typically used in business environments, encompassing systems ranging in technology from the key telephone system (KTS) to the private branch exchange (PBX).
Class of service is a parameter used in data and voice protocols to differentiate the types of payloads contained in the packet being transmitted. The objective of such differentiation is generally associated with assigning priorities to the data payload or access levels to the telephone call.
Direct inward dialing (DID), also called direct dial-in (DDI) in Europe and Oceania, is a telecommunication service offered by telephone companies to subscribers who operate a private branch exchange (PBX) system. The feature provides service for multiple telephone numbers over one or more analog or digital physical circuits to the PBX, and transmits the dialed telephone number to the PBX so that a PBX extension is directly accessible for an outside caller, possibly by-passing an auto-attendant.
A network tap is a system that monitors events on a local network. A tap is typically a dedicated hardware device, which provides a way to access the data flowing across a computer network.
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).
A VoIP VPN combines voice over IP and virtual private network technologies to offer a method for delivering secure voice. Because VoIP transmits digitized voice as a stream of data, the VoIP VPN solution accomplishes voice encryption quite simply, applying standard data-encryption mechanisms inherently available in the collection of protocols used to implement a VPN.
Mobile VoIP or simply mVoIP is an extension of mobility to a voice over IP network. Two types of communication are generally supported: cordless telephones using DECT or PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using 3G or 4G protocols.
H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.
Mp3/3gpp/wav
An IP PBX is a system that connects telephone extensions to the public switched telephone network (PSTN) and provides internal communication for a business. An IP PBX is a PBX system with IP connectivity and may provide additional audio, video, or instant messaging communication utilizing the TCP/IP protocol stack.
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. Most unified communications applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard.
SunComm Technology is a Taiwan multinational computer technology and GSM Voice over IP gateway manufacturer. The main products in 2010 focused on GSM VoIP gateways & IP surveillance camera devices. Core members have been engaging in the communication & networks industry since 1977.
Call recording hardware, or a telephone recorder, is hardware that can be used to record telephone conversations. Call recording hardware is most often used by law enforcement, lawyers, journalist, and call centers to record phone transaction with customers.
The 1100-series IP phones are 6 different desktop IP clients manufactured by Avaya for Unified communications which can operate on the SIP or UNIStim protocols. The SIP Firmware supports presence selection and notification along with secure instant messaging.
A VoIP gateway is a gateway device that uses Internet Protocols to transmit and receive voice communications (VoIP).