System for real time communication buffer management

Details System for real time communication buffer management System for real time communication buffer management

1999-05-28

2002-08-13

Coulter, Kenneth R. (Department: 2154)

Electrical computers and digital processing systems: multicomput

Multicomputer data transferring via shared memory

Plural shared memories

C709S231000, C711S148000

Reexamination Certificate

active

06434606

ABSTRACT:

REFERENCE TO COMPUTER PROGRAM LISTING APPENDICES SUBMITTED ON COMPACT DISC
The originally filed specification for the present application included Appendices A-C, which contained paper printouts of several computer program listings and an output file. Two compacts discs containing electronic text copies of the computer program listings and output file of Appendices A-C have been submitted for the present application. These electronic copies of Appendices A-C have been labeled with the appropriate identification for this application, and one of the compact discs has been labeled “Copy 1,” while the other has been labeled “Copy 2.” The compact disc labeled “Copy 2” is identical to the one labeled “Copy 1,” and both compact discs are specifically incorporated herein by reference.
Each of the submitted compact discs is formatted for a PC type workstation with an MS-Windows based operating system, and includes the serial label number of 011129
—
1352. The following is a list of the folders and files on each of the two submitted compact discs:
Folder—Appendix A
File—buffer_mgmt.cc.txt (Size: 9 KB; Dated: Nov. 29, 2001)
Folder—Appendix B
File—VoIP Output File.txt (Size: 7 KB; Dated: Nov. 29, 2001)
Folder—Appendix C
File—buffer.h.txt (Size: 7KB; Dated: Nov. 29, 2001)
File—voicebuffer.cc.txt (Size: 13 KB; Dated: Nov. 29, 2001)
File—voicebuffer.h.txt (Size: 3 KB; Dated: Nov. 29, 2001)
COPYRIGHT NOTICE AND AUTHORIZATION
A portion of the disclosure of this patent document contains material that is subject to copyright protection. The copyright owner has no objection to the facsimile reproduction by anyone of the patent document or the patent disclosure, as it appears in the Patent and Trademark Office patent file or records, but otherwise reserves all copyright rights whatsoever.
BACKGROUND OF THE INVENTION
A. Field of the Invention
This invention relates to the field of telecommunications and more specifically to a method and apparatus for choosing buffer size and error correction coding for real time communication over packet networks.
B. Description of Related Art and Advantages of the Invention
Real time communications such as audio or video can be encoded using various compression techniques. The encoded information can then be placed in data packets with time and sequence information and transported via non-guaranteed Quality of Service (QoS) packet networks. Non-guaranteed packet switched networks include a Local Area Network (LAN), Internet Protocol Network, frame relay network, or an interconnected mixture of such networks such as an Internet or Intranet. One underlying problem with non-guaranteed packet networks is that transported packets are subject to varying loss and delays. Therefore, for real-time communications, a tradeoff exists among the quality of the service, the interactive delay, and the utilized bandwidth. This tradeoff is a function of the selected coding scheme, the packetization scheme, the redundancy of information packeted within the packets, the receiver buffer size, the bandwidth restrictions, and the transporting characteristics of the transporting network.
One technique for transporting real time communication between two parties over a packet switched network requires that both parties have access to multimedia computers. These computers must be coupled to the transporting network. The transporting network could be an Intranet, an Internet, a wide area network (WAN), a local area network (LAN), or other type of network utilizing technologies such as Asynchronous Transfer Mode (ATM), Frame Relay, Carrier Sense Multiple Access, Token Ring, or the like. As in the case for home personal computers (PCs), both parties to the communication may be connected to the network via telephone lines. These telephone lines are in communication with a local hub associated with a central office switch and a Network Service provider. As used herein, the term “hub” refers to an access point of a communication infrastructure.
This communication technique however, has a number of disadvantages. For example, for a home-based PC connected to a network using an analog telephone line, the maximum bandwidth available depends on the condition of the line. Typically, this bandwidth will be no greater than approximately 3400 Hz. A known method for transmitting and receiving data at rates of up to 33.6 kbits/second over such a connection is described in Recommendation V.34, published by the International Telecommunication Union, Geneva, Switzerland.
Aside from a limited bandwidth, various delays inherent in the PC solution, such as sound card delays, modem delays, and other related delays are relatively high. Consequently, the PC-based communication technique is generally unattractive for real-time communication. As used herein, “real-time communication” refers to real-time audio, video, or a combination of the two.
Another typical disadvantage of PC-based communication, particularly with respect to PC-based telephone communications, is that the communicating PC receiving the call generally needs to be running at the time the call is received. This may be feasible for a corporate PC connected to an Intranet. However, such a connection may be burdensome for a home based PC, since the home PC may have to tie up a phone line.
Another disadvantage is that a PC-based conversation is similar to conversing over a speakerphone. Hence, privacy of conversation may be lost. Communicating over a speakerphone may also present problems in a typical office environment having high ambient noise or having close working arrangements.
In addition, PC-based telephone systems often require powerful and complex voice encoders and therefore require a large amount of processing capability. Even if these powerful voice encoders run on a particularly powerful PC, the encoders may slow down the PC to a point where the advantage of document sharing decreases, since the remaining processing power may be insufficient for a reasonable interactive conversation. Consequently, a caller may have to use less sophisticated encoders, thereby degrading the quality of the call.
A general problem encountered in packet switched networks, however, is that the network may drop or lose data packets. Packets may also be delayed during transportation from the sender to the receiver. Therefore, some of the packets at a receiving destination will be missing and others will be “jittered” and therefore arrive out of order.
In a packet switched network whose transporting characteristics vary relatively slowly, the immediate past transporting characteristics can be used to infer information about the immediate future transporting characteristics. The dynamic network transporting characteristics may be measured using such variables as packet loss, packet delay, packet burst loss, loss auto-correlation, bandwidth, and delay variation.
IP gateways, such as IP telephony receivers, may employ a configuration of computational buffers or jitter buffers to mask network-induced expansion and contraction of packet inter-arrival times. Although IP telephony transmitters may send packets with deterministic inter-departure times, IP networks such as the Internet will “jitter” (i.e., introduce delay variance) and lose packets as the packets are transported through some number of switches and routers before the packets arrive at the IP gateway, such as the IP telephony receiver. The greater the jitter buffer depth, the more jitter that the communication channel can mask.
If packet arrivals are highly skewed with respect to buffer depth, packets may be lost due to buffer overflow or buffer underflow. However, due to the interactive nature of real time communication over IP, particularly IP telephony, it is desirable to introduce as little jitter buffer latency as possible. Therefore, a buffer having a shallow depth is generally desired. IP telephony end-user quality of service is also degraded by packet loss introduced by the IP network itself. For example, an intermediate IP router in between the source and destination of the real-time communication ma

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