Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching
Reexamination Certificate
1998-06-05
2002-10-15
Chin, Wellington (Department: 2665)
Multiplex communications
Pathfinding or routing
Combined circuit switching and packet switching
C370S218000, C370S237000, C370S394000, C370S473000, C714S776000, C714S781000
Reexamination Certificate
active
06466574
ABSTRACT:
FIELD OF THE INVENTION
This invention relates to the field of real-time data/voice/media transmission over the internet, intranet, cable, and other any sort of packet switching networks. More specifically, the invention relates to a way to improve the quality of real-time packet transmission by using redundant transmission of packets.
BACKGROUND OF THE INVENTION
Internet Telephony and Internet Media transmission have huge business opportunities and many industry key players and major Telecom companies are rushing into this area. Many companies are marketing internet telephony gateway and internet telephony PC software. Companies are providing internet telephony services for low-cost long-distance calls and telecom companies are viewing Internet telephony as a way to unifi telephony and data infrastructure.
Internet Media transmission includes sending media packets (containing any of the following: n-dimensional images, animation, music, text, movies, video shots, still pictures, voice, data, etc.) over packet switching networks (e.g., a wide area network—WAN- and/or local area network—LAN) between two or more computers with special application software. Internet Telephony is a particular version of Internet Media where packets contain voice information (and sometimes video information). When the voice processed by an input device is captured at a source computer, an application running on the source computer will transform the continuous voice analog signals into a series of discrete digitally compressed packets. There are some well known industry standards to define this transformation process and the format of these discrete (often digitally compressed) packets, for example, PCM, GSM, G.723, etc.
There are other known processes defined by standards (e.g., IP, UDP, and RTP protocols) to augment the packets with necessary headers and trailers so that these packets can travel over the common packet switching network(s) to a destination computer. With these headers and trailers, packets usually travel over the packet switching network(s) independently. (See U.S. Pat. No. 5,371,852 to Attanasio et al. issued on Dec. 6, 1994 which is herein incorporated by reference in its entirety.) At the destination computer, arriving packets are stored in a buffer and are then transformed back into the form which is close to the original analog signal. The same industry standard (e.g., PCM, GSM, G.723, etc.) defines this transformation.
Some of the prior art has disclosed duplicating messages and transmitting them over “multiple disjointed routes” over a network topology to improve reliability and timely delivery of these messages. See “Delivery of Time-Critical Messages Using a Multiple Copy Approach” by P. Ramanathan and K. G. Shin, ACM fransactions on Computer Systems, Vol. 10, No. 2, May 1992, (here after the “Shin reference”) which is herein incorporated by reference in its entirety.
STATEMENT OF PROBLEMS WITH THE PRIOR ART
Quality is a serious problem in sending media over packet switching networks, including Internet and Intranet. This problem comes from the two general characteristics of packet switching networks, namely: (A) packet switching networks cannot guarantee the delivery of packets, e.g., a packet can be lost on the way to the destination and (B) packet switching networks cannot guarantee the delivery of packets within given time, e.g., if the network is congested, packets are delayed inside the network.
These two characteristics come from the fact that packet switching networks comprise commonly used routers and links connecting them. Since these resources are shared by many packets, waiting queues for these resources are built into the network. When the network is congested, packets are forced to wait in these queues. When traffic volume exceeds the capacity of these queues, packets can be discarded. Due to these characteristics of packet switching networks, packet delays and losses are unavoidable for packet transmission over packet switching networks.
TCP (Transmission Control Protocol) remedies some of these shortcomings of packet switching networks by introducing a packet re-transmission mechanism outside of the network between source and destination computers. It arranges a buffer to store received packets internally. If some packets do not arrive in a given time, re-transmission of these packets is requested. Until all packets are received, with potentially multiple retries, the received data will not be released to the receiving application. Thus, TCP guarantees that all packets arrive but sacrifices transmission time, i.e., there are delays. Because of the delay caused by this automatic re-transmission, TCP protocol is not used for internet media transmission where delay is fatal.
For two-way Internet media transmission, long delays are fatal. While accepting some packets being lost, Internet media usually uses a protocol without built-in packet re-transmission (e.g., UDP or User Datagram Protocol). Even with this protocol, however, some packets may be lost in the network and there is no guarantee of a minimum time for transmission over the network (without delays). Usually, the upper layer application software controls the size of a waiting buffer and the maximum waiting time for packet arrivals.
One prior art system is described in “The 2nd Annual Internet Telephony, Summit” of Jul. 14-15, 1997, which is herein incorporated by reference. In particular, the “Motorola IP Telephony in Corporate Intranets” describes one instance of packet delay and losses in Internet media transmission. Here a series of packets were sent for a certain time from a source computer to a destination computer over the Internet. The article shows a graph of the arrival delay for each packet. In the graph, the X-axis corresponds to the packets from the first to the last. The Y-axis shows the time required for the packet to travel to the destination. The unit of the Y-axis is milliseconds. The graph shows that using the prior art Internet network to transmit Internet media is not as reliable as using the telephone network.
Some prior art literature compares the current state of the art of Internet/Intranet transmissions to transmissions over the telephone network as the follows. Compared to the telephone network that provides acceptable voice over 99.999% of use time, the Internet provides acceptable voice only in 94% of use time and the Internet provides acceptable voice only of 61% of use time. (See the article by Tom Nolle: President of CIMI Corp., entitled “Convergence 2000?” published in “Information Week”, Apr. 15, 1998, page 141, which is herein incorporated by reference in its entirety.)
An example of a typical prior art networking system
100
for transmitting media information, including voice data, is shown as a block diagram in FIG.
1
. The networking system
100
comprises a plurality of computers
160
that are connected to one or more networks
130
through well known network connectors such as modems and/or LAN adapters
150
. The computers
160
typically can be any generally known computer system, such as a personal computer (like an IBM ThinkPad) or workstation (like an IBM AS400). For one way communications, one computer
160
would be the source computer
160
S originating the transmission of information and one or more of the computers
160
would be the destination computer
160
D that would receive the information. However, in many applications, both the source computer
160
S and the destination computer
160
D functions are contained in a single computer, e.g.
160
, that can perform both these communication functions, i.e., sending and receiving, to enable point to point two way, one to many, and/or many to many communications. The computers
160
will have well known input and output devices like microphones
131
, speakers
132
, keyboards, mice, cameras, video recorders, screens, recorders, musical instruments, pen inputs, touch screens (not shown), etc. The combination of one or more multimedia interfaces
133
, e.g. a sound card and/or video card
133
,
Fujisaki Tetsunosuke
Medan Yoav
Chin Wellington
International Business Machines - Corporation
Percello, Esq. Louis J.
Phan M.
Ryan & Mason & Lewis, LLP
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