Title Of Paper Streaming data transmission between multiple points in a network. Architecture and Protocols – An approach. Authors Rajib Kumar Saha Kaniska Mandal Speaker Kaniska Mandal
Real time streaming data transmission technology offers an opportunity to design global multimedia communication system between different clients in a network. Internet Telephony, Voice over IP are some applications of this technology. Introduction
Through this paper it is our endeavor to describe the basic architecture and application layer protocols that are specific to our application model for streaming data transmission . In this paper we have proposed a new Connection initiation scheme - Connection Initiation Protocol(CIP) to provide better performance in terms of network traffic and real time delivery of streaming data.
Figure 1: the protocol stack for this application.
CIP encapsulates the basic connection management tasks. CIP is used along with RTP (for streaming data transmission) and RTCP (for Control Signaling). RTP is used to transmit audio/video over unicast or multicast packet network service. RTP provides mechanisms for handling the problems of jitter and loss, timing recovery and intermedia synchronization. RTP implements audio/video encoding schemes(MPEG-2/ADPCM). Application Layer Protocols
RTCP is used for controlling the media streams. It monitors the QOS at the Receiver, conveys the reports and information about other participants to the sender. It accompanies RTP. SIP is an application layer protocol that can be used to establish, modify and terminate multimedia sessions with one or more participants. These sessions can be Internet Telephony call,videoconferencing or a distance-learning session . CIP resembles with SIP (Session Initiation Protocol) in certain operations but differs from it in the basic architecture.
Basic Operation This application integrates CIP, RTP and RTCP . In the application layer, the Connection Manager operates according to the rules of CIP (Connection Initiation Protocol). Client can originate “LOGGING” request using UDP connection. Accordingly, the Connection manager includes the current user( name,ip address,media info.) in the list of “available clients”. On successful logging and registration, the client receives a list of users already logged in. Connection manager maintains this list and sends a dynamically modified copy to each of the logged client.
After receiving the list of available clients, an user can send an “INVITE” request to another available client through the Connection Manger using UDP. Subsequently the inviting client opens an RTCP port to listen to the “ACKNOWLEDGE” from the actual recipient of the “INVITE” request .
Connection manager forwards the “INVITE” request to the proper destination. The invited client sends the “ACKNOWLEDGE” response to the inviting client as the first RTCP packet. On receiving the acknowledgement, an RTP session is to be established between the participating clients for the streaming data transfer.
1.Unlike an application using SIP, this system using CIP establishes a virtual communication channel between the participating parties before transmitting streaming data. This ensures shorter path to be traversed by the data packets.
In case of an audio conference , an RTP session is identified by an IP multicast group address and a pair of consecutive UDP port numbers. The first port no.(even) is for the RTP audio stream and the other port no.(odd) is for the RTCP stream. Features of streaming data transfer model ( implemented by this approach ) :
2.The establishment of RTP session and transmission of RTP,RTCP streams do not involve CIP server(Connection Manager). This approach reduces network traffic and ensures improved performance in terms of real time delivery of streaming data.
3.On the contrary, in SIP-based system all the requests from the client during streaming data transfer are to be passed through SIP server leading to heavy network traffic.
4.This application can invite multiple parties in a common conference using Multicast sockets as CIP supports multicasting.
JMF - The Java TM Media Framework is an API for incorporating media data such as audio and video into Java applications and applets. It is a powerful tool for implementing RTP and RTCP protocols.
Implementation Tools Scope of usage :
This real-time data transfer model can be implemented in any IP network to transmit media streams cost-effectively. Implementation is platform independent. It has a huge potential for large-scale usage. We have already observed how this system improves response time, offers shorter path and reduces network traffic.