쥐스킨트의 작은 서재

System in package

A system in package (SiP) or system-in-a-package is a number of integrated circuits enclosed in a single module (package). The SiP performs all or most of the functions of an electronic system, and is typically used inside a mobile phone, digital music player, etc. Dies containing integrated circuits may be stacked vertically on a substrate. They are internally connected by fine wires that are bonded to the package. Alternatively, with a flip chip technology, solder bumps are used to join stacked chips together.

SiP dies can be stacked vertically or tiled horizontally, unlike slightly less dense multi-chip modules, which place dies horizontally on a carrier. SiP connects the dies with standard off-chip wire bonds or solders bumps, unlike slightly denser three-dimensional integrated circuits which connect stacked silicon dies with conductors running through the die.

Many different 3-D packaging techniques have been developed for stacking many more-or-less standard chip dies into a compact area.

An example SiP can contain several chips –such as a specialized processor, DRAM, flash memory – combined with passive components– resistors and  capacitors – all mounted on the same substrate. This means that a complete functional unit can be built in a multi-chip package, so that few external components need to be added to make it work.

Session Initiation Protocol

The Session Initiation Protocol (SIP) is acommunications protocol for signaling and controlling multimedia communicationsessions. The most common applications of SIP are in Internet telephony forvoice and video calls, as well as instant messaging, over Internet Protocol(IP) networks.

The protocol defines the messages that aresent between endpoints, which govern establishment, termination and otheressential elements of a call. SIP can be used for creating, modifying andterminating sessions consisting of one or several media streams. SIP is anapplication layer protocol designed to be independent of the underlyingtransport layer. It is a text-based protocol, incorporating many elements ofthe Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol(SMTP).

SIP works in conjunction with several otherapplication layer protocols that identify and carry the session media. Media identificationand negotiation is achieved with the Session Description Protocol (SDP). Forthe transmission of media streams (voice, video) SIP typically employs theReal-time Transport Protocol (RTP) or Secure Real-time Transport Protocol(SRTP). For secure transmissions of SIP messages, the protocol may be encryptedwith Transport Layer Security (TLS).

Real-time Transport Protocol

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.

RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction witha signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network.

Overview

RTP is designed for end-to-end, real-time, transfer of streaming media. The protocol provides facilities for jitter compensation and detection of out of sequence arrival in data, which are common during transmission on an IP network. RTP allows data transfer to multiple destinations through IP multicast. RTP is regarded as the primary standard for audio/video transport in IP networks and is used with an associated profile and payload format.

Real-time multimedia streaming applications require timely delivery of information and often can tolerate some packet lossto achieve this goal. For example, loss of a packet in audio application may result in loss of a fraction of a second of audio data, which can be made unnoticeable with suitable error concealment algorithms. The Transmission Control Protocol (TCP), although standardized for RTP use, is not normally used in RTP applications because TCP favors reliability over timeliness. Instead themajority of the RTP implementations are built on the User Datagram Protocol(UDP). Other transport protocols specifically designed for multimedia sessions are SCTP and DCCP, although, as of 2010, they are not in widespread use.

Protocol components

The RTP specification describes two sub-protocols, RTP and RTCP.

The data transfer protocol, RTP, facilitates the transfer of real-time data. Information provided by this protocol include timestamps, sequence numbers and the payload format which indicates the encoded format of the data.

The control protocol RTCP is used tospecify quality of service (QoS) feedback and synchronization between the media streams. The bandwidth of RTCP traffic compared to RTP is small, typically around 5%.