Abstract
The development of internet technology which is rapidly pushing towards convergence with other communications technology. Namely VoIP (Voice Internet Protocol) technology which is able to pass the voice traffic, video and data in the packet network via IP (Internet Protocol).
Through this technology, VoIP can be designed that allows many people to see each other through the screen with the available transmit video in a large amount of bandwidth, which was then known as the videoconference.
Designing videoconference via VoIP network device support will be discussed in any build a network, through the launch of the network type, protocol tandadisasi, audio and video codecs and display.
I INTRODUCTION
I.1 Background Problems
The development of internet technology at this time as we encourage students to develop and use more functionality of the system. After the occurrence of the Voice over Internet Protocol or VoIP is well known to the teleconference with his, now with the capacity to provide greater bandwidth plus the equipment and support infrastructure inside, through the IP network could be equipped with VoIP display image among users. Namely videoconference with the H.323 protocol standard that is still a key to videoconferencing over IP.
I.2 Objectives and Benefits of Writing
Videoconferencing has been used to separate the area of communication, such as the nuclear facilities, operating room, and board space. Videoconferencing allows for the process of learning to teach as a lecturer with the students, who intalasi close to the volcano, a remote lighthouse, telescope and microscope. Videoconference was also used to track wildlife in natural habitats, to perform the supervision and security, combined with micro-instrumentation, to browse in the human body. Side of videoconferencing mungikin not come quickly as the increase of communication simple, but this can become a better advantage.
II. Formulation MATTERS
In the making of this proposal, will be made in the design of a videoconference-based Internet Protocol (IP). In this design involves a variety of infrastructure which is the basic technology, Voice over Internet Protocol (VoIP) through the process of digitizing, compression, and pemaketan audio-video data. Then the design will be formulated on this issue:
Technical VoIP
Set up and run the component videoconference
III. REFERENCES study
III.1 VoIP
Voice over Internet Protocol (VoIP) technology is able to pass the voice traffic, video and data in the packet through the IP network. IP network itself is a data communication network based packet-switch, so bertelepon use in the IP network or the Internet. VoIP can be installed in any ethernet and IP address, unlike the conventional telephone port that should have retired in the Central or PBX.
The development of internet technology which is rapidly pushing towards convergence with other communications technology. Standards in the communication protocol such as H.323 VoIP technology has enabled communication integrated with other communication networks such as PSTN.
III.2 Network Type
Standard IP Videoconferencing is designed for videoconferencing that took place on a data network without the standard quality-of-service, such as the Internet. Such a network is not used for sending data in real-time applications. Network data used for various purposes: e-mail, web browsing, and other activities that are used with IP videconferencing.
There are five key issues for network vidoconferencing on the network such as the Internet. They are the data rate, packet loss, latency, jitter and policies.
• Data Rate
Rate is the basic data that will provide adequate space in a flow network for all the packages that can be freely through. For any reason, ISDN videoconference menggunaakn around 128-384 Kbps. H.323 video-based system can use the same data rate, although in general tend to be higher for less expensive network, the data rate of 384-768 Kbps is very common. Data rate needed for videoconferencing speed is higher than in the IP network on the ISDN network. About 20% overhead. For example an IP Conference at a speed of 384 Kbps actually used data rate around 450 Kbps. A higher quality of H.323 and SIP videoconference can reach 1.5-3.0 Mbps, and if you want to have a quality broadcast with modifier 1/2/4 codecs include MPEG and MJPEG, the limit of 6-20 Mbps for transmission NTSC / PAL, 20 -- 50 Mbps for HDTV recorded earlier and higher for live broadcast.
Rate data should be symmetrical. Means that each end point / end point should be able to send and receive the same amount of data to the call / call. This connection speed, 384 Kbps plus overhead as an example, if the maximum call data rate / call. This connection speed Kbps plus overhead as an example, if the maximum data rate calls / call akan use in any direction. Kecapatan actually vary depending on how the video images or static active. When a call is activated, there will be a spike in the use of bandwidth as the entire video frame is sent, then the die will fall or when the image has been updated has been sent. A graph illustrating the rate of data throughout the initialization of a call / call 384 Kbps shown in the picture below. This image shows only outbound traffic, with basically the video after the initiation, with such a significant change in the bandwidth used after the call / call has started.
In the videoconference multipoint should always note that the MCU / bridge view all the streams at the same time. So if there are 8 locations videoconference running on 384 Kbps data rate, each location to send and receive MCU of 384 Kbps, and the MCU to receive and distribute 8 x 384 Kbps = 3 Mbps.
• Packet Loss (Packet Loss)
Packet loss / packet loss occurs when the package failed to arrive with the right. This can be caused by a lack of bandwidth along the path (when the impasse occurs, routers will stop the package), or perhaps a mistake or an error happens in the transmission. Errors occur most commonly on the wireless connection, such as microwave, satellite or wireless Ethernet Local. In fact, can also occur in the transmission or copper fibel relationship. Impact of packet loss on the bottom such as "tiling / tilling" in the video window, or lose the discount on empty area in the video window, and / or interference in the audio.
• latency / delay time
Latency is the time antaa an event that happened and see that the remote end. Latency introduced through the process of encoding / decoding, and therefore depends on the equipment used, and also at the time of taking the package to intersect the network. Optimally, a connection from the United States to Europe in the fiber optics network will have a time delay 90 ms (seperseribu seconds). By using a satellite connection can be up to 200 ms.
• Jitter
Is a random variation in the latency in the connection process to compete in the terminal (for example on the PC desktop), traffic to impede the flow through the routers along the path, or even flow changes during videoconference network. Random variation of this is one of the few things that cause the package to arrive from the damaged ordered.
• Pollicies / Policy
Policies introduced by things such as firewalls and Network equipment Addres Translation (NAT), which is usually used to hide or business to protect network elements from the wider Internet.
III.3 Protocol Standardization
III.3.1 H.323
For effective communication, a language that is common between two or more users must be determined. Without the same language, misunderstandings will occur. Created on a computer network standards and protocols, so that applications that depend on the network can operate with both. There are several organizations working to develop these standards. are ITU (International Telecommunication Union), the organization that develops international standards on the computer, radio, and other equipment to interact each other.
The standards of family H.32x encompass multimedia communications. This family includes H.320 (communication over ISDN [Integrated Service Digital Networks]) and H.324 (Communication through SCN [circuit switched Digital Networks], with a regular traditional phone service).
H.323 is a standard communication made by the ITU, appear in 1996, and aims to combine multimedia communication area through LAN (Local Area Networks). This is the development of H.320 technology, but more for the internet. H.323 has been reviewed for entry into the voice over IP and IP telephony, such as communication-to-Gatekeeper Gatekeeper and other data communications that include a packet-based network. This network, including IP network such as the Internet, Internet Packe Exchange (IPX) LAN and WAN's. H.323 is widely supported by various vendors and used throughout the world through education and commercial markets.
Standard H.323 explain information about the parts of components that interact with the environment H.323. This explains the parts to provide complete communication services:
Terminals, either PC or standalone equipment, this is the endpoints of the communications path.
Gatekeepers, the brain's network, providing services such as experience or identification, authorization and management bandwdth.
Gateway, which provides services such as translator when connecting different networks (such as H.324 for example)
MCU's (Multipoint Control Units), which enables multipoint conferencing, or communication between more than two users in one time. (More like a traditional conference on the phone).
In the case of type component, also describes the standard H.323 protocol, audio and video codecs, RAS (registration, admission, and status), call signaling, control and signaling. H.323 explain keberadan level of fulfillment and support the above specification for all terminals in the network.
III.4 Audio and Video Display
III.4.1 Web-Cam
Camera with high quality, color camera with a fast action pan / tilt / zoom and focus are useful for displaying images to the opponents to speak.
III.4.2 Mikrofon-Speaker/Headset
With high-quality headset for videoconferencing equipped with a microphone and speaker that allows you to listen and talk to the opponent to speak.
III.4.3 Personal Computer (PC)
Device equipped with a computer monitor that allows you to display the display image.
III.5 Codecs
Codec is the heart of the system. Main tasks of the Codec is compressed audio, video and data compressed out and the information came from.
System software is simple, usually working with a webcam and a headset and are very quick to install. This system provides quality and convenience to videoconferencing personally. Example: Netmeeting, VCON vPoint.
The development of internet technology which is rapidly pushing towards convergence with other communications technology. Namely VoIP (Voice Internet Protocol) technology which is able to pass the voice traffic, video and data in the packet network via IP (Internet Protocol).
Through this technology, VoIP can be designed that allows many people to see each other through the screen with the available transmit video in a large amount of bandwidth, which was then known as the videoconference.
Designing videoconference via VoIP network device support will be discussed in any build a network, through the launch of the network type, protocol tandadisasi, audio and video codecs and display.
I INTRODUCTION
I.1 Background Problems
The development of internet technology at this time as we encourage students to develop and use more functionality of the system. After the occurrence of the Voice over Internet Protocol or VoIP is well known to the teleconference with his, now with the capacity to provide greater bandwidth plus the equipment and support infrastructure inside, through the IP network could be equipped with VoIP display image among users. Namely videoconference with the H.323 protocol standard that is still a key to videoconferencing over IP.
I.2 Objectives and Benefits of Writing
Videoconferencing has been used to separate the area of communication, such as the nuclear facilities, operating room, and board space. Videoconferencing allows for the process of learning to teach as a lecturer with the students, who intalasi close to the volcano, a remote lighthouse, telescope and microscope. Videoconference was also used to track wildlife in natural habitats, to perform the supervision and security, combined with micro-instrumentation, to browse in the human body. Side of videoconferencing mungikin not come quickly as the increase of communication simple, but this can become a better advantage.
II. Formulation MATTERS
In the making of this proposal, will be made in the design of a videoconference-based Internet Protocol (IP). In this design involves a variety of infrastructure which is the basic technology, Voice over Internet Protocol (VoIP) through the process of digitizing, compression, and pemaketan audio-video data. Then the design will be formulated on this issue:
Technical VoIP
Set up and run the component videoconference
III. REFERENCES study
III.1 VoIP
Voice over Internet Protocol (VoIP) technology is able to pass the voice traffic, video and data in the packet through the IP network. IP network itself is a data communication network based packet-switch, so bertelepon use in the IP network or the Internet. VoIP can be installed in any ethernet and IP address, unlike the conventional telephone port that should have retired in the Central or PBX.
The development of internet technology which is rapidly pushing towards convergence with other communications technology. Standards in the communication protocol such as H.323 VoIP technology has enabled communication integrated with other communication networks such as PSTN.
III.2 Network Type
Standard IP Videoconferencing is designed for videoconferencing that took place on a data network without the standard quality-of-service, such as the Internet. Such a network is not used for sending data in real-time applications. Network data used for various purposes: e-mail, web browsing, and other activities that are used with IP videconferencing.
There are five key issues for network vidoconferencing on the network such as the Internet. They are the data rate, packet loss, latency, jitter and policies.
• Data Rate
Rate is the basic data that will provide adequate space in a flow network for all the packages that can be freely through. For any reason, ISDN videoconference menggunaakn around 128-384 Kbps. H.323 video-based system can use the same data rate, although in general tend to be higher for less expensive network, the data rate of 384-768 Kbps is very common. Data rate needed for videoconferencing speed is higher than in the IP network on the ISDN network. About 20% overhead. For example an IP Conference at a speed of 384 Kbps actually used data rate around 450 Kbps. A higher quality of H.323 and SIP videoconference can reach 1.5-3.0 Mbps, and if you want to have a quality broadcast with modifier 1/2/4 codecs include MPEG and MJPEG, the limit of 6-20 Mbps for transmission NTSC / PAL, 20 -- 50 Mbps for HDTV recorded earlier and higher for live broadcast.
Rate data should be symmetrical. Means that each end point / end point should be able to send and receive the same amount of data to the call / call. This connection speed, 384 Kbps plus overhead as an example, if the maximum call data rate / call. This connection speed Kbps plus overhead as an example, if the maximum data rate calls / call akan use in any direction. Kecapatan actually vary depending on how the video images or static active. When a call is activated, there will be a spike in the use of bandwidth as the entire video frame is sent, then the die will fall or when the image has been updated has been sent. A graph illustrating the rate of data throughout the initialization of a call / call 384 Kbps shown in the picture below. This image shows only outbound traffic, with basically the video after the initiation, with such a significant change in the bandwidth used after the call / call has started.
In the videoconference multipoint should always note that the MCU / bridge view all the streams at the same time. So if there are 8 locations videoconference running on 384 Kbps data rate, each location to send and receive MCU of 384 Kbps, and the MCU to receive and distribute 8 x 384 Kbps = 3 Mbps.
• Packet Loss (Packet Loss)
Packet loss / packet loss occurs when the package failed to arrive with the right. This can be caused by a lack of bandwidth along the path (when the impasse occurs, routers will stop the package), or perhaps a mistake or an error happens in the transmission. Errors occur most commonly on the wireless connection, such as microwave, satellite or wireless Ethernet Local. In fact, can also occur in the transmission or copper fibel relationship. Impact of packet loss on the bottom such as "tiling / tilling" in the video window, or lose the discount on empty area in the video window, and / or interference in the audio.
• latency / delay time
Latency is the time antaa an event that happened and see that the remote end. Latency introduced through the process of encoding / decoding, and therefore depends on the equipment used, and also at the time of taking the package to intersect the network. Optimally, a connection from the United States to Europe in the fiber optics network will have a time delay 90 ms (seperseribu seconds). By using a satellite connection can be up to 200 ms.
• Jitter
Is a random variation in the latency in the connection process to compete in the terminal (for example on the PC desktop), traffic to impede the flow through the routers along the path, or even flow changes during videoconference network. Random variation of this is one of the few things that cause the package to arrive from the damaged ordered.
• Pollicies / Policy
Policies introduced by things such as firewalls and Network equipment Addres Translation (NAT), which is usually used to hide or business to protect network elements from the wider Internet.
III.3 Protocol Standardization
III.3.1 H.323
For effective communication, a language that is common between two or more users must be determined. Without the same language, misunderstandings will occur. Created on a computer network standards and protocols, so that applications that depend on the network can operate with both. There are several organizations working to develop these standards. are ITU (International Telecommunication Union), the organization that develops international standards on the computer, radio, and other equipment to interact each other.
The standards of family H.32x encompass multimedia communications. This family includes H.320 (communication over ISDN [Integrated Service Digital Networks]) and H.324 (Communication through SCN [circuit switched Digital Networks], with a regular traditional phone service).
H.323 is a standard communication made by the ITU, appear in 1996, and aims to combine multimedia communication area through LAN (Local Area Networks). This is the development of H.320 technology, but more for the internet. H.323 has been reviewed for entry into the voice over IP and IP telephony, such as communication-to-Gatekeeper Gatekeeper and other data communications that include a packet-based network. This network, including IP network such as the Internet, Internet Packe Exchange (IPX) LAN and WAN's. H.323 is widely supported by various vendors and used throughout the world through education and commercial markets.
Standard H.323 explain information about the parts of components that interact with the environment H.323. This explains the parts to provide complete communication services:
Terminals, either PC or standalone equipment, this is the endpoints of the communications path.
Gatekeepers, the brain's network, providing services such as experience or identification, authorization and management bandwdth.
Gateway, which provides services such as translator when connecting different networks (such as H.324 for example)
MCU's (Multipoint Control Units), which enables multipoint conferencing, or communication between more than two users in one time. (More like a traditional conference on the phone).
In the case of type component, also describes the standard H.323 protocol, audio and video codecs, RAS (registration, admission, and status), call signaling, control and signaling. H.323 explain keberadan level of fulfillment and support the above specification for all terminals in the network.
III.4 Audio and Video Display
III.4.1 Web-Cam
Camera with high quality, color camera with a fast action pan / tilt / zoom and focus are useful for displaying images to the opponents to speak.
III.4.2 Mikrofon-Speaker/Headset
With high-quality headset for videoconferencing equipped with a microphone and speaker that allows you to listen and talk to the opponent to speak.
III.4.3 Personal Computer (PC)
Device equipped with a computer monitor that allows you to display the display image.
III.5 Codecs
Codec is the heart of the system. Main tasks of the Codec is compressed audio, video and data compressed out and the information came from.
System software is simple, usually working with a webcam and a headset and are very quick to install. This system provides quality and convenience to videoconferencing personally. Example: Netmeeting, VCON vPoint.
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