Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching
Reexamination Certificate
1999-05-28
2004-08-31
Nguyen, Hanh (Department: 2662)
Multiplex communications
Pathfinding or routing
Combined circuit switching and packet switching
C370S394000, C370S400000, C704S200000
Reexamination Certificate
active
06785261
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to data transmission systems and more particularly to a method and apparatus for facilitating correction of data loss in such a system. The invention is suitable for use in any telecommunications network or transmission path that includes an end-to-end or node-to-node connection for communication of multiple data streams between a pair of devices.
By way of example, and without limitation, the invention will be described in the context of transmitting packet based real time voice, video, both voice and video, or other media signals over a packet switched computer network, for use in internet-based telephony (e.g., voice over IP (VoIP)). These are generally referred to herein as multimedia signals. However, the invention may also be suitably employed to transmit other types of signals and over other networks (such as local area (LAN), metropolitan area (MAN) or wide area (WAN) networks, and circuit switched networks, for example) or direct end-to-end connections, as well as with other transmission protocols.
2. Description of the Related Art
Packet switched networks now provide interactive communications services such as telephony and multi-media conferencing. In the context of packet switched networks operating according to the Internet Protocol (IP), this technology is presently known as internet telephony, IP telephony or, where voice is involved, Voice over IP (VoIP).
VoIP presents an attractive technology for use in long distance telephone calls, as compared to the public switched telephone network (PSTN), which has been the traditional transmission medium. The advantage of VoIP calls over PSTN calls is cost. In the United States, for instance, long distance service providers for the PSTN provide domestic services at rates ranging from roughly 10 to 30 cents per minute, and international rates for substantially more, depending on the time of day, day of the week, and the distances involved. In contrast, the cost of a VoIP call anywhere in the world is potentially the cost of a local telephone call to a local internet telephony service provider at one end and the cost of a local call from an internet telephony service provider at the far end to the destination telephone. Once the call is routed from the local VoIP provider onto the IP network, the cost to transmit the data from the local internet telephony provider to the far end internet telephony provider can be free for all practical purposes, regardless of where the two parties are located, Similarly, the cost to facilitate a direct dial internet telephony call can theoretically be free, except for possible access fees charged by local exchange carriers. VoIP service providers can thus potentially charge users far less for VoIP calls than the users would pay for comparable calls placed strictly over the PSTN.
a. Packet Switched Network Communications
In a packet switched network, a message to be sent is divided into blocks, or data packets, of fixed or variable length. The packets are then sent individually over the network through multiple locations and then reassembled at a final location before being delivered to a user at a receiving end. To ensure proper transmission and re-assembly of the blocks of data at the receiving end, various control data, such as sequence and verification information, is typically appended to each packet in the form of a packet header. At the receiving end, the packets are then reassembled and transmitted to an end user in a format compatible with the user's equipment.
To facilitate packet-based communication over interconnected networks that may include computers of various architectures and operating systems, the networks and computers typically operate according to an agreed set of packet switching protocols. A variety of such protocols are available, and these protocols range in degree of efficiency and reliability. Those skilled in the art are familiar, for instance, with the Transport Control Protocol/Internet Protocol (TCP/IP) suite of protocols, which is used to manage transmission of packets throughout the Internet and other packet switched networks.
Each protocol in the TCP/IP suite is designed to establish communication between common layers on two machines, or hosts, in the network. The lowest layer in the Internet is the “physical” layer, which is concerned with ensuring that actual bits and bytes of information pass along physical links between nodes of the network. The next layer is the link layer, which ensures a reliable connection between nodes in the network. The next layer is the “network” or “IP” layer, which is concerned with permitting hosts to inject packets of data into the network to be routed independently to a specified destination. The next layer in turn is the “transport” layer, which is concerned with allowing peer entities on source and destination hosts to carry on a conversation. Generally speaking, the IP and transport layers of the Internet are not concerned with the physical arrangement of the network, such as whether source and destination machines are on the same sub-network or whether there are other sub-networks between them.
The transport layer of TCP/IP can utilize two end-to-end protocols, TCP (Transport Control Protocol) and UDP (User Datagram Protocol). TCP is a reliable connection-oriented protocol, which includes intelligence necessary to confirm successful transmission between the sending and receiving ends in the network. UDP, in contrast, is an unreliable connectionless protocol, which facilitates sending and receiving of packets but does not include any intelligence to establish that a packet successfully reached its destination. In general, UDP is used by applications that do not want TCP's sequencing or flow control and wish to provide their own.
According to UDP, the transport layer takes a data stream to be transmitted and breaks it up into independent connectionless segments or “datagrams.” UDP adds to each of these packages an 8 byte header, which includes overhead information such as a source port number, a destination port number and a length and a checksum designed to allow the receiving end to properly reassemble the datagrams into the original message. The transport layer then “passes” each of these packages to the IP layer.
The IP layer in turn adds another header to each package, providing additional overhead information, such as a source IP address and a destination IP address. The IP layer then transmits the resulting packages through the Internet, possibly fragmenting each package into pieces as it goes. As the pieces of the package finally reach the destination machine, they are reassembled by the IP layer and passed to the transport layer.
For real time data or media signals (such as voice or video) to be transmitted over packet switched networks, the packets to be transmitted may be encapsulated by one or more additional header layers according to established higher level protocols. An example of one such higher level protocol is Real Time Protocol or RTP. RTP may provide each packet with at least a 12 byte header containing timestamps and sequence numbers. Included in this header may be a 7 bit payload type, which may define the type of payload in the underlying data packet. In practice, when the transmitting and receiving network ends establish communication of such signals, they will negotiate a mutually acceptable meaning for these RTP payload types. By way of example, the RTP payload type may indicate the type of voice or video codec (e.g., G.729, G.723.1, etc.) used to compress the underlying media signal, thereby facilitating proper decoding at the receiving end.
Packet switched networks such as the Internet thus serve to provide end-to-end (or node-to-node) communication between a pair of network devices or machines. These network devices may access or be connected to the Internet through any suitable configuration. In a usual arrangement, for instance, each device is connected via a communications link (such as the pu
Borella Michael S.
Kostas Thomas J.
Schuster Guido M.
Sidhu Ikhlaq S.
3Com Corporation
McDonnell Boehnen & Hulbert & Berghoff LLP
Nguyen Hanh
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