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6 From Video Into IP Packets
Published in Wes Simpson, Video Over IP, 2013
Transport protocols are used to control the transmission of data packets in conjunction with IP. We will discuss three major protocols commonly used in transporting real-time video:UDP, or User Datagram Protocol: This is one of the simplest and earliest of the IP protocols. UDP is often used for video and other data that is very time sensitive.TCP, or Transmission Control Protocol: This is a well-established Internet protocol that is widely used for data transport. The vast majority of the devices that connect to the Internet are capable of supporting TCP over IP (or simply TCP/IP).RTP, or Real-time Transport Protocol (or Real Time Protocol, if you prefer): This protocol has been specifically developed to support real-time data transport, such as video streaming.
Video Servers
Published in Jerry D. Gibson, The Communications Handbook, 2018
A. L. Narasimha Reddy, Roger Haskin
The research community and standards organizations have also been active in transporting multimedia over the Internet. Real-time transport protocol (RTP) defines an encapsulation of any of a number of audio or video data formats into UDP datagrams that allow it to be streamed over the Internet. Rather than retransmitting lost packets as does TCP, RTP timestamps packets to allow missing packets to be detected. RTP contains no flow-control mechanism and leaves the task of recovering from lost packets to the video/audio codec. A number of commercial products are based on RTP (Apple's QuickTime, Precept FlashWare, Netscape LiveMedia, Microsoft's Netmeeting and Real Player). A number of multimedia servers are available from major vendors, for example, IBM's videocharger.
Live Video and On-Demand Streaming
Published in Borko Furht, Syed Ahson, Handbook of Mobile Broadcasting, 2008
I. S. Venieris, E. Kosmatos, C. Papagianni, G. N. Prezerakos
The real-time transport protocol (RTP) provides end-to-end delivery services for data with realtime characteristics, such as video and audio. In general, multimedia applications require appropriate timing in data transmission and playing back. RTP provides time-stamping, sequence numbering, and other mechanisms to take care of the timing issues. Through these mechanisms, RTP provides end-to-end transport for real-time data over communication networks.
Design and implementation of a VoIP PBX integrated Vietnamese virtual assistant: a case study
Published in Journal of Information and Telecommunication, 2023
Hai Son Hoang, Anh Khoa Tran, Thanh Phong Doan, Huu Khoa Tran, Ngoc Minh Duc Dang, Hoang Nam Nguyen
The basic steps to make a call in VoIP are as follows: The caller determines where to call (e.g. country code, province code) and dials the number to call.Connections are established between the caller and the receiver.When speaking into a headset or microphone the voice produces an electromagnetic signal, which is an analogue signal. These are converted into digital signals using a unique algorithm. The digitized voice is then encapsulated and sent over the IP network. During the process, a protocol such as SIP or H323 controls the call by setting up, dialling, or disconnecting, for example. A Real-time Transport Protocol (RTP) is used to ensure reliability and to maintain the quality of service during transmission.Data are transferred over the initially established connection.Data containing the spoken sounds are converted back into sound that can be understood by the listener.Finally, the spoken sound is played on the receiver's side.