# A Reliable Scheme for Synchronizing Multimedia Data Streams under Multicasting Environment
## 2 Related Work  
Computer networks were designed to connect computers on different locations so that they can share data and resources with each other. Initially, most of the data carried on networks were textual data but today in the current world, with the rise of multimedia and network technologies, multimedia data has become an indispensable feature on the Internet. Animations, voice and video clips (both real-time and streaming) have become more and more popular over the Internet [8]. Using the internet as a platform for multimedia communication gave birth to multimedia networking products like IP telephony, IP-TV, video conferencing, etc. In future, possibly people would enjoy more improved multimedia products in distance learning, distributed work groups, and in other real-time contextual services [8]. One fundamental characteristic of packet switched network is the delay incurred during packet delivery. Other issues include packet loss, packet corruption, and duplication and delayed arrivals. The current architecture of IP networks does not guarantee the Quality of Service (QoS) in terms of bandwidth, delay and packet loss [6]. In multimedia applications, the source packetizes the information and then transmit it over the network. These packets might get delayed, lost, the order changed and could be corrupted before received by the receiver. The receiver receives the packets and tries to reconstruct the original signal. Multimedia (audio/video) data is expected to be played back continuously at the sampling rate. It was also synchronized with other streams that require the timing relations to be maintained through the network. However, the packet switched networks distorts the timing relations between the packets. The natural behavior of IP networks demands some new transport protocols to take care of timing information and other multimedia data characteristics. Research in this area helped to reach the invention of media layer protocol named RTP/RTCP [8] to address properly the issues raised above. Currently, the key standard for media transport in IP networks is the RTP protocol framework. It provides transport and control services for multimedia data (audio/video) over IP networks. These services include timing information, source identification, quality feedback, synchronization, and membership management, etc. [9; 10]. As stated earlier that most of the multimedia applications use RTP/RTCP protocol to transport media data over the internet. RTP [7] is a real-time end-to-end transport layer protocol, mostly used upon UDP. RTP does not guarantee the delivery of packets in a timely manner, nor does it do the packet sequencing. RTP provides payload type identification, source identification, sequence numbering, and time stamping required most of the multimedia applications. The accompanying RTCP provides feedback on the quality of the data delivery and information about session participants [11]. An RTP session usually is composed of an RTP port number (UDP port), an RTCP port number (consecutive UDP port) and the participant's IP address. The RTP data transport is augmented by a control protocol (RTCP) [7], which provides the RTP session participants feedback on the quality of the data distribution. RTCP provides QoS provisioning, congestion control and estimating the session lengths. QoS monitoring information is directly embedded in the RTCP packet to handle these issues occurred with the original data during transmission over the network. The other most important protocol used in multimedia data transportation is Network Time Protocol (NTP) that is widely used to synchronize computer clocks on the Internet [12]. NTPv4 is widely used to synchronize system clocks among a set of distributed time servers and clients. The purpose of the NTP protocol is to convey timekeeping information from these primary servers to secondary time servers and clients via both private networks and the public Internet.