Multimedia networking is to build the multimedia on network and distributed systems, so different users on different machines can share image, sound, video, voice and many other features and to communicate with each under these tools.

The Integrated Services working group in the IETF (Internet Engineering Task Force) developed an enhanced Internet service model that includes best-effort service and real-time service.

The Resource Reservation protocol (RSVP), together with Real time Transport Protocol (RTP), Real Time Control Protocol(RTCP), Real-Time Streaming Protocol (RTSP), provides a working foundation for this architecture that is a comprehensive approach to provide applications with the type of service they need in the quality they choose.

RSVP - Resource Reservation Protocol

RSVP is a network control protocol that allows Internet applications to obtain special qualities of-service (QoS's) for their data flows. This generally (but not necessarily) requires reserving resources along the data path(s).

RSVP is used to set up reservations for network resources. When an application in a host (the data stream receiver) requests a specific quality of service (QoS) for its data stream, it uses RSVP to deliver its request to routers along the data stream paths.

RSVP is responsible for the negotiation of connection parameters with these routers. If the reservation is setup, RSVP is also responsible for maintaining router and host states to provide the requested service.

Reservation at a node on the data flow path

Each node capable of resource reservation has several local procedures for reservation setup and enforcement. Policy control determines whether the user has administrative permission to make the reservation. In the future, authentication, access control and accounting for reservation will also be implemented by policy control. Admission control keeps track of the system resources and determines whether the node has sufficient resources to supply the requested QoS.

RTP --- Real-time Transport Protocol

Real-time transport protocol (RTP) is an IP-based protocol providing support for the transport of real-time data such as video and audio streams. The services provided by RTP include time reconstruction, loss detection, security and content identification.

RTP is primarily designed for multicast of real-time data, but it can be also used in unicast. It can be used for one-way transport such as video-on-demand as well as interactive services such as Internet telephony.

RTP is designed to work in conjunction with the auxiliary control protocol RTCP to get feedback on quality of data transmission and information about participants in the on-going session.

There are two transport layer protocols in the Internet protocol suite, TCP and UDP. TCP provides a reliable flow between two host. It is connection-oriented and thus can't be used for multicast. UDP provides a connectionless unreliable datagram service.

To use UDP as a transport protocol for real-time traffic, some functionality has to be added.

Functionality that is needed for many real-time applications is combined into RTP, the real-time transport protocol. RTP is standardized in RFC 1889. Applications typically run RTP on top of UDP as part of the transport layer protocol, as shown in Figure below:

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 datagram network.

RTCP---Real-Time Control Protocol

RTCP is the control protocol that works in conjunction with RTP. It provides support for real time conferencing for large groups within an internet, including source identification and support for gateways and multicast-to-unicast translators. It is standardized in RFC 1889 and RFC 1890. RTCP performs the following four functions

Provide information to application:

The primary function is to provide information to an application regarding the quality of data distribution. Experiments with IP multicasting have established the importance of user feedback from RTCP to diagnose distribution faults.

Identify RTP source:

RTCP carries a transport-level identifier for an RTP source, called the canonical name (CNAME). This CNAME is used to keep track of the participants in an RTP session

Control RTCP transmission interval:

To prevent control traffic from overwhelming network resources and to allow RTP to scale up to a large number of session participants, control traffic is limited to at most 5 percent of the overall session traffic.

Convey minimal session control information:

As an optional function, RTCP can be used as a convenient method for conveying a minimal amount of information to all session participants.

RTSP---Real-Time Streaming Protocol

The application-level Real Time Streaming Protocol, RTSP, aims to provide an extensible framework to enable controlled delivery of real-time data, such as audio and video. Sources of data can include both live data feeds, such live audio and video, and stored content, such as prerecorded events.

It is designed to work with established protocols such as RTP, HTTP, and others to provide a complete solution for streaming media over the Internet. It supports multicast as well as unicast. It also supports interoperability between clients and servers from different vendors.

RTSP provides the following specific benefits to Internet content providers and users:

• Bidirectionality enabling full stream control
• High reliability over current net infrastructure
• Low overhead data delivery
• Ready to fully exploit of emerging technologies and protocols (e.g. IP Multicast, RTP etc.)
• Security
• Intellectual property rights protection
• Scalability
• The design is based on field-proven techniques