Multiplex communications – Pathfinding or routing – Switching a message which includes an address header
Reexamination Certificate
1999-09-29
2004-09-14
Nguyen, Chau (Department: 2663)
Multiplex communications
Pathfinding or routing
Switching a message which includes an address header
C370S394000, C370S401000, C370S474000
Reexamination Certificate
active
06791982
ABSTRACT:
TECHNICAL FIELD
The present invention relates to data communication over a network, more particularly to a segmentation method used for transmission of large data packets.
BACKGROUND
Recent advances in hardware and communication technologies have introduced the era of mobile computing over wired and wireless links. The proliferation of powerful notebook computers and wireless communications promises to provide users with network access at any time and in any location over the Internet. This continuous connectivity will allow users to be quickly notified of changing events and provide them with the resources necessary to respond to them even when in transit.
In mobile networks, such as that proposed by Internet Engineering Task Force (IETF), a mobile host is allowed to roam freely on the Internet while still maintaining the same IP address. In such systems, data transfer delay requirements are critical and transmissions must support efficient transport. These requirements are even more critical for real-time applications, such as voice or video. The Internet community has a well-developed and mature set of layered transport and network protocols, which are quite successful in offering to end-users both connection-oriented transport protocols, such as Transport Control Protocol (TCP), and connectionless transport protocols, such as User Datagram Protocol (UDP), over connectionless network services, such as Internet Protocol (IP). Many popular network applications have been built directly on top of the TCP and UDP over the past decade. These have helped these Internet services and protocols to become widely-spread de facto standards.
Interconnection layer protocols and interfaces there between are defined to provide specifications for communication between a process or program being executed on one host computer's operating system and another process or program running on another computer. Transmission control protocol/internet protocol (TCP/IP) are two protocols that are part of a protocol suite or family of protocols layered and designed to connect computer systems that use different operating systems and network technologies.
FIG.
1
(
a
) illustrates conceptual layers for TCP/IP as well as the format of objects passed between adjacent protocol layers. TCP/IP is a four layer protocol suite (the hardware layer is not counted) which facilitates interconnection on the same or different networks, and in certain networks such as the Internet, is a requirement for interoperability. TCP, which is a transport layer protocol, is used to access applications on other hosts, and IP permits identification of source and destination addresses for communication between hosts on the same or different networks. The fundamental internetwork service consists of a packet delivery system, and the internetwork protocol (IP) defines that delivery mechanism, i.e., the basic unit of data transfer.
The basic data transfer unit is often called a “datagram” as is well known in the art and is divided into header and data areas, as shown in FIG.
1
(
b
). The header contains source and destination addresses and a type field that identifies the contents of the datagram. For example, a UDP header consists of a UDP source port and UDP destination port. A UDP message length field indicates the number of octets in a UDP datagram, and a UDP check sum provides an optional checksum of UDP and some parts of the IP header. The IP protocol only specifies the header format including the source and destination IP addresses; it does not specify the format of the data area.
The IP protocol also performs a routing function by choosing a path over which data will be sent. Using special procedure called routing protocols, routers exchange information among themselves and the hosts to which they are connected. This allows them to build tables, called routing tables, which are used to select a path for any given packet from a source to a destination. Although there can be more than one router along the path, each router makes only an individual forwarding decision as to which is the next host or router, i.e., the next network hop. This method is called hop-by-hop routing and is distinguished from end-to-end protocol that is implemented at transport through application layers.
Forwarding decisions at each node are based on fields within the IP header and based on entries in the nodes's IP routing table. FIG.
1
(
c
) illustrates a standard IP header which consists of a number of predefined fields. Some of the fields in IP header remain constant throughout the path between the source and destination. For example, fields SOURCE IP ADDRESS and DESTINATION IP ADDRESS, which, in IPv4, contain the thirty-two bit IP addresses of the datagram sender and intended recipient, remain unchanged throughout the path. As each node makes its forwarding decision, other IP header fields, may change according to a constant parameter, for example, sequentially, or they may change in a more unpredictable way.
In order to carry data that has real-time properties, a protocol known as Real-time Transport Protocol (RTP) is defined for providing end-to-end delivery services, such as interactive audio and video, with a growing interest in using RTP as one step to achieve interoperability among different implementations of network audio/video applications. The delivery services include payload type identification, sequence numbering, time stamping and delivery monitoring. Although RTP may be used with a number of suitable underlying network or transport protocols, such as TCP, applications typically run RTP on top of UDP to make use of its multiplexing and checksum services, with both RTP and UDP protocols contributing parts of the transport protocol functionality. For Internet environment, of course, the underlying network service or layer for such session and transport layers is the IP.
As stated above, over end-to-end connections, each of the RTP, UDP, or IP has an overhead associated with corresponding headers, with header overhead for RTP, UDP, and IPv4 being 12 bytes, 8 bytes and 20 bytes, respectively, for a total of 40 bytes of combined header overhead. Occasionally, this 40-byte combined overhead is larger than the actual payload itself. Because a large transmission bandwidth is required to accommodate such a large overhead, especially over low speed lines, such as dial-up modems at 14.4 or 28.8 kb/s, a header compression technique has been proposed as an IETF Standard Protocol by Casener et al. titled “Compressing IP/UDP/RTP Headers for Low-Speed Serial Links,” February 1999. This document is identified by IETF as Request for Comment 2508 (hereinafter referred to as RFC 2508) and is hereby incorporated by reference. Similar to TCP header compression, the proposed IP/UDP/RTP header compression in RFC 2508 relies partly on the assumption that some of the bytes in headers remain constant over the life of the connection. Moreover, differential coding on changing header fields is used to reduce their size and to eliminate the changing fields entirely for common cases by calculating the changes from a previous packet length, as indicated by the underlying link-level protocol.
The header compression of RFC 2508 offers a reduction in the combined compression of IP, UDP and RTP headers to two bytes for packets when UDP checksums is not sent, or four bytes when UDP checksums is sent. Although the proposed compression may be applied to the RTP header alone on an end-to-end basis, the compression of the combination of IP, UDP and RTP headers on a link-by-link basis is preferred because the resulting header overhead is approximately the same (2-4 bytes) in either case, and because compressing on a link-by-link basis provides better performance due to lower delay and loss rate.
The use of IP/UDP/RTP compression over a particular link is a function of the link-layer protocol, which defines negotiation rules for reliable transfer of data packets between two nodes. One known link layer protocol is the Point-to-Point Protocol (PPP) which provides a stan
Hyun Soon-Dong
Nguyen Chau
Telefonaktiebolaget LM Ericsson
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