It stands for an English Voice Over Internet Protocol (ie Voice over Internet Protocol) - is a way to bind voice conversations over the Ahabkp (Internet) or via any network that uses Internet Protocol Internet Protocol. And therefore can be any number of people connected together to a single network using Internet Protocol (IP) - such as the Internet - that Ithadthoa phone using this technique. [Edit] the principle of technical work
The technology converts audio signals analogue Analog Signals from the phone into digital signals Digital Signals are split this signal into packets Packets and uses the IP to send these packets of digital several paths through the same network data and upon the arrival of these packets to the specified destination (future) re-compilation packets sent in order to be heard clearly in contrast to the usual contacts are using one path is selected and if the other party (the future) a normal phone signal is converted back to acoustic signals in order to be understood in the future. [Edit] Steps technical work Convert analogue signals Analog Signals to Digital Signals Digital Signals. Compression packages are good (the bandwidth is too small), there are several Muaiq (protocols) can choose to compress the package, including more sophisticated, so do not get delay in the sound. The integration of voice packets within the packet data using Meevaq (Protocol) real-time RTP-Real Time Protocol. We need to contact the user signals (bell) ITU-T H323. At the receiver are analyzed package and extract data from them and convert digital signals to the audio again and sent to the phone. Should happen in real time Real Time in order not to get cut in the sound. [Edit] requirements necessary for the work of technical
You must use the PBX to determine the course of conversation. Convert audio signals Analog Signals to Digital Signals Digital Signals. The use of Internet Protocol IP. Must divide the digital signal into small pieces called packets digital transfer of several paths through the network data itself. You must press the packages are good (the bandwidth is too small), there are several protocols can choose to compress the package, including more sophisticated, so do not get delay in the sound. There should be a place for temporary storage to collect packets Buffer in order not to be a delay in the sound. Should happen in real time Real Time in order not to get cut in the sound. [Edit] technical standards for Voice over Internet Protocol
A set of rules and conditions that govern the process of telephone calls and are divided into: - [Edit] Closed systems And are based on standards (closed not free source) programs such as Skype and IP Cisco famous (Skinny Client Control Protocol SCCP), a protocol is closed to control Balpartyyat developed mainly by the company Celsius Selsius Corporation, owns and develop specifications now Cisco Cisco System Inc. Among the most famous equipment operating in accordance with this Protocol Series, Cisco 7900. [Edit] Open Systems Includes open standards-based protocols such as open-source: - Protocol H.323: - Is a set of standard protocols emanating from the system developed by the International Telecommunication Union ITU-T for the NFL audio and images via computer network packets using the Packet-Based This protocol is used in the majority of popular applications such as NetMeeting. Protocol SIP: - Is an acronym for Phase Initiation Protocol, a protocol for telephone signals associated with Internet protocols, which are used to start, modify, and terminate phone calls from the type of VOIP. The teams have developed Internet Engineering Task Force IETF to this Protocol, have been published on the RFC 3261 protocol at the beginning. Possible for SIP to describe the communication is necessary to start the phone call became the start of phase protocol as a boom in the world of mobile VOIP. This Protocol is largely semi-protocol HTTP, the protocol that script, and easy to understand, and flexible to use, but it was resolved the SIP protocol H323 replaced the standard in use on a large scale.
Protocol IAX2: - A protocol to communicate between the programs, a program Asterisk Asterisk Open Source PBX telephone and provides voice over Internet protocol between Asterisk servers Asterisk and IAX2 clients and the package well under pressure. To get an idea of the size of waste produced during the transfer of voice over the Internet, remember that the audio CD, which occupies an area of 5.6 kilobits per second will need to 18 kilobits per second of bandwidth, comprising the difference between 5.6 Kbps and 18 Kbps of the headers of packets that will carry the data. This header contains all the necessary information (such as IP Address) for the transfer of voice packets to the future. We have the IAX2 protocol to reduce this waste is a wonderful bits by determining the amount of the additional permitted use for each package, and also took advantage of the principle of compilation of the talks sent to the same destination and included in the same packages.
The technology converts audio signals analogue Analog Signals from the phone into digital signals Digital Signals are split this signal into packets Packets and uses the IP to send these packets of digital several paths through the same network data and upon the arrival of these packets to the specified destination (future) re-compilation packets sent in order to be heard clearly in contrast to the usual contacts are using one path is selected and if the other party (the future) a normal phone signal is converted back to acoustic signals in order to be understood in the future. [Edit] Steps technical work Convert analogue signals Analog Signals to Digital Signals Digital Signals. Compression packages are good (the bandwidth is too small), there are several Muaiq (protocols) can choose to compress the package, including more sophisticated, so do not get delay in the sound. The integration of voice packets within the packet data using Meevaq (Protocol) real-time RTP-Real Time Protocol. We need to contact the user signals (bell) ITU-T H323. At the receiver are analyzed package and extract data from them and convert digital signals to the audio again and sent to the phone. Should happen in real time Real Time in order not to get cut in the sound. [Edit] requirements necessary for the work of technical
You must use the PBX to determine the course of conversation. Convert audio signals Analog Signals to Digital Signals Digital Signals. The use of Internet Protocol IP. Must divide the digital signal into small pieces called packets digital transfer of several paths through the network data itself. You must press the packages are good (the bandwidth is too small), there are several protocols can choose to compress the package, including more sophisticated, so do not get delay in the sound. There should be a place for temporary storage to collect packets Buffer in order not to be a delay in the sound. Should happen in real time Real Time in order not to get cut in the sound. [Edit] technical standards for Voice over Internet Protocol
A set of rules and conditions that govern the process of telephone calls and are divided into: - [Edit] Closed systems And are based on standards (closed not free source) programs such as Skype and IP Cisco famous (Skinny Client Control Protocol SCCP), a protocol is closed to control Balpartyyat developed mainly by the company Celsius Selsius Corporation, owns and develop specifications now Cisco Cisco System Inc. Among the most famous equipment operating in accordance with this Protocol Series, Cisco 7900. [Edit] Open Systems Includes open standards-based protocols such as open-source: - Protocol H.323: - Is a set of standard protocols emanating from the system developed by the International Telecommunication Union ITU-T for the NFL audio and images via computer network packets using the Packet-Based This protocol is used in the majority of popular applications such as NetMeeting. Protocol SIP: - Is an acronym for Phase Initiation Protocol, a protocol for telephone signals associated with Internet protocols, which are used to start, modify, and terminate phone calls from the type of VOIP. The teams have developed Internet Engineering Task Force IETF to this Protocol, have been published on the RFC 3261 protocol at the beginning. Possible for SIP to describe the communication is necessary to start the phone call became the start of phase protocol as a boom in the world of mobile VOIP. This Protocol is largely semi-protocol HTTP, the protocol that script, and easy to understand, and flexible to use, but it was resolved the SIP protocol H323 replaced the standard in use on a large scale.
Protocol IAX2: - A protocol to communicate between the programs, a program Asterisk Asterisk Open Source PBX telephone and provides voice over Internet protocol between Asterisk servers Asterisk and IAX2 clients and the package well under pressure. To get an idea of the size of waste produced during the transfer of voice over the Internet, remember that the audio CD, which occupies an area of 5.6 kilobits per second will need to 18 kilobits per second of bandwidth, comprising the difference between 5.6 Kbps and 18 Kbps of the headers of packets that will carry the data. This header contains all the necessary information (such as IP Address) for the transfer of voice packets to the future. We have the IAX2 protocol to reduce this waste is a wonderful bits by determining the amount of the additional permitted use for each package, and also took advantage of the principle of compilation of the talks sent to the same destination and included in the same packages.
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