Methods and computer executable instructions for improving...

Multiplex communications – Communication techniques for information carried in plural... – Combining or distributing information via time channels

Reexamination Certificate

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Details

C370S216000

Reexamination Certificate

active

06212206

ABSTRACT:

BACKGROUND OF THE INVENTION
1. The Field of the Invention
The present invention relates generally to packet switching networks. More specifically, the present invention relates to methods and computer executable instructions for improving communications between two or more terminal locations in a packet switching network. Even more specifically, the present invention relates to improving communications occurring contemporaneously between terminal locations such as occurs during a video conference.
2. The Relevant Technology
In general, a computer communications network is an infrastructure of switching nodes, access lines and trunks that interconnect a variety of terminal locations to one another for the purposes of allowing communications between the end users at the terminal locations. Frequently, these communications are contemporaneous, such as in a video conference, but many times are of entirely different time intervals, such as with electronic mail. The communications can assume a variety of forms. Some examples include the conveyance or exchange of information such as voice, text, data, graphics, video and documents.
Although many systems and methods are available that allow this communication between terminal locations of a network, three primary switching paradigms exist. Circuit switching is a method that physically connects the end users at the terminal locations before communication takes place. This is analogous to an ordinary phone call and a dedicated transmission path is often allocated for the duration of the call. Message switching exchanges messages between end users in a “store-and-forward” manner analogous to an ordinary postal system. Unlike circuit switching, method switching does not require the destination terminal location to be active at the same time that the originating terminal location is active. Packet switching is a method that exchanges packets, i.e., fragments of messages, between the terminal locations that are sequenced together at the destination location to form a complete message. With respect to video conferencing communications, packet switching is the typical paradigm.
In many packet switching networks (PSN's), the path traversed by a packet through the network is established during the call set-up procedures. This is commonly referred to as a virtual circuit PSN. In other PSN's, however, a service typically allows users to transmit individually addressed packets without the need for a call set-up. This is commonly referred to as a datagram PSN.
In either virtual circuit or datagram PSN's, numerous terminal locations are simultaneously sending packets to various other terminal locations throughout the network. In sending these packets, each packet is progressed through the network in its own distinctive route or transmission path. In general, this transmission path is established as follows: (i) a packet arrives at a switching node at some random time; (ii) the switching node which received the packet determines the next node in the transmission path to which the packet will be sent; and (iii) the switching node places the packet in a queue for transmission to that next node over some network trunk. As between switching nodes, however, one trunk may have a higher or lower transmission capacity as compared to another trunk. Thus, packets are known to arrive at the destination terminal location in a sporadic manner. This is a phenomenon known commonly as “latency.”
In general, network latency is variable. Depending upon how many or how few factors contribute to the latency, the latency can range from a few fractions of a second with in-house networks to as many as 10 seconds, or more, with heavily bottle-necked networks.
Other factors also contribute to network latency. Such other factors include, but are not limited to: network volume causing congestion during peak hours; re-routing of packets when switching nodes become incapacitated; individual nodal processing capabilities; queuing delays caused by nodal software processes; protocol incompatibility between terminal locations and the network; performance capabilities of the terminal location and other similarly related factors.
However, with contemporaneous style communications between end users, such as video conferencing, latency can be extremely detrimental. For example, in a typical video conference, video and audio packets are continuously exchanged back-and-forth between two or more end users. Upon their arrival at the destination terminal location, the packets are stored in buffers until played on appropriate video and audio equipment.
If the latency is too long and a packet is caused to be delayed for more than about three-tenths of a second before being played, the video and/or audio communication from the originating terminal location will appear stilted to the end user at the destination terminal location. This frequently causes hesitation and/or “cross-talk” communications between both end users. As a result, the advantages of communicating by video conference will be diminished because the end users at both terminal locations must behave in an unnatural speaking manner to overcome this shortcoming.
Latency, in combination with the buffer size at the destination terminal location, is also problematic. For instance, when packets arrive, multiple packets are stored together in a singular buffer. If the latency is too long, some packet content may be omitted from inclusion within the buffer. Ultimately, the audio or video packet contents will not be played at the destination terminal location and the communication will appear incomplete. Again, hesitation and/or cross-talk can result.
As such, some prior art video conferencing devices attempt to control the buffer size. In these devices, the buffer size is set by the end user(s) during initialization processes such as the conference call set-up. Often, the end user(s) are unaware of the implications of buffer size and choose a buffer size in a haphazard manner. This leads to inaccurate buffer size selections. Moreover, in most of these devices, once the buffer size is selected, adjustments cannot be made. If the latency is too long, the end users must suffer through communications having omitted audio and/or video.
One prior art device, however, does provide for a change in buffer size. Yet, with this device, the buffer size may only be increased. Also, this increase is essentially instantaneous and causes either an audible or visual jerk, or both.
Accordingly, it is desirous to overcome the problems of latency experienced in PSN's. Ultimately, this will improve communications between terminal locations therein, especially for contemporaneous style communications like video conferencing.
OBJECTS AND SUMMARY OF THE INVENTION
It is, therefore, an object of the present invention to provide improved communications between terminal locations in a packet switched network.
It is another object of the present invention to provide software and methods that overcome the effects of latency in packet switched networks.
It is a further object of the present invention to provide software and methods that essentially eliminate end user intervention in establishing buffer sizes for the reception of audio and/or video packets during a video conference in a packet switched network.
It is yet another object of the present invention to provide software and methods that essentially eliminate any audio or visual jerks during a video conference in a packet switched network.
It is still yet another object of the present invention to provide software and methods that provide smooth audio and visual transitions during a video conference while corrective actions overcoming effects of latency are being implemented.
It is an even further object of the present invention to provide software and methods that allow for decreases in the sizes of buffers that receive audio and/or video packets during a video conference in a packet switched network.
In accordance with the invention as embodied and broadly described herein, the foregoing a

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