[US Patent & Trademark Office, Patent Full Text and Image Database]
[Home] [Boolean Search] [Manual Search] [Number Search] [Help]
[Bottom]
[View Shopp ing Cart] [Add to Shopping Cart]
[Image]
  ( 1 of 1 )
United States Patent 7,082,106
Sharma ,   et al. July 25, 2006
**Please see images for: ( Certificate of Correction ) **
Computer-based multi-media communications system and method

Abstract

Outgoing multi-media data components are provided in a communication system and method implemented under control of a user interface program. Each of the outgoing multi-media data components includes one or more types of information (e.g., textual information, graphical information, and audio information). Two or more outgoing multi-media data components are combined under control of a local user via input to the user interface program resulting in combined outgoing multi-media mail. At least one of the two or more outgoing multi-media data components is of a type of information different than another of the two or more outgoing multi-media data components. The combined outgoing multi-media mail is provided for communication to a remote location.

Inventors: Sharma; Raghu (North Oaks, MN), Davis; Jeffrey P. (Ham Lake, MN), Maitra; Sidhartha (Saratoga, CA), Gunn; Timothy D. (Mounds View, MN)
Assignee: Multi-Tech Systems, Inc. (Moundsview, MN)
Family ID: 46277279
Appl. No.: 09/768,748
Filed: January 24, 2001
Prior Publication Data
Document IdentifierPublication Date
US 20020075815 A1Jun 20, 2002
Related U.S. Patent Documents
Application NumberFiling DatePatent NumberIssue Date
09400607Sep 20, 1999
08338340Nov 10, 19946009082
08002467Jan 8, 19935452289

Current U.S. Class: 370/276; 370/352
Current CPC Class: H04L 12/6418 (20130101); H04L 27/3416 (20130101); H04L 29/06 (20130101); H04M 1/2478 (20130101); H04M 1/253 (20130101); H04M 1/6033 (20130101); H04M 1/663 (20130101); H04M 3/436 (20130101); H04M 3/5307 (20130101); H04M 3/533 (20130101); H04M 3/567 (20130101); H04M 11/06 (20130101); H04M 11/068 (20130101); H04N 1/00204 (20130101); H04N 1/00206 (20130101); H04L 69/04 (20130101); H04M 7/0033 (20130101); H04M 7/0036 (20130101); H04M 7/128 (20130101); H04M 7/1295 (20130101); G10L 19/04 (20130101); H04L 29/06027 (20130101); H04L 2012/6424 (20130101); H04L 2012/6427 (20130101); H04L 2012/6459 (20130101); H04L 2012/6475 (20130101); H04L 2012/6481 (20130101); H04L 2012/6491 (20130101); H04L 2012/6494 (20130101); H04M 1/57 (20130101); H04M 3/002 (20130101); H04M 7/006 (20130101); H04M 2203/4509 (20130101)
Current International Class: H04B 1/56 (20060101)
Field of Search: ;370/270,204,280,281,295,296,405,535,537,477,476,435,468,351,352 ;375/345,222,265 ;380/23 ;341/143,61 ;716/17 ;379/142.15,395,88.1,88.19,43.01,406.13,387.01,68,345,455,434,93.02,93.06,93.05,93.08,93.15,93.17,93.23,100,13,118,127,128,126,130

References Cited [Referenced By]
U.S. Patent Documents
3304372 February 1967 Filipowsky et al.
3789165 January 1974 Campanella et al.
3904830 September 1975 Every, Sr. et al.
3973081 August 1976 Hutchins
3973089 August 1976 Puckette
3997732 December 1976 Every, Sr. et al.
4040014 August 1977 Gehrking
4045774 August 1977 Morrison
4059731 November 1977 Green et al.
4059800 November 1977 Jones, Jr.
4074081 February 1978 Humm
4100377 July 1978 Flanagan
4107471 August 1978 Reed
4124773 November 1978 Elkins
4178480 December 1979 Carbrey
4205201 May 1980 Kahn
4205202 May 1980 Kahn
4216354 August 1980 Esteban et al.
4284850 August 1981 Clingelpeel
4310721 January 1982 Manley et al.
4320265 March 1982 Biggiogera et al.
4354273 October 1982 Araseki et al.
4363122 December 1982 Black et al.
4377860 March 1983 Godbole
4403322 September 1983 Kato
4425512 January 1984 Tomooka et al.
4425625 January 1984 Seligman et al.
4425661 January 1984 Moses et al.
4445213 April 1984 Baugh et al.
4447675 May 1984 Arntsen et al.
4450554 May 1984 Steensma et al.
4455649 June 1984 Esteban et al.
4476559 October 1984 Brolin et al.
4479195 October 1984 Herr et al.
4479213 October 1984 Galand et al.
4491945 January 1985 Turner
4494230 January 1985 Turner
4495620 January 1985 Steele et al.
4500987 February 1985 Hasegawa
4503288 March 1985 Kessler
4503533 March 1985 Tobagi et al.
4523055 June 1985 Hohl et al.
4524244 June 1985 Faggin et al.
4528425 July 1985 Melindo et al.
4528659 July 1985 Jones, Jr.
4531023 July 1985 Levine
4532495 July 1985 Gagnon
4533948 August 1985 McNamara et al.
4534024 August 1985 Maxemchuk et al.
4535448 August 1985 Baxter et al.
4546212 October 1985 Crowder, Sr.
4549290 October 1985 Bell
4549308 October 1985 LoPinto
4555599 November 1985 Hackett-Jones et al.
4555805 November 1985 Talbot
4578537 March 1986 Faggin et al.
4581735 April 1986 Flamm et al.
4587651 May 1986 Nelson et al.
4589107 May 1986 Middleton et al.
4593389 June 1986 Wurzburg et al.
4597101 June 1986 Kishimoto et al.
4598397 July 1986 Nelson et al.
4609788 September 1986 Miller et al.
4610022 September 1986 Kitayama et al.
4611323 September 1986 Hessenmiller
4611342 September 1986 Miller et al.
4622680 November 1986 Zinser
4625081 November 1986 Lotito et al.
4629829 December 1986 Puhl et al.
4630262 December 1986 Callens et al.
4631746 December 1986 Bergeron et al.
4644561 February 1987 Paneth et al.
4652703 March 1987 Lu et al.
4653086 March 1987 Laube
4660194 April 1987 Larson et al.
4660218 April 1987 Hashimoto
4665514 May 1987 Ching et al.
4670874 June 1987 Sato et al.
4679190 July 1987 Dias et al.
4697281 September 1987 O'Sullivan
4698803 October 1987 Haselton et al.
4700341 October 1987 Huang
4707831 November 1987 Weir et al.
4716585 December 1987 Tompkins et al.
4718057 January 1988 Venkitakrishnan et al.
4718082 January 1988 Parker et al.
4723238 February 1988 Isreal et al.
4725977 February 1988 Izumi et al.
4730311 March 1988 Carse et al.
4740963 April 1988 Eckley
4750169 June 1988 Carse et al.
4751510 June 1988 Desaint Michel et al.
4751736 June 1988 Gupta et al.
4757497 July 1988 Beierle et al.
4757527 July 1988 Beniston et al.
4761646 August 1988 Choquet et al.
4764955 August 1988 Galand et al.
4769833 September 1988 Farleigh et al.
4771425 September 1988 Baran et al.
4782485 November 1988 Gollub
4782515 November 1988 Phillips et al.
4785472 November 1988 Shapiro
4792945 December 1988 Mark
4794595 December 1988 Ohyama
4797878 January 1989 Armstrong
4802221 January 1989 Jibbe
4807250 February 1989 Tanaka
4809271 February 1989 Kondo et al.
4813040 March 1989 Futato
4817089 March 1989 Paneth et al.
4819228 April 1989 Baran et al.
4821312 April 1989 Horton et al.
4827085 May 1989 Yaniv et al.
4835765 May 1989 Bergmans et al.
4839802 June 1989 Wonak et al.
4839906 June 1989 Leveque et al.
4845746 July 1989 Li
4847677 July 1989 Music et al.
4847829 July 1989 Tompkins et al.
4847900 July 1989 Wakim
4862449 August 1989 Hoefkens et al.
4864559 September 1989 Perlman
4866704 September 1989 Bergman
4866732 September 1989 Carey et al.
4872160 October 1989 Hemmady et al.
4873715 October 1989 Shibata
4887265 December 1989 Felix
4890282 December 1989 Lambert et al.
4890316 December 1989 Walsh et al.
4890326 December 1989 Euler
4890327 December 1989 Bertrand et al.
4891806 January 1990 Farias et al.
4893308 January 1990 Wilson et al.
4893326 January 1990 Duran et al.
4893339 January 1990 Bright et al.
4901333 February 1990 Hodgkiss
4903261 February 1990 Baran et al.
4905231 February 1990 Leung et al.
4905282 February 1990 Mcglynn et al.
4907277 March 1990 Callens et al.
4912705 March 1990 Paneth et al.
4912745 March 1990 Fitzpatrick et al.
4912756 March 1990 Hop
4912758 March 1990 Arbel
4914586 April 1990 Swinehart et al.
4914650 April 1990 Sriram
4926448 May 1990 Kraul et al.
4926482 May 1990 Frost et al.
4932022 June 1990 Keeney et al.
4932048 June 1990 Kenmochi et al.
4935927 June 1990 Kaewell, Jr. et al.
4935954 June 1990 Thompson et al.
4942569 July 1990 Maeno
4953210 August 1990 Mcglynn et al.
4965789 October 1990 Bottau et al.
4972457 November 1990 O'Sullivan
4972462 November 1990 Shibata
4972483 November 1990 Carey
4974099 November 1990 Lin et al.
4977591 December 1990 Chen et al.
4981371 January 1991 Gurak et al.
4985891 January 1991 Fujiwara et al.
4989203 January 1991 Phinney
4991169 February 1991 Davis et al.
4991172 February 1991 Cidon et al.
4991197 February 1991 Morris
4994802 February 1991 Critchlow et al.
4995059 February 1991 Ishikawa
4996697 February 1991 Critchlow et al.
4998241 March 1991 Brox et al.
4998248 March 1991 Matsuzaki
5001710 March 1991 Gawrys et al.
5001729 March 1991 Tjahjadi et al.
5001745 March 1991 Pollock
5005183 April 1991 Carey et al.
5007050 April 1991 Kasparian et al.
5008901 April 1991 Wallach et al.
5008926 April 1991 Misholi
5014232 May 1991 Andre
5014267 May 1991 Tompkins et al.
5014306 May 1991 Rodgers et al.
5020058 May 1991 Holden et al.
5022024 June 1991 Paneth et al.
5023903 June 1991 Bowen
5025443 June 1991 Gupta
5034962 July 1991 Yamamoto et al.
5036513 July 1991 Greenblatt
5042028 August 1991 Ogawa
5044010 August 1991 Frenkiel et al.
5046188 September 1991 Molnar
5051720 September 1991 Kittirutsunetorn
5054055 October 1991 Hanle et al.
5062133 October 1991 Melrose
5065395 November 1991 Shenoi et al.
5065425 November 1991 Lecomte et al.
5065427 November 1991 Godbole
5070526 December 1991 Richmond et al.
5072308 December 1991 Lin et al.
5073926 December 1991 Suzuki et al.
5079769 January 1992 Ehricke et al.
5081647 January 1992 Bremer
5083310 January 1992 Drory
5086425 February 1992 Le Goffic et al.
5086471 February 1992 Tanaka et al.
5099472 March 1992 Townsend et al.
5107519 April 1992 Ishikawa
5115429 May 1992 Hluchyj et al.
5119375 June 1992 Paneth et al.
5121385 June 1992 Tominaga et al.
5121391 June 1992 Paneth et al.
5127001 June 1992 Steagall et al.
5127041 June 1992 O'Sullivan
5130985 July 1992 Kondo et al.
5132966 July 1992 Hayano et al.
RE34034 August 1992 O'Sullivan
5136586 August 1992 Greenblatt
5138662 August 1992 Amano et al.
5146470 September 1992 Fujii et al.
5150357 September 1992 Hopner et al.
5150410 September 1992 Bertrand
5151937 September 1992 Chujo et al.
5153897 October 1992 Sumiyoshi et al.
5155726 October 1992 Spinney et al.
5155847 October 1992 Kirouac et al.
5162812 November 1992 Aman et al.
5164982 November 1992 Davis
5170427 December 1992 Guichard et al.
5177734 January 1993 Cummiskey et al.
5179588 January 1993 Nowicki
5182762 January 1993 Shirai et al.
5187591 February 1993 Guy et al.
5187692 February 1993 Haneda et al.
5193110 March 1993 Jones et al.
5195086 March 1993 Baumgartner et al.
5195130 March 1993 Weiss et al.
5206900 April 1993 Callele
5208812 May 1993 Dudek et al.
5208850 May 1993 Kino
5214650 May 1993 Renner et al.
5214656 May 1993 Chung et al.
5228026 July 1993 Albrow et al.
5228033 July 1993 Okouchi
5228076 July 1993 Hopner et al.
5233660 August 1993 Chen
5235595 August 1993 O'Dowd
5247591 September 1993 Baran
5249218 September 1993 Sainton
5257410 October 1993 Meszko et al.
5258983 November 1993 Lane et al.
5260783 November 1993 Dixit
5261027 November 1993 Taniguschi et al.
5263019 November 1993 Chu
5272695 December 1993 Makino et al.
5272700 December 1993 Hansen et al.
5276703 January 1994 Budin et al.
5278900 January 1994 Van Gerwen et al.
5280519 January 1994 Nakajima et al.
5282197 January 1994 Kreitzer
5283638 February 1994 Engberg et al.
5283819 February 1994 Glick et al.
5289539 February 1994 Maruyama
5295136 March 1994 Ashley et al.
5297203 March 1994 Rose et al.
5301246 April 1994 Archibald et al.
5305312 April 1994 Fornek et al.
5307413 April 1994 Denzer
5309434 May 1994 Maekawa
5309562 May 1994 Li
5313498 May 1994 Sano
5317604 May 1994 Osterweil
5319682 June 1994 Clark
5327520 July 1994 Chen
5329472 July 1994 Sugiyama
5341374 August 1994 Lewen et al.
5343473 August 1994 Cidon et al.
5343521 August 1994 Jullien et al.
5347572 September 1994 Avni
5355365 October 1994 Bhat et al.
5355406 October 1994 Chencinski et al.
5365576 November 1994 Tsumura et al.
5365577 November 1994 Davis et al.
5369700 November 1994 Koura et al.
5371853 December 1994 Kao et al.
5375068 December 1994 Palmer et al.
5377260 December 1994 Long
5379340 January 1995 Overend et al.
5381412 January 1995 Otani
5384780 January 1995 Lomp et al.
H1413 February 1995 Gunn
5388150 February 1995 Schneyer et al.
5390239 February 1995 Morris et al.
5390250 February 1995 Janse et al.
5402474 March 1995 Miller et al.
5406557 April 1995 Baudoin
5406560 April 1995 Kondo et al.
5408401 April 1995 Miyazaki
5408522 April 1995 Ikehata et al.
5408527 April 1995 Tsutsu
5414773 May 1995 Handelman
5414796 May 1995 Jacobs et al.
5416776 May 1995 Panzarella et al.
5425089 June 1995 Chan et al.
5426652 June 1995 Heiman
5438536 August 1995 Salzman
5438614 August 1995 Rozman et al.
5440547 August 1995 Easki et al.
5440619 August 1995 Cann
5440698 August 1995 Sindhu et al.
5444770 August 1995 Davis et al.
5448555 September 1995 Bremer et al.
5450425 September 1995 Gunn et al.
5452289 September 1995 Sharma et al.
5452329 September 1995 Sato
5453986 September 1995 Davis et al.
5463616 October 1995 Kruse et al.
RE35104 November 1995 Murakami et al.
5465267 November 1995 Todoroki
5471470 November 1995 Sharma et al.
5472351 December 1995 Greco et al.
5473676 December 1995 Frick et al.
5477547 December 1995 Sugiyama
5479407 December 1995 Ko et al.
5479475 December 1995 Grob et al.
5483576 January 1996 Staples
5490060 February 1996 Malec et al.
5493609 February 1996 Winseck, Jr. et al.
5500859 March 1996 Sharma et al.
5502727 March 1996 Catanzaro et al.
5515398 May 1996 Walsh et al.
5521975 May 1996 Hashimoto
5524047 June 1996 Brown et al.
5526423 June 1996 Ohuchi et al.
5535204 July 1996 Li
5541640 July 1996 Larson
5546395 August 1996 Sharma et al.
5546448 August 1996 Caswell et al.
5550816 August 1996 Hardwick et al.
5559793 September 1996 Maitra et al.
5574725 November 1996 Sharma et al.
5577041 November 1996 Sharma et al.
5592586 January 1997 Maitra et al.
5596628 January 1997 Klein
5600649 February 1997 Sharma et al.
5617423 April 1997 Li et al.
5619508 April 1997 Davis et al.
5628030 May 1997 Tuckner
5631745 May 1997 Wong et al.
5644629 July 1997 Chow
5673257 September 1997 Sharma et al.
5673268 September 1997 Sharma et al.
5675640 October 1997 Tappert et al.
5699410 December 1997 Imamura et al.
5699417 December 1997 Khojasteh
5699418 December 1997 Jones
5717742 February 1998 Hyde-Thomson
5745558 April 1998 Richardson, Jr. et al.
5754589 May 1998 Maitra et al.
5764627 June 1998 Sharma et al.
5764628 June 1998 Davis et al.
5790532 August 1998 Sharma et al.
5812534 September 1998 Davis et al.
5815503 September 1998 Li
5821987 October 1998 Larson
5864560 January 1999 Li et al.
6009082 December 1999 Caswell et al.
6012113 January 2000 Tuckner
6181784 January 2001 Duran et al.
Foreign Patent Documents
2104701 Jul 1994 CA
3344140 Jun 1984 DE
3426055 Jan 1985 DE
3409532 Sep 1985 DE
3504064 Aug 1986 DE
3726366 Feb 1988 DE
3630469 Mar 1988 DE
3531173 Mar 1989 DE
3490532 Apr 1989 DE
0 309 993 Apr 1989 EP
0 309 993 Apr 1989 EP
0 364 866 Jan 1990 EP
0 374 828 Jun 1990 EP
0 429 054 May 1991 EP
0 443 548 Aug 1991 EP
0 453 128 Oct 1991 EP
0 480 634 Apr 1992 EP
0 488 685 Jun 1992 EP
0 488 685 Jun 1992 EP
0 488 685 Jun 1992 EP
0 488 865 Jun 1992 EP
0 496 427 Jul 1992 EP
0 496 427 Jul 1992 EP
0 510 411 Oct 1992 EP
0 526 104 Feb 1993 EP
0 581 528 Feb 1994 EP
0 582 537 Feb 1994 EP
0 614 305 Sep 1994 EP
0 630 141 Dec 1994 EP
0 650 286 Apr 1995 EP
0 425 163 May 1995 EP
2 025 735 Jan 1980 GB
2 131 654 Jun 1984 GB
2 173 070 Oct 1986 GB
2 184 327 Jun 1987 GB
2 195 866 Apr 1988 GB
2 208 986 Apr 1989 GB
2 210 237 Jun 1989 GB
2 212 028 Jul 1989 GB
2 227 625 Aug 1990 GB
2 260 670 Apr 1993 GB
2 268 663 Jan 1994 GB
58-88961 May 1983 JP
59-44140 Mar 1984 JP
59-081947 May 1984 JP
61077449 Apr 1986 JP
62-61437 Mar 1987 JP
62-77730 Apr 1987 JP
62-221267 Sep 1987 JP
63-019951 Jan 1988 JP
63-48030 Feb 1988 JP
63-54052 Mar 1988 JP
63-64460 Mar 1988 JP
63-105533 May 1988 JP
63-124658 May 1988 JP
63-155831 Jun 1988 JP
63-169850 Jul 1988 JP
61054052 Aug 1988 JP
64-16043 Jan 1989 JP
64-18329 Jan 1989 JP
01068052 Mar 1989 JP
1-181344 Jul 1989 JP
1-198144 Aug 1989 JP
02034077 Feb 1990 JP
2-67847 Mar 1990 JP
2-107027 Apr 1990 JP
2-145097 Jun 1990 JP
193489 Jul 1990 JP
257748 Oct 1990 JP
03009643 Jan 1991 JP
3-58582 Mar 1991 JP
3-157027 Jul 1991 JP
03153152 Jul 1991 JP
03162052 Jul 1991 JP
03237853 Oct 1991 JP
U 3-115458 Nov 1991 JP
03280797 Dec 1991 JP
4-35137 Feb 1992 JP
04319847 Nov 1992 JP
5-504687 Jul 1993 JP
2790977 Jul 1995 JP
WO 87/01254 Feb 1987 WO
WO 91/05419 Apr 1991 WO
WO 91/07044 May 1991 WO
WO 91/15067 Oct 1991 WO
WO 92/06550 Apr 1992 WO
WO 92/20028 Nov 1992 WO
WO 93/11643 Jun 1993 WO
WO 93/22869 Nov 1993 WO
WO 94/26056 Nov 1994 WO
WO 96/15601 May 1996 WO
WO 96/15612 May 1996 WO

Other References

CCITT--The International Telegraph and Telephone Consultative Committee--Blue Book. vol. VIII--Fascicle VIII.1--Rec. V42 on pp. 296-370, "Data Communication Over the Telephone Network" Series V Recommendations. IX.sup.th Plenary Assembly, Melbourne, Nov. 14-25, 1988. Geneva 1989. ISBN 92-61-03661-9. cited by other .
CCITT--The International Telegraph and Telephone Consultative Committee--Blue Book. vol. VIII--Fascicle VIII.2, Data Communication Networks: Services and Facilities, Recommendations X.1-X.32, IX.sup.th Plenary Assembly, Melbourne, Nov. 14-25, 1988. Geneva 1989. ISBN 92-61-03671-9. Entire Book. cited by other .
Data Sheet. "High Level Data Link Control (HDLC) and Derivatives" pp. 1-5, [retrieved on Apr. 29, 2004] Retrieved from the Internet http://www.ricksiltshire.com/hdlc.htm. cited by other .
Data Sheet. Data Networks Resource. "SDLC, HDLC and LLC" pp. 1-9, [retrieved on Apr. 29, 2004] Retrieved from the Internet http://www.rhyshaden.com/hdlc.htm. cited by other .
Data Sheet. Telescada: CDPD Communications, "Cellular Digital Packet Data" Retrieved on Apr. 29, 2004. Retrieved from the Internet http://www.telescada.com/cdpd.htm. cited by other .
Data Sheet. Advanced Relay Corporation: Hardware and Software Products. "Physical Layer". 3 pages. Retrieved from the Internet on Apr. 29, 2004 (Google Search) http://www.advancedrelay.com/html/products/physical.html. cited by other .
"EIA/TIA Interim Standard--Cellular System Dual-Mode Mobile Station--Base Station Compatibility Standard". IS-54-B (Revision of IS/54-A) Apr. 1992. Telecommunications Industry Association. pp. 1-280. cited by other .
Hardwick, "The Data Link Layer", ISDN Design Professional and Technical Series. Title Page, Table of Contents, and pp. 46-48. cited by other .
ITU--International Telecommunication Union, Telecommunication Standardization Sector of ITU. Q.920 (Mar. 1993) Digital Subscriber Signalling System No. 1 (DSS1)--Digital Subscriber Signaling System No. 1 (DSS1)--ISDN User-Network Interface Data Link Layer--General Aspects. ITU-T Recommendation Q.920 (previously CCITT Recommendation). Title page, Table of Contents. pp. 1-20. cited by other .
ITU--International Telecommunication Union, Telecommunication Standardization Sector of ITU. Q.921 (Sep. 1997) "Series Q: Switching and Signalling--Digital subscriber Signaling System No. 1--Data link layer". ISDN User-Network Interface--Data Link Layer Specification. Title page, Table of Contents, pp. 1-256. cited by other .
Mobile/Cellular Technology. The Website for Mobile Communications. Diagram of IS-54B time slot structure. 1 page. Retrieved from the Internet on Apr. 27, 2004 http://www.mobilecomms-technology.com/projects/indiana/indiana5.html. cited by other .
MultiTech Systems. "The New MultiModem X.25, V.32TERBO X.25 Desktop Modem". 1 page. (1993). cited by other .
MultiTech Systems. Technical Specifications. 1 page (1993). cited by other .
MultiTech Systems. "The New MultiMuxX25, 8 and 16 port X.25 PADs for Private and Public Data Networks" 4 pages. (Feb. 1990). cited by other .
"Privateline.com:Mobile Telephone History", 4 pages. Retrieved from the Internet on Apr. 27, 2004 http://www.privateline.com/PCS/history10.htm. cited by other .
Tom Sheldon's Linktionary--Networking Defined and Hyperlinked. "LAP (Link Access Procedure". 2 pages. Retrieved from the Internet Apr. 29, 2004 (Google Search) http://www.linktionary.com/l/lap.html. cited by other .
Rahnema, "Overview of the GSM System and Protocol Architecture", IEEE Communications Magazine, Apr. 1993, 9 pages. cited by other .
Stallings, William . . . ISDN and Broadband ISDN, Second Edition. Copyright .COPYRGT. 1992 by MacMillan Publishing Company, a division of Macmillan, Inc. Entire Book. cited by other .
"X.25 Devices and Protocol Operation" Internetworking Technologies Handbook, Fourth Edition. Chapter 17 (pp. 17-1 to 17-10). Table of Contents. All contents are Copyright .COPYRGT. 1992-2002 Cisco Systems, Inc. All rights Reserved. Retrieved from the Internet on May 12, 2004--http://www.ciscopress.com/bookstore/LargeCover.asp?isbn=1587051192. cited by other .
"1983 Index, IEEE Journal on Selected Areas in Communications vol. SAC-1", pp. 1-13 (1983). cited by other .
"A history of video conferencing (VC) technology", 2 pages (date unknown). cited by other .
"A Family of 2-Wire, Duplex Modems Operating at Data Signalling Rates", CCITT, V. 32, Facsimile Recommendation (1988). cited by other .
"Applications of the MT8841 Calling Number Identification Circuit", Mitel Microelectronics Digital/ Analog Communications Handbook, Issue 1, pp. 12-307 (1993). cited by other .
"Communications for the Information Age", IEEE Global Telecommunications Conference & Exhibition, 1 of 3, pp. 34-36 (Nov. 28-Dec. 1). cited by other .
"Composite Mini-Packet Switching", IBM Technical Disclosure Bulletin, 32(5A):162-164 (1989). cited by other .
"Data Communication Networks: Services and Facilities, Terminal Equipment and Interfaces, Recommendations X.1-X.29", CCITT Yellow Book, vol. VIII-Fascicle VIII.2, pp. 100-190 (1981). cited by other .
"Data Communication Over the Telephone Network", CCITT Blue Book, vol. VIII, (1989). cited by other .
"Data Communication Networks: Transmission, Signalling and Switching, Network Aspects, Maintenance, Administrative Arrangements, Recommendations X.40-X.180", CCITT Yellow Book, vol. VIII-Fascicle VIII.3, pp. 142-208 (1980). cited by other .
"Data Communication Over the Telephone Network: Data Compression Procedures for Data Circuit Terminating Equipment (DCE) Using Error Correction Procedures", CCITT Recommendation V. 42 bis (1990). cited by other .
"Data Communication Over the Telephone Network: Definitions of Terms Concerning Data Communication over the Telephone Network", ITU-T Recommendation V. 7 (1993). cited by other .
"DEC Sound/Picture Information Network, (DECspin) for ULTRIX, Version 1.0", Digital Corp., 4 pages (1992). cited by other .
"Dial-up comm server", LAN Technology, 7(12):87 (1991). cited by other .
"Error Control for Voice/Data Packet Network", IBM Technical Disclosure Bulletin, 29(7):3092-3095 (1986). cited by other .
"Error-Correcting Procedures for DCES Using Asynchronous-to-Synchronous Conversion", CCITT V.42, vol. VIII, pp. 296-370 (1988). cited by other .
"Experimental Internet Stream Protocol, Version 2 (ST-II)", Network Working Group, CIP Working Group, 148 pages (1990). cited by other .
"Forthcoming Special Issues and Mini-Special Issues of IEEE Journal on Selected Areas in Communications and IEEE Transactions on Communications for 1984-1985", IEEE Journal on Selected Areas in Communications, SAC-1(6):1155-1158 (1983). cited by other .
"Full Duplex Speakerphone", IBM Technical Disclosure Bulletin, 29(12):5599-5602 (1987). cited by other .
"Go Global Without Leaving Your Office", Artel Technology, 4 pages (date unknown). cited by other .
"High Speed Data Pump Chip Sets", AT&T Microelectronics, pp. 1-91 & F1-F20 (Dec. 1991). cited by other .
"Information Processing Techniques Program. vol. II. Wideband Integrated Voice/Data Technology", Massachusetts Inst. of Tech., Lexington, Lincoln Lab, 36 pages (1978). cited by other .
"Integrated Voice/Data PABX", IBM Technical Disclosure Bulletin, pp. 1537-1538 (Sep. 1986). cited by other .
"ISDN Systems: Achitecture, Technology, and Applications", Prentice-Hall, Inc., Englewood Cliffs, New Jersey (1990). cited by other .
"Jane's Military Communications, 1981", Jane's Publishing Company Limited, London (1981). cited by other .
"Jane's Military Communications, 1982", Jane's Publishing Company Limited, London (1982). cited by other .
"Jane's Military Communications, 1983", Jane's Publishing Company Limited, London (1983). cited by other .
"Jane's Military Communications, 1986", Jane's Publishing Company Limited, London (1986). cited by other .
"Jane's Military Communications, 1987", Jane's Publishing Company Limited, London (1987). cited by other .
"Mechanism for Integrating Voice and Data on a Transmission Channel", IBM Technical Disclosure Bulletin, 30(5):208-209 (1987). cited by other .
"Method and Apparatus for the Statistical Multiplexing of Voice, Data, and Image Signals", IBM Technical Disclosure Bulletin, 35, pp. 409-411 (Nov. 1992). cited by other .
"Mitel MT8841 Calling Number Identification Circuit", Mitel Microelectronics Digital/Analog Communications Handbook, Issue 9, pp. 8-93-8-102 (1993). cited by other .
"Mitel MT8841 Calling Number Identification Circuit, Issue 4", 16 pages (1995). cited by other .
"Mitsubishi Still Color Picture TV Phone", Techno Japan, 23(6) (Jun. 1990). cited by other .
"MultiX25--X.25 PAD, The New MultiX25 PAD 8 Port X.25 Packet Assembler/Disassembler for Public and Private Data Networks", MultiTech Systems. cited by other .
"Network Speech. System Implications of Packetized Speech", Massachusetts Institute of Tech., Lexington, Lincoln Laboratory, 30 pages (1976). cited by other .
"Northern Telecom's Displayphone", What Telephone, pp. 56-59 (year unknown). cited by other .
"One Per Desk (OPD)", Product Description, 24 pages (dated 2000). cited by other .
"Output Throttle Mechanism", IBM Technical Disclosure Bulletin, 32(11):274-279 (1990). cited by other .
"PC-Based ISDN Visual Communication System, Preliminary Information", Artel Technology, 2 pages (date unknown). cited by other .
"Pen Telephone Brochure", Shimadzu, Published Prior to Applicant's Invention. cited by other .
"PICFON Card Brochure", Specom Technologies Corp., Published Prior to Applicant's Invention. cited by other .
"PICTURETEL, PictureTel's High Performance VideoConferencing is Redefining the Way the World Meets", PictureTel Corporation, 3 pages (date unknown). cited by other .
"Radish System Lets Users Send Voice, Data Simultaneously", PC Week, 9(19):53 (May 11, 1992). cited by other .
"Report on the MBone evaluations", 3 pages (date unknown). cited by other .
"Series V: Data Communication over the Telephone Network, Error control, Error-correcting procedures for DCEs using asynchronous-to-synchronous conversion", ITU-T Recommendation V. 42, pp. 1-66 (1996). cited by other .
"Speech Data Adaptive Multiplexer", IBM Technical Disclosure Bulletin, 27(2):969 (Jul. 1994). cited by other .
"STU-III Handbook for Industry", Department of Defense Security Institute, 10 pages (Feb. 1997). cited by other .
"T7540 Digital Telephone CODEC Preliminary Data Sheet", AT&T Microelectronics, pp. 1-64 (Jan. 1991). cited by other .
"T7540 Digital Telephone CODEC Data Sheet and Addendum", AT&T Microelectronics, pp. 1-4 (Jul. 1991). cited by other .
"TechTips--A Periodic Round-up of Technical Applications, Notes and Information on MultiTech's Data Communications Products", MultiTech Systems, 2(2) (May 1992). cited by other .
"Telewriter Product Description", Optel Communications, Inc., Published Prior to Applicants Invention. cited by other .
"Telewriting Terminal Equipment", CCITT, Recommendation T. 150, pp. 228-277 (1988). cited by other .
"The Photophone", GTE Product Brochure, (Feb. 15, 1990). cited by other .
"Token-Ring Migration to Switched LAN Interconnect--A technical White Paper", pp. 1-19 (date unknown). cited by other .
"Videowriter '91 Product Description", Optel Communications, Inc. (1991). cited by other .
"Voice/Data Concentrator", IBM Technical Disclosure Bulletin, pp. 3066-3067 (Jan. 1979). cited by other .
"WE DSP16C Digital Signal Processor/CODEC Preliminary Data Sheet", AT&T Microelectronics, pp. 1-64 (May 1991). cited by other .
"Z84C01 Z80 CPU with Clock Generator/Controller", Zilog Intelligent Peripheral Controllers, pp. 43-73 (1991). cited by other .
"Z84C90 CMOS Z80 KIO Serial/Parallel/counter/timer", Zilog Intelligent Peripheral Controllers, pp. 205-224 (1991). cited by other .
Abbiate et al., "Voice and Data Transmission", IBM Technical Disclosure Bulletin, 27(1B):524-525 (1984). cited by other .
Addeo et al., "Personal Multi-Media Multi-Point Communication Services for Broadband Networks", Globecom '86: IEEE Global Telecommunications Conference, Dec. 1-4, 1986, Houston, Texas, Conference Record, vol. 1 of 3, pp. 0053-0057 (1988). cited by other .
Ades et al., "Protocols for Real Time Voice Communication on a Packet Local Network", University of Cambridge Computer Laboratory, 6 pages (date unknown). cited by other .
Adolphs et al., "Adapters for the Public ISDN", Electrical Communication, 61(1) 9 pages (1987). cited by other .
Adolphs et al., "Subsets, Terminals, and Terminal Adapters for the Public ISDN", Electrical Communication, 61(1):72-80 (1987). cited by other .
Aguilar et al., "Architecture for a Multimedia Teleconferencing System", ACM 0-89791-201-2/86/0800-0126, pp. 126-136 (1986). cited by other .
Ahuja et al., "The Rapport Multimedia Conferencing System", Conference on Office Information Systems, Mar. 23-25, 1988, Palo Alto, California, Association for Computing Machinery, Inc., pp. 1-8 (1988). cited by other .
Aikoh et al., "A New TDM Technique for Voice, Data and Video Transmission", Globecom '87, Nov. 15-18, 1987, Tokyo, Japan, pp. 28.7.1-28.7.5 (1987). cited by other .
Akaiwa et al., "An Integrated Voice and Data Radio Access System", IEEE, pp. 255-258 (1992). cited by other .
Akiyama et al., Digital Sketchphone--Telewriting Terminal for INS Model System--Review of the Electrical Communication Laboratories, 33(2):250-259 (1985). cited by other .
Anderson et al., "Toolkit Support for Multiuser Audio/Video Applications", pp. 230-241 (date unknown). cited by other .
Andrews et al., "Faster Packets For Tomorrow's Telecommunications", AT&T Technology, 3(4):25-33 (date unknown). cited by other .
Angebranndt et al., "Integrating Audio and Telephony in a Distributed Workstation Environment", Proceedings of the Summer 1991 Usenix Conference, Jun. 10-14, 1991, Nashville, Tennessee, pp. 419-435 (1991). cited by other .
Aoki et al., "Protocol Processing for High-Speed Packet Switching Systems", Proceedings of the International Seminar, Zurich, Switzerland, Mar. 11-13, 1986, pp. 125-130 (1986). cited by other .
Arango et al., "Touring Machine: A Software Platform for Distributed Multimedia Applications", Submitted to the 1992 IFIP International Conference on Upper Layer Protocols, Architectures, and Applications, Vancouver, Canada, May 1992, 12 pages (1992). cited by other .
Armbruster, Heinrich, "Broadband ISDN--The Network of the Future: Applications and Compliance with User Requirements", Globecom '86: IEEE Global Telecommunications Conference, Dec. 1-4, 1986, Houston, Texas, Conference Record, vol. 1 of 3, pp. 0484-0490 (1986). cited by other .
Arthurs et al., "Traffic Analysis Tools for Integrated Digital Time-Division Link Level Multiplexing of Synchronous and Asynchronous Message Streams", IEEE Journal on Selected Areas in Communications, SAC-1(6):1112-1123 (1983). cited by other .
ATT&T Modem Designer's Guide, AT&T Microelectronics (Oct. 25, 1993). cited by other .
Ball et al., "A Multimedia Network Interface", 6 pages (date unknown). cited by other .
Barberis et al., "Analysis and Optimal Design of a Packet-Voice Receiver", IEEE Transactions on Communications, COM-28(2):217-227 (1980). cited by other .
Barberis et al., "Coded Speech in Packet-Switched Networks: Models and Experiments", IEEE Journal on Selected Areas in Communications, SAC-1(6):1028-1038 (1983). cited by other .
Beadle et al., "The Human Side of Multimedia Telecommunications", Appears in Proceedings of the 2.sup.nd Australian Multimedia Communications, Applications and Technology Workshop, AOTC Ltd, Melbourne, Jul. 13-14, 1992, 11 pages (1992). cited by other .
Beckmann, William H., "Voice and Data Integration: An Applications Perspective", LOCALNET '84, Proceedings of the conference held in San Diego USA, Oct. 1, Online Publications, New York, pp. 303-327 (1984). cited by other .
Bellamy, John, "Digital Telephony" John Wiley & Sons, New York, pp. 392-412 (1982). cited by other .
Bellcore Technical Reference TR-NWT-000030, Issue 2,(Oct. 1992). cited by other .
Bially et al., "Voice Communication in Integrated Digital Voice and Data Networks", IEEE Transactions on Communications, COM-28(9):1478-1490 (1980). cited by other .
Blair et al., "Incorporating Multimedia in Distributed Open Systems", Presented at EUUG Spring Conference, Tromso, Norway, Apr. 1991, 12 pages (1991). cited by other .
Blair et al., "Orchestration Services for Continuous Media Communications", Presented at 4.sup.th Bangor Communication Symposium, Bangor, Gwynedd, May 27-28th, 1992, 10 pages (1992). cited by other .
Blanc et al., "Voice/Data Multiplexing in a Multi-Signal Processor-Synchronized Architecture", IBM Technical Disclosure Bulletin, 28(5):2059-2060 (1985). cited by other .
Boese et al., "The Multimedia SCP", IEEE/IEICE Global Telecommunications Conference 1987, Nov. 15-18, 1987, Tokyo, Japan, Conference Record, vol. 1 of 3, pp. 112 and 116 (1991). cited by other .
Borgnis-Desbordes et al., "Variable-Speed Data Transmission", IBM Technical Disclosure Bulletin, 27(4A):2269-2270 (1984). cited by other .
Borgnis-Desbordes et al., "Voice and Data Transmission", IBM Technical Disclosure Bulletin, 25(8):4474-4475 (1983). cited by other .
Brown, Howard, "British Telecom's ISDN", Computer Communications, 11(4):177-180 (1988). cited by other .
Bubenik et al., "Multipoint Connection Management in High Speed Networks", INFOCOM '91: Proceedings of the Tenth Annual Joint Conference of the IEEE Computer and Communications Societies, pp. 59-68 (1991). cited by other .
Cain, Roger W., "Microprocessor Based 9600 BPS Modem", ICASSP 85: IEEE International Conference on Acoustics, Speech, and Signal Processing, Mar. 26-29, 1985, Tampa, Florida, Proceedings, vol. 4 of 4, pp. 1633-1636 (1985). cited by other .
Callens et al., "Speech Data Adaptive Multiplexer", IBM Technical Disclosure Bull., 27:969 (Jul. 1984). cited by other .
Campbell et al., "A Continuous Media Transport and Orchestration Service", Presented at ACM SIGCOMM '92, Maryland, USA, 12 pages (1992). cited by other .
Casale et al., "Statistical voice/high speed data multiplexing on a 64 kbits/s channel", Phoenix Conference on Computers and Communications, Scottsdale, AZ, pp. 459-464 (Mar. 27, 1991). cited by other .
Casner et al., "First IETF Internet Audiocast", Reprinted from ACM SIGCOMM Computer Communications Review, vol. 22, No. 3 (Jul. 1992), ISI Reprint Series ISI/RS-92-293, pp. 1-6 (1992). cited by other .
Casner et al., "N-Way Conferencing with Packet Video", Reprinted from the Proceedings of The Third International Workshop on Packet Video, Mar. 22-23, 1990 in Morristown, New Jersey, ISI Reprint Series ISI/RS-90-252, pp. 1-6 (1990). cited by other .
Cerf, V., "Packet Satellite Technology Reference Sources", Defense Advanced Research Projects Agency, 5 pages (dated 1982). cited by other .
Chao, H. Jonathan, "Design of Transmission and Multiplexing Systems for Broadband Packet Networks", IEEE Journal on Selected Areas in Communications, 6(9):1511-1520 (1988). cited by other .
Chao et al., "A Packet Video System Using The Dynamic Time Division Multiplexing Technique", Globecon '86: IEEE Global Telecommunications Conference, Conference Record, vol. 3 of 3, pp. 0767-0772 (1986). cited by other .
Chen et al., "A Fair Integrated Voice/Data Protocol for Token Ring Networks", Globecom '90: IEEE Telecommunications Conference & Exhibition, Dec. 2-5, 1990, San Diego, California, Conference Record, vol. 3 of 3, pp. 1416-1420 (1990). cited by other .
Chen et al., "Integrated Voice/Data Switching", IEEE Communications Magazine, 26(6):16-26 (1988). cited by other .
Chen, Juin-Hwey, "A Robust Low-Delay CELP Speech Coder at 16Kbits/s", CH2682-3/89/0000-1237, IEEE, pp. 1237-1241 (1989). cited by other .
Chiariglione et al., "Multimedia Communication at the Basic ISDN Access", CH2190-7/85/0000-0363, IEEE, pp. 363-367 (1985). cited by other .
Chiou et al., "GATENET: A Voice/Data Internet Transport System", IEEE Infocom '86: Fifth Annual Conference, Apr. 8-10, 1986, Miami, Florida, pp. 39-46 (1986). cited by other .
Chlamtac, Imrich, "An Ethernet Compatible Protocol for Real-Time Voice/Data Integration", Computer Networks and ISDN Systems, 10(2):81-96 (1985). cited by other .
Chlamtac et al., "Performance of Integrated Services(Voice/Data) CSMA/CD Networks", ACM 0-89791-169-5/85/007/0087, pp. 87-93 (1985). cited by other .
Choi et al., "An Efficient Multiplexing Technique for Packet-Switched Voice/Data Networks", Proceedings of the IEEE, 76(9):1254-1256 (1988). cited by other .
Chujo et al., "Personal Communication Terminals for ISDN", Globecom '87, Nov. 15-18, 1987, Tokyo, Japan, pp. 19.4.1-19.4.5 (1987). cited by other .
Clark et al., "Supporting Real-Time Applications in an Integrated Services Packet Network: Architecture and Mechanism", ACM 0-89791-526-7/92/0008/0014, pp. 14-26 (1992). cited by other .
Cohen, Danny, "A Network Voice Protocol NVP-II", USC/Information Sciences Institute, 71 pages (dated 1981). cited by other .
Cohen, Danny, "Packet communication of online speech", 1981 National Computer Conference, May 4-7, 1981, Chicago Illinois, Conference Proceedings, pp. 169-176 (1981). cited by other .
Cohen, Danny, "Satellite Communication of Real-Time Packet Video Images", Preprinted from the Proceedings of the Seventh International Conference on Computer Communication, Sydney, Australia, Oct. 30, Nov. 2, 1984, pp. 1-8 (1984). cited by other .
Cohen, Danny, "Specifications for the Network Voice Protocol (NVP)", NSC Note 68, 35 pages (1977). cited by other .
Cohen, Danny, "Using Local Area Networks for Carrying Online Voice", Local Computer Networks, North-Holland Publishing Company, Amsterdam, pp. 13-21 (1982). cited by other .
Cohen, Danny, "A Voice Message System", Proceedings of the IFIP TC-6 International Symposium on Computer Message Systems, Apr. 6-8, 1981, Ottawa, Canada, pp. 17-28 (1981). cited by other .
Cole, Randy, "Packet Voice: When it Makes Sense", Speech Technology, pp. 52-61 (Sep./Oct. 1982). cited by other .
Cole, Randy, "PVP-A Packet Video Protocol", Information Sciences Institute, University of Southern California, 15 pages (dated 1981). cited by other .
Coulson et al., "Group Presentation of Multimedia Applications in IXION", In Computer Communications, 14(4):205-215 (1991). cited by other .
Crauwels et al., "Voice Response for Data Processing Equipment", IBM Technical Disclosure Bulletin, 24(11B):5890-5895 (1982). cited by other .
Crowcroft et al., "Multimedia TeleConferencing over International Packet Switched Networks", TriComm '91: IEEE Conference on Communications Software: Communications for Distributed Applications & Systems, Apr. 18-19, 1991, Chapel Hill, North Carolina, pp. 23-33 (1991). cited by other .
Cruz et al., " Tools to Support Conversational Multimedia", Globecom '91: IEEE Global Telecomunications Conference, Dec. 2-5, 1991, Phoenix, Arizona, Conference Record, vol. 3 of 3, pp. 2064-2068 (1991). cited by other .
Cuperman et al, "Backward Adaptive Configurations for Low-delay Vector Excitation Coding", Advances in Speech Coding, pp. 13-23 (Jan. 1, 1996). cited by other .
Data Sources, 11(2):G-38 (1992). cited by other .
Davies et al, "Experiences of Handling Multimedia in Distributed Open Systems", Published in Proc. 3.sup.rd Usenix International Symposium on Experiences with Distributed and Multiprocessor Systems, Newport Beach, CA, USA, Mar. 1992, 19 pages (1992). cited by other .
Davies et al., "A Technological Perspective on Multimedia Computing", Computer Communications, 14(5):1-19 (1992). cited by other .
Decina et al., "Voice by the Packet?", IEEE Journal on Selected Areas in Communications, SAC-1(6):961-962 (1983). cited by other .
Delgrossi et al., "HeiTP-A Transport Protocol for ST-II", 0-7803-0608-2/92/53, IEEE, 1 page (1992). cited by other .
DeSchon, Annette,"Voice Terminal (VT) Program", USC Information Sciences Institute, 20 pages (dated 1991). cited by other .
DeTreville et al., "A Distributed Experimental Communications System", IEEE Journal on Selected Areas in Communications, SAC-1(6):1070-1075 (1983). cited by other .
Devault et al., "The `Prelude` ATD Experiment: Assessments and Future Prospects", IEEE Journal on Selected Areas in Communications, 6(9):1528-1537 (1988). cited by other .
Doll, Dixon R., "Data Communications: Facilities, Networks, and Systems Design", John Wiley & Sons, New York (1978). cited by other .
Donovan, Terry, "The mux and modem march on", Communicate, pp. 30-33 (1990). cited by other .
Dunlap, Jr. et al., "Personal Computer Communications Via Telephone Facilities", IEEE Journal on Selected Areas in Communications, SAC-3(3):399-407 (1985). cited by other .
Dunlop, John, "Techniques for the integration of packet voice and data on IEEE 802.3 LANs", Computer Communications, 12(5):273-280 (1989). cited by other .
Economou et al., "FBC-Net: a reverse round robin LAN for voice and data traffic", Computer Communications, 14(1):44-52 (1991). cited by other .
Eigen, Daryl J., "Narrowband and Broadband ISDN CPE Directions", IEEE Communications Magazine, pp. 39-46 (Apr. 1990). cited by other .
Eigen et al., "Subscriber Access to Broadband ISDN", CH2298-9/86/0000-1410, IEEE, pp. 1410-1415 (1986). cited by other .
Elbert, Bruce R., "Private Telecommunication Networks", Artech House, Inc., Norwood, Masachusetts (1989). cited by other .
Elhakeem et al., "Spread-Spectrum Access to Mixed Voice-Data Local Area Networks", IEEE Journal on Selected Areas in Communications, SAC-1(6):1054-1057 (1983). cited by other .
Erskine et al., "VoicePump: Voice-Over-Data Modem technology Utilizing ASM-CELP Speech Compression at 4800, 7200 and 9600 Bits per Second", Proceedings of the International Conference on Signal Processing Applications and Technology, 2:1425-1429 (Sep. 28, 1993). cited by other .
Falk et al., "Integration of Voice and Data in the Wideband Packet Satellite Network", IEEE Journal on Selected Areas in Communications, SAC-1(6): 1076-1083 (1983). cited by other .
Fang, Russell J.P., "Personal and Thin-Route Communications via K-Band Satellite Transponders", CH2981-9-91-0000-1177, IEEE, pp. 1177-1182 (1991). cited by other .
Ferrari et al., "A Scheme for Real-Time Channel Establishment in Wide-Area Networks", IEEE Journal on Selected Areas in Communications, 8(3):368-379 (1990). cited by other .
Field et al., ".alpha.-Channel, A Network Level Abstraction To Support Real-Time Communication", Computer Science Department, University of Pittsburgh, 12 pages (date unknown). cited by other .
Forgie, James W., "Voice Conferencing in Packet Networks", ICC '80: International Conference on Communications, Jun. 8-12, 1980, Seattle, Washington, Conference Record, vol. 1 of 3,21.3.1- 21.3.4 (1980). cited by other .
Forgie et al., "An Efficient Packetized Voice/Data Network Using Statistical Flow Control", 1977 International Conference on Communications, Jun. 12-15, Chicago, Illinois, vol. 3 of 3, pp. 44-48 (1977). cited by other .
Forgie, James W., "Speech transmission in packet-switched store-and-forward networks", AFIPS Conference Proceedings, 1975 National Computer Conference, May 19-22, 1975, Anaheim, California, pp. 137-142 (1975). cited by other .
Forsdick et al., "Explorations into Real-time Multimedia Conferencing", Computer Message Systems, 85:331-347 (1986). cited by other .
Frangoudes et al., "Software for Broadband Systems: An ISDN Based Approach", pp. 286-291 (date unknown). cited by other .
Friedman et al., "Packet Voice Communications Over PC-Based Local Area Networks", IEEE Journal on Selected Areas in Communications, 7(2):211-218 (1989). cited by other .
Gaddis et al., "A Call Model For Multipoint Communication in Switched Networks", From ICC '92: Proceedings of the IEEE International Conference on Communications, pp. 609-615 (1992). cited by other .
Gergaud et al., "Voice and Data Transmission", IBM Technical Disclosure Bulletin, 25(6):2727-2728 (1982). cited by other .
Gergaud, C., "Voice Data Integration on a Transmission Link", IBM Technical Disclosure Bulletin, 26(11):5780-5781 (1984). cited by other .
Gitman et al., "Economic Analysis of Integrated Voice and Data Networks: A Case Study", Proceedings of the IEEE, 66(11):1549-1570 (1978). cited by other .
Gitman et al., "Issues in Integrated Network Design", ICC 77: 1977 International Conference on Communications, Jun. 12-15, Chicago, Illinois, Conference Record, vol. 3 of 3, pp. 38.1-36-38.1-43 (1977). cited by other .
Globecom '85: IEEE Global Telecommunications Conference, Dec. 2-5, 1985, New Orleans, Louisiana, Conference Record, vol. 1 of 3, 1 page (2 copies) (1985). cited by other .
Globecom '92: IEEE Global Telecommunications Conference, Dec. 6-9, 1992, Orlando, Florida, Conference Record, vol. 1 of 3, 1 page (1992). cited by other .
Gold, Bernhard, "Digital Speech Networks", Proceedings of the IEEE, 65(12):1636-1658 (1977). cited by other .
Golestani, S. Jamaloddin, "Duration-Limited Statistical Multiplexing of Delay-Sensitive Traffic in Packet Networks", IEEE Infocom '91: Tenth Annual Joint Conference of the IEEE Computer and Communications Societies, Proceedings, vol. 1, pp. 0323-0332 (1991). cited by other .
Gonet, P., "Fast packet approach to integrated broadband networks", Computer Communications, 9(6):292-298 (1986). cited by other .
Gopalakrishnan et al., "Integrating voice and data SALAN: an experimental local area network", Computer Communications, 9(4):186-194 (1986). cited by other .
Goutmann, Michel M., "Digital Conferencing Units, Third Quarterly Report", Defense Technical Information Center, AD18849, 36 pages (1974). cited by other .
Gruber et al., "Performance Requirements for Integrated Voice/Data Networks", IEEE Journal on Selected Areas in Communications, SAC-1(6):981-1005 (1983). cited by other .
Gulick et al., "Interface the ISDN to your PC with a Voice/Data Board", Electronic Design, 35(29):85-88 (Dec. 10, 1987). cited by other .
Hart et al., "A Low Cost System for Interactive Transmission of Text and Voice in Distance Education", Second Symposium on Computer Conferencing and Allied Technologies, University of Guelph, Guelph, Ontario, Canada, Jun. 1-4, 1987, pp. 263-269 (1987). cited by other .
Hattori et al., "Integrated Digital Switching System with Queueing Storage Facility", IEEE Transactions on Communications, COM-30(8):1900-1905 (1982). cited by other .
Heggestad, Harold M. "Voice and Data Communications on an Experimental Wideband Internetwork System", Journal of Telecommunication Networks, pp. 131-146 (date unknown). cited by other .
Hoberecht, William L., "A Layered Network Protocol for Packet Voice and Data Integration", IEEE Journal on Selected Areas in Communications, SAC-1(6): 1006-1013 (1983). cited by other .
Honda et al., "Communications Processing Gateway Switching System", Review of the Electrical Communication Laboratories, 32(6):957-966 (1984). cited by other .
Hopper, Andy, "Design and Use of High Speed Networks in Multimedia Applications", Published in Proceedings of 3.sup.rd IFIP Conference on High Speed Networking Berlin, Mar. '91, 15 pages (1991). cited by other .
Hopper, Andy, "Digital Video on Computer Workstations", To be published in Proceedings of Eurographics '92, Sep. 1992, 15 pages (dated 1992). cited by other .
Hopper, Andy, "Pandora-an experimental system for multimedia applications", published in Operating Systems Review, pp. 1-16 (dated Jan. 12, 1990). cited by other .
Hosoda et al., "A 32 kbit/s ADPCM Algorithm Having High Performance for Both Voice and 9.6 kbit/s Modem Signals", IEEE Journal on Selected Areas in Communications, pp. 262-272 (1988). cited by other .
Huang et al., "MHTP-A Multimedia High-Speed Transport Protocol", Globecom '92: IEEE Global Telecommunications Conference, Dec. 6-9, 1992, Conference Record, vol. 3 of 3, pp. 1364-1369 (1992). cited by other .
Huang et al., "Multifunction Workstation Integrated Voice/Data/Graphics for the ISDN Environment", IEEE Pacific Rim Conference on Communications, Computers and Signal Processing, May 9-10, 1991, Victoria, B.C., Canada, Conference Proceedings, vol. 2 of 2, pp. 441-444 (1991). cited by other .
Huang, Nian-Chyi, "An Integrated Voice/Data Network Architecture using Virtual Circuit Switching", CH2190-7/85/0000-0542, IEEE, pp. 542-546 (1985). cited by other .
Huelamo et al., "End User Premises Equipment and Terminals for Broadband Applications", Electrical Communication, 64(2/3), 11 pages (1990). cited by other .
Ibe et al., "Treenet: an Integrated Metropolitan Area Network", Globecom '85: IEEE Global Telecommunications Conference, Dec. 2-5, 1985, New Orleans, Louisiana, Conference Record, vol. 1 of 3, pp. 262-266 (1985). cited by other .
Ichikawa et al., "High-Speed Packet Switching Systems for Multimedia Communications", IEEE Journal on Selected Areas in Communications, SAC-5(8):1336-1345 (1987). cited by other .
Isozaki, Kiyoshi, "Commercial CAPTAIN System", JTR, pp. 246-253 (Oct. 1983). cited by other .
Jacobson, V., "TCP Extensions for High Performance", Network Working Group, 35 pages (dated 1992). cited by other .
Jacobson et al., "TCP Extension for High-Speed Paths", Network Working Group, 20 pages (dated 1990). cited by other .
Jacobson, D.W., "Design and Implementation of an Integrated Voice, Data, and Video Services Network",IEEE Pacific Rim Conference on Communications, Computers and Signal Processing, Jun. 1.sup.st -2.sup.nd, 1989, Victoria, B.C., Canada, pp. 500-505 (1989). cited by other .
Janakiraman, Natesa, "An Overview of Recent Developments in the Designs and Applications of Customer Premises Switches", IEEE Communications Magazine, 23(10):32-45 (1985). cited by other .
Jander, Mary, "A Voice/Data/Fax Multiplexer", New Products/WAN, 19(6):146 (1990). cited by other .
Jander, Mary, "Voice over packet networks: A solution available now ", Data Communications, 19(2):129 (1990). cited by other .
Jeffay et al., "Designing a Workstation-Based Conferencing System Using the Real-Time Producer/Consumer Paradigm", Defense Technical Information Center, AD-A242 046, 20 pages (1990). cited by other .
Jeffay et al., "Kernel Support for Live Digital Audio and Video", University of North Carolina at Chapel Hill, Department of Computer Science, pp. 10-20 (date unknown). cited by other .
Jenq, Yih-Chyun, "Performance Analysis of a Packet Switch Based on Single-Buffered Banyan Network", IEEE Journal on Selected Areas in Communications, SAC-1(6):1014-1038 (1983). cited by other .
Johnson et al., "A Local Access Network for Packetized Digital Voice Communication", IEEE Transactions on Communications, COM-29(5):679-688 (1981). cited by other .
Johnston et al., AT&T News Release, Monday, Jan. 6, 1992, 2 pages. cited by other .
Jordan et al., "Communications and Networking for IBM PC", Robert J. Brady Company, Bowie Maryland, 3 pages (1983). cited by other .
JP 63054052, Patent Abstracts of Japan, vol. 15, No. 397 (E-1120), Oct. 8, 1991. cited by other .
JP 3162052, Patent Abstracts of Japan, vol. 12, No. 273 (E-639), Jul. 29, 1988. cited by other .
Kadoch et al., "A Distributed Queue Metropolitan Area Network for Combined Video, Voice, Wideband Data and Narrowband Data", IEEE Global Telecommunications Conference & Exhibition, Nov. 27-30, 1989, Dallas, Texas, Conference Record, vol. 1 of 3, pp. 0152-0156 (1989). cited by other .
Kamae, T., "Visual Terminals and User Interfaces", pp. 257-278 (date unknown). cited by other .
Kamae, T., "Voice/Data Integration in the INS Model System and Local Area Networks", IEEE Communications Magazine, 24(12):7-15 (1986). cited by other .
Kamae et al., "Customer Equipment for New Services in INS Model System", Proceedings of the Seventh International Conference on Computer Communication, Sydney, Australia, Oct. 30-Nov. 2, 1984, pp. 72-76 (1984). cited by other .
Kapauan et al., "Wideband Packet Access for Workstations: Integrated Voice/Data/Image Services on the UNIX PC", CH2298-9/86/0000-1439, IEEE, pp. 1439-1441 (1986). cited by other .
Kashorda et al., "Simultaneous Voice/Data Transmission over a Single 64 Kb/s ISDN Bearer Channel", Third IEE Conference on Telecommunications, Institution of Electrical Engineers, London, pp. 281-300 (1991). cited by other .
Kessler, Gary C., "ISDN: Concepts, Facilities, and Services", McGraw-Hill, Inc., New York, pp. 224-231 (1990). cited by other .
Kim, Byung G., "Characterization of Arrival Statistics of Multiplexed Voice Packets", IEEE Journal on Selected Areas in Communications, SAC-1(6):1133-1141 (1983). cited by other .
Kim, Byung G., "Integrated Voice/Data Multiplexers with Slotted and Nongated Packet Services", IEEE Transactions on Communications, 36(2):228-231 (1988). cited by other .
Kim et al., "Dynamic Flow Control Protocols for Packet-Switching Multiplexers Serving Real-Time Multipacket Messages", IEEE Transactions on Communications, COM-34(4):348-356 (1986). cited by other .
Kimura et al., "Basic Scheme for the INS Network", Review of the Electrical Communications Laboratories, 33(5):751-756 (1985). cited by other .
King, Tony, "Pandora: An Experiment in Distributed Multimedia", To be published in the Proceedings of Eurographics '92, Sep. 1992, 12 pages (1992). cited by other .
Kishimoto et al., "Simultaneous Transmission of Voice and Handwriting Signals: Sketchphone System", IEEE Transaction on Communication, COM-29, 12, pp. 1982-1986, (Dec. 1981). cited by other .
Kitawaki et al., "Speech Coding Technology for ATM Networks", IEEE Communications Magazine, pp. 21-27 (Jan. 1990). cited by other .
Kleckner, C. C., "A New Family of Modems for Faster Data Communications", 3 pages (date unknown). cited by other .
Kobayashi et al., "Videotex Terminals for the INS Model System", Review of the Electrical Communication Laboratories, 33(2):277-283 (1985). cited by other .
Komiya et al., "An approach to the multifuction graphic terminal for the ISDN environment", IEEE Global Telecommunications Conference and Exhibition, 1:32-36 (Nov. 28, 1988). cited by other .
Kositpaiboon et al., "Packetized Radiographic Image Transfers over Local Area Networks for Diagnosis and Conferencing", IEEE Journal on Selected Areas in Communications, 7(5):842-856 (1989). cited by other .
Kramer, Matt, "Integrated voice/data circus comes to town", PC Week, 7(11):63, 4 pages (1990). cited by other .
Krol, Ed, "Faster Than a Speeding Modem . . . Able to Cary Voice and Data over a Single Telephone Line . . . It's ISDN", UIUCnet V5-7, 3 pages (1992). cited by other .
Kuhar, E., "Digital Data Communication over Twisted Pair Voice Telephone Wiring", IBM Technical Disclosure Bulletin, 25(7A):3482-3487 (1982). cited by other .
Laube, M., "Audiographic Terminal", Electrical Communication, 60(1) (1986). cited by other .
Lee et al., "Performance Analysis of Statistical Voice/Data Multiplexing Systems with Voice Storage", IEEE Transactions on Communications, COM-33(8):809-819 (1985). cited by other .
Leung et al., "A Set of Operating System Mechanisms to Support Multi-Media Applications", 1988 International Zurich Seminar on Digital Communications, Mar. 8-10, 1988, Eth Zurich, Switzerland, pp. B4.1-B4.6 (1988). cited by other .
Leung et al., "A Software Architecture for Workstations Supporting Multimedia Conferencing in Packet Switching Networks", IEEE Journal on Selected Areas in Communications, 8(3):380-390 (1990). cited by other .
Leung et al., "The Throughput Efficiency of the Go-Back-N ARQ Scheme Under Markov and Related Error Structures", IEEE Transactions on Communications, 36(2):231-234 (1988). cited by other .
Leydekkers et al., "Synchronization of Multimedia Data Streams in Open Distributed Environments", pp. 94-104 (date unknown). cited by other .
Lichtenberg et al., "Digital PABX Range for the Evolving ISDN", Electrical Communication, 61(1), 10 pages (1987). cited by other .
Little et al., "Scheduling of Bandwidth-Constrained Multimedia Traffic", pp. 120-131 (date unknown). cited by other .
Liu et al., "A Modified ATM Scheme for Low Bit Rate Multi-Media Applications", CH 3006-4/91/0000-1033, IEEE, pp. 1033-1036 (date unknown). cited by other .
Loudermilk et al., "CDPD: a network that wants the airwaves: cellular technology is primed for growth", PC Week, 9(49):26 5 pages (1992). cited by other .
Loudermilk, Stephen, "Radish system lets phone users send voice, data simultaneously", PC Week, 9:(19):53 (1992). cited by other .
Mahmoud et al., "An Integrated Voice/Data System for VHF/UHF Mobile Radio", IEEE Journal on Selected Areas in Communications, SAC-1(6):1098-1111 (1983). cited by other .
Maise et al., "Experiments in Wideband Packet Technology", Digital Communications: New Directions in Switching and Networks, Proceedings of the International Seminar, Zurich, Switzerland, Mar. 11-13, 1986, IEEE Switzerland Chapter on Digital Communication Systems, pp. 135-139 (1986). cited by other .
Malek, Manu, "Integrated Voice and Data Communications Overview", IEEE Communications Magazine, 26,(6):5-15 (1988). cited by other .
Malyat et al., "Using Reservation-Virtual-Time CSMA with Split-Channel Reservation for Packet Voice and Data", pp. 209-212 (date unknown). cited by other .
McBain, Alistiar, "The GEC 420 DataPhone--terminal for the ISDN", Computer Communications, 11(1):32-36 (1988). cited by other .
McCanne, Steven, "A Distributed Whiteboard for Network Conferencing", Computer Science Division, Electrical Engineering and Computer Science, University of California, Berkeley, 13 pages (1992). cited by other .
McNamara, R.P., "METRONET: An Overview of CATV Regional Data Network", Eascon '81: IEEE Electronics and Aerospace Systems Conventions, Nov. 16-19, 1981, Washington, D.C., pp. 249-253 (1981). cited by other .
McNamara, R.P., "Some Considerations of the Voice-Data Capabilities of Broadband Cable Networks", Compcon 82: Twenty-Fourth IEEE Computer Society International Conference, Feb. 22-25, 1982, San Francisco, California, digest of papers, pp. 312-314 (1982). cited by other .
Mednick, Ralph, "Office Information Network: An Integrated LAN", Globecom '85: IEEE Global Telecommunications Conference, Dec. 2-5, 1985, New Orleans, Louisiana, Conference Record, vol. 1 of 3, pp. 451-456 (1985). cited by other .
Mera et al., "Multimedia Communication Terminal for Broadband Network", Globecom '87, Nov. 15-18, 1987, Tokyo, Japan, pp. 19.2.1-19.2.5 (1987). cited by other .
Merritt, Ian H., "Providing Telephone Line Access to a Packet Voice Network", Information Sciences Institute, ISI/RR-83-107, 15 pages (1983). cited by other .
Metcalfe et al., "Ethernet: Distributed Packet Switching for Local Computer Networks", reprinted from Communications of the ACM, 19(5):395-404 (1976). cited by other .
Mills et al., "A Data and Voice System for the General Service Telephone Network", Proceedings of the International Conference on Industrial Electronics, Control, and Instrumentation, pp. 1143-1148 (Nov. 1987). cited by other .
Mochizuki et al., "Digital Videotex System", Review of the Electrical Communication Laboratories, 32(6):1044-1050 (1984). cited by other .
Montgomery, Warren A., "Techniques for Packet Voice Synchronization", IEEE Journal on Selected Areas in Communications, SAC-1(6):1022-1028 (1983). cited by other .
Morikawa et al., "Hardware Design of Integrated Centralized Extension System", Review of the Electrical Communication Laboratories, 32(2):325-332 (1984). cited by other .
Mowafi et al., "Integrated Voice/Data Packet Switching Techniques for Future Military Networks", IEEE Transactions on Communications, COM-28(9):1655-1662 (1980). cited by other .
Muraoka et al., "Communication Processing Services in INS Model System", Proceedings of the Seventh International Conference on Computer Communication, Sydney, Australia, Oct. 30-Nov. 2, 1984, pp. 66-71 (1984). cited by other .
Musser et al., "A Local Area Network as a Telephone Local Subscriber Loop", IEEE Journal on Selected Areas in Communications, SAC-1(6):1046-1069 (1983). cited by other .
Nash et al., "Simultaneous Transmission of Speech and Data over an Analog Telephone Channel", CH2190-7/85/0000-0094, IEEE, pp. 4.2.1-4.2.4 (1985). cited by other .
Newman, Bill, "An ISDN Data and Voice Terminal Based on a Personal Computer", Globecom '85: IEEE Global Telecommunications Conference, Dec. 2-5, 1985, New Orleans, Louisiana, Conference Record, vol. 3 of 3, pp. 1048-1052 (1985). cited by other .
Nicolaou, Cosmos, "An Architecture for Real-Time Multimedia Communication Systems", IEEE Journal on Selected Areas in Communications, 8(3):391-400 (1990). cited by other .
Noerpel et al., "Radio Link Access Procedure for a Wireless Access Communications System", IEEE, 2:1111-1115 (1994). cited by other .
Nojima et al., "Integrated Services Packet Network Using Bus Matrix Switch", IEEE Journal on Selected Areas in Communications, SAC-5(8):1284-1292 (1987). cited by other .
Nussbaum, Eric, "Communications Network Needs and Technologies--A Place for Photonic Switching?", EH0301-2/91/0000/0441, IEEE, pp. 441-448 (1988). cited by other .
O'Leary, G.C., "Local Access Area Facilities for Packet Voice", Proceedings of the Fifth International Conference on Computer Communication, Oct. 27-30, 1980, Atlanta, Georgia, pp. 281-286 (1980). cited by other .
O'Leary et al., "A modular approach to packet voice terminal hardware design", 1981 National Computer Conference, May 4-7, 1981, Chicago Illinois, Conference Proceedings, pp. 183-189 (1981). cited by other .
Ohmori et al., "Facsimile Intelligent Communication System, Present and Future", Review of the Electrical Communication Laboratories, 31(4):463 (1983). cited by other .
Oltmanns et al., "A Voice and Communications System for the IBM PC", Speech Technology, pp. 94-99 (Mar./Apr. 1986). cited by other .
Opderbeck, H., "Throughput Degradations for Single Packet Messages", UCLA-NMC, Network Working Group, 6 pages (dated 1974). cited by other .
Palmer et al., "DECspin: A Networked Desktop Videoconferencing Application", Digital Technical Journal, 5(2):1-13 (1993). cited by other .
Palmer et al., "Desktop Meeting", LAN Magazine, 6(11):111-121 (1991). cited by other .
Pasquale et al., "Operating System and Window System Research for Distributed Multi-Media Applications: A Status Report", UCSD Technical Report CS90-176, Department of Computer Science and Engineering, University of California, San Diego, 17 pages (dated 1990). cited by other .
Pennebaker et al., "A High Performance Freeze Frame Videoconferencing System", IBM T. J. Watson Research Center, 8 pages (date unknown). cited by other .
Perkins, Roderick E., "Spider: an Investigation in Collaborative Technologies and their Effects on Network Performance", Globecom '91: IEEE Global Telecommunications Conference, Dec. 2-5, 1991, Phoenix, Arizona, Conference Record, vol. 3 of 3, pp. 2074-2080 (1991). cited by other .
Perkins, D., "The Point-to-Point Protocol for the Transmission of Multi-Protocol Datagrams Over Point-to-Point Links", Network Working Group, 48 pages (dated 1990). cited by other .
Pierce, Jr., "Don't Just Tell Them, Show Them!", Automation (Aug. 1990). cited by other .
Product Catalog, title unknown, random collection of pages, 24 pages. cited by other .
Product Catalog, title unknown, random collection of pages, 100 pages. cited by other .
Radcliffe, "Video Calls Use Basic Phone Lines", MacWeek (Aug. 3, 1992). cited by other .
Rangan et al., "Software Architecture for Integration of Video Services in the Etherphone System", IEEE Journal on Selected Areas in Communications, 9(9):1395-1404 (1991). cited by other .
Raviraj et al., "An Integrated Services Local Area Network with Hybrid Access", IEEE Infocom '89: Eighth Annual Joint Conference of the IEEE Computer and Communications Societies, pp. 439-447 (1989). cited by other .
Reynolds et al., "Voice File Intechange Protocol (VFIP)", Network Working Group, 5 pages (dated 1986). cited by other .
Roberts, Lawrence G., "The Evolution of Packet Switching", Proceedings of the IEEE, 66(11):1307-1313 (1978). cited by other .
Rockwell Application Note, Order No. 873, (Oct. 1991). cited by other .
Romkey, J., "A Nonstandard for Transmission of IP Datagrams over Serial Lines: SLIP", Network Working Group, 6 pages (dated 1988). cited by other .
Rosner, Roy D., "Packet Switching: Tomorrow's Communications Today", Wadsworth, Inc., Belmont, California (1982). cited by other .
Ross et al., "Approaches to the Integration of Voice and Data Telecommunications", National Telecommunications Conference, Nov. 27-29, 1979, Washington, DC, 46.6.1-46.6.8 (1979). cited by other .
Ross, M.J., "Alternatives for Integrating Voice and Data", 1981 International Switching Symposium, Montreal, Sep. 21-25, 1981, 8 pages (1981). cited by other .
Rowe, J.W., "Service Protection in Managed Networks", Third IEE Conference on Telecommunications, The Institution of Electrical Engineers, pp. 296-300 (1991). cited by other .
Ruston et al., "Integrating Computing and Telecommunications: A tale of Two Architectures", pp. 57-68 (date unknown). cited by other .
Rybczynski et al., "Interworking Between Packet Networks and Integrated Voice/Data Access Systems", Links for the Future, IEEE/Elsiver Science Publishers B.V., North-Holland, pp. 1309-1312 (1984). cited by other .
Saito et al., "Voice Packet Communication System for Private Networks", Globecom '89: IEEE Global Telecommunications Conference & Exhibition, Nov. 27-30, 1989, Dallas Texas, Conference Record, vol. 3 of 3, pp. 1874-1878 (1989). cited by other .
Saltzberg, Burton R., "Theoretical Foundations of Digital Communication", pp. 1-48 (date unknown). cited by other .
Sammartino et al., "Desktop Multimedia Communications--Breaking the Chains", ICC '91, International Conference on Communications, Jun. 23-26, 1991, Denver, Colorado, Conference Record, vol. 1 of 3, pp. 0073-0077 (1991). cited by other .
Sarin, Sunil, "Computer-Based Real-Time Conferencing Systems", CH0018-9162/85/1000-0033, IEEE, pp. 33-44 (1985). cited by other .
Sasaki et al., "Variable rate voice coding system", MILCOM, 92(1):364-367 (Oct. 11, 1992). cited by other .
Saunders, Stephen, "Voice Compression: When Less Is More", Data Communications, pp. 58-64 (Jul. 1991). cited by other .
Schooler et al., "An Architecture for Multimedia Connection Management", Reprinted from the Proceedings IEEE 4.sup.th Comsoc International Workshop on Multimedia Communications, pp. 271-274, Monterey, CA (Apr. 1992), ISI Reprint Series ISI/RS-92-294, 6 pages (1992). cited by other .
Schooler, Eve M., "Case Study: Multimedia Conference Control in a Packet-switched Teleconferencing System", Reprinted from Journal of Internetworking: Research and Experience, 4(2):99-120 (1993). cited by other .
Schooler, Eve M., "The Connection Control Protocol: Specification", USC/Information Sciences Institute, 30 pages (dated 1992). cited by other .
Schooler, Eve M., "A Distributed Architecture for Multimedia Conference Control", ISI Research Report, pp. 1-18 (1991). cited by other .
Schooler et al., "Multimedia Conferencing: Has it come of age?", Information Sciences Institute, ISI/RS-91-286, 10 pages (1991). cited by other .
Schooler et al., "A Packet-switched Multimedia Conferencing System", Information Sciences Institute, University of Southern California, pp. 12-22 (date unknown). cited by other .
Schulzrinne, Henning, "A Transport Protocol for Audio and Video Conferences and other Multiparticipant Real-Time Applications", Dept. of Computer Science, University of Massachusetts at Amherst, 21 pages (dated 1992). cited by other .
Schulzrinne, Henning, "Voice Communication Across the Internet: A Network Voice Terminal", Dept. of Computer Science, University of Massachusetts at Amherst, 35 pages (dated 1992). cited by other .
Schwartz, Mischa, "Telecommunication Networks: Protocols Modeling and Analysis", Addison-Wesley Publishing Company, Menlo Park, California (1987). cited by other .
Scott et al., "Communications Support for Multimedia Workstations", Third IEE Conference on Telecommunications, Institution of Electrical Engineers, London, pp. 67-72 (1991). cited by other .
Seveque et al., "In-House ISDN Applications", Electrical Communication, 63(1):17-24 (1989). cited by other .
Shacham et al., "Speech Transport in Packet-Radio Networks with Mobile Nodes", IEEE Journal on Selected Areas in Communications, SAC-1(6):1084-1097 (1983). cited by other .
Sharrock et al., "A Framed, Movable-Boundary Protocol for Integrated Voice/Data in A LAN", ACM 0-89791-201-2/86/0800-0111, IEEE, pp. 111-119 (1986). cited by other .
Shepherd et al., "Protocol Support for Distributed Multimedia Applications", Presented at Second International Workshop on Network and Operating System Support for Digital Audio and Video, IBM ENC, Heidelberg, Germany, Nov. 18-19, 1991, 13 pages (1991). cited by other .
Sherman, Ken, "Data Communications: A User's Guide", Prentice-Hall, Inc., Englewood Cliffs, New Jersey, pp. 296-407 (1990). cited by other .
Sheih et al., "Decentralized token-CSMA/CD protocol for integrated voice/data LANs", Computer Communications, 14(4):223-230 (1991). cited by other .
Shoch, John, "Carrying Voice Traffic Through an Ethernet Local Network: A General Overview", Local Networks for Computer Communications, North-Holland Publishing Company, Amsterdam, pp. 429-446 (1981). cited by other .
Sipos, E., "Communication System", IBM Technical Disclosure Bulletin, 12(11):2029-2030 (1970). cited by other .
Smith, Tom, "ACT to intro sub T-1 voice/data/fax mux; SDM-T will support data rates up to 64K bit/sec and compress voice signals by as much as 8-to-1", Network World, p. 31 (Apr. 16, 1990). cited by other .
Smith et al., "Voice/Data Concentrator", IBM Technical Disclosure Bulletin, 21(8):3066-3067 (1979). cited by other .
Smith, "Wrist TVs Still Fiction, but Police Videophones Take Hold", TE&M (Dec. 15, 1987). cited by other .
Sohraby et al., "ISDN Primary Rate Interface Impact on Performance on Integrated on CSMA/CD Networks--A measurement and Simulation Study", Globecom '90: IEEE Telecommunications Conference & Exhibition, Dec. 2-5, 1990, San Diego, California, Conference Record, vol. 2 of 3, pp. 0912-0919 (1990). cited by other .
Sparrell, Duncan, "Wideband Packet Technology", IEEE Global Telecommunications Conference & Exhibition, Nov. 28-Dec. 1, 1988, Hollywood, Florida, Conference Record, vol. 3 of 3, pp. 1612-1616 (1988). cited by other .
Spilling et al., "Digital Voice Communications in the Packet Radio Network", ICC '80: International Conference on Communications, Jun. 8-12, 1980, Seattle, Washington, Conference Record, vol. 1 of 3, 21.4.1-21.4.7 (1980). cited by other .
Spragins et al., "Telecommunications--Protocols and Design", Communications Equipment Interface Design, Chapter 4, Addison-Wesley Publishing Company, pp. 156, 304, and 327 (1991). cited by other .
Sriram et al., "Discrete-Time Analysis of Integrated Voice/Data Multiplexers With and Without Speech Activity Detectors", IEEE Journal on Selected Areas in Communications, SAC-1(6):1124-1132 (1983). cited by other .
Stallings, William, "ISDN: An Introduction", Macmillan Publishing Company, New York, 3 pages (1989). cited by other .
Stallings, "Protocols and Architecture", Data and Computer Communications, 2.sup.nd Edition, pp. 100-103 (1988). cited by other .
Stern, H. P., "An Improved Integrated Voice/Data Mobile Radio System", CH2846-4/90/0000-0311, IEEE, pp. 311-316 (1990). cited by other .
Stern, H.P., "Design and Performance Analysis of an Integrated Voice/Data Mobile Radio System", CH2682-3/89/0000-0433, IEEE, pp. 0433-0437 (1989). cited by other .
Stevens, Jim, "Much More Idle Chatter About Reference Models", 9 pages (dated 1987). cited by other .
Sudhaker et al., "Design and Performance Evaluation Considerations of a Multimedia Medical Database", IEEE Transactions on Knowledge and Data Engineering, 5(5):888-894 (1993). cited by other .
Sunshine et al., "Broad-Band Personal Computer LAN's", IEEE Journal on Selected Areas in Communications, SAC-3(3):408-415 (1985). cited by other .
Swinehart, Daniel C., "Telephone Management in the Etherphone System", Globecom '87, Nov. 15-18, 1987, Tokyo, Japan, pp. 30.3.1-30.3.5 (1987). cited by other .
Taka et al., "Low-Bit-Rate Speech Codecs and Wideband Speech Codecs", Review of the Electrical Communications Laboratories, 36(1):49-55 (1988). cited by other .
Takase et al., "An Experimental B-ISDN System for Man Application Study", Globecom '91: Global Telecommunications Conference, Dec. 2-5, 1991, Phoenix, Arizona, Conference Record, vol. 3 of 3, pp. 2100-2104 (1991). cited by other .
Tamamura et al., "Picture and Audio Information Retrieval Service Using 65Kb/s Digital Network", Globecom '84: IEEE Global Telecommunications Conference, Nov. 26-29, 1984, Atlanta, Georgia, Conference Record, vol. 1 of 3, pp. 323-327 (1984). cited by other .
Tani et al., "High-speed Multimedia Communication System on A Wideband LAN", Globecom '87, Nov. 15-18, 1987, Tokyo, Japan, pp. 577-581 (1987). cited by other .
Terek et al., "Experiences with Audio Conferencing Using the X Window System, UNIX, and TCP/IP", Proceedings of the Summer 1991 Usenix Conference, pp. 405-418 (1991). cited by other .
Thompson, Geoff, "RE: 802.3 Full Duplex vs. CSMA/CD", retrieved from <http://www.hep.net/mail/802exec/tojul97/0334.html>, 5 pages (dated 1996). cited by other .
Tobagi et al., "Expressnet: A High-Performance Integrated-Services Local Area Network", IEEE Journal on Selected Areas in Communications, SAC-1(5):898-913 (1983). cited by other .
Tomcik et al., "Facsimile Compression Through Demodulation in Packet Networks", Globecom '92: IEEE Global Telecommunications Conference, Dec. 6-9, 1992, Orlando, Florida, Conference Record, vol. 3 of 3, pp. 1699-1703 (1992). cited by other .
Tominaga et al., "Comparison of Coding Schemes for Telewriting Systems", ICCC(1986). cited by other .
Tomsu et al., "ITT 5200 BCS, Digital Integrated Voice and Data Communication System", CH2175-8/85/0000-0855, IEEE, pp. 855-862 (1985). cited by other .
Tsuda et al., "An Approach to Muti-Service Subscriber Loop System Using Packetized Voice/Data Terminals", The International Symposium on Subscriber Loops and Services, Mar. 20-24, 1978, Georgia Institute of Technology, Atlanta, Georgia, IEEE cat. No. 78CH1279-9 COM, pp. 161-165 (1978). cited by other .
Tsuruta et al., "Intelligent Communication Terminal for Integrating Voice, Data and Video Signals", ICC '86: IEEE International Conference on Communications '86, Jun. 22-25, 1986, Toronto, Canada, Conference Record, vol. 3 of 3, pp. 1509-1513 (1986). cited by other .
Turner, Jonathan S., "Design of an Integrated Services Packet Network", IEEE Journal on Selected Areas in Communications, SAC-4(3):1373-1380 (1986). cited by other .
U.S. West Caller ID, 1 p. (Jul. 18, 1994). cited by other .
Ueda et al., "Evaluation of an Experimental Packetized Speech and Data Transmission System", IEEE Journal on Selcted Areas in Communications, SAC-1(6):1039-1045 (1983). cited by other .
Ulug et al., "Statistical Multiplexing of Data and Encoded Voice in a Transparent Intelligent Network", Carleton University, Ottawa, Canada, 7 pages (date unknown). cited by other .
Unagami, Shigeyuki, "Prospects of High-Speed Voice Band Modems", Technical Survey, 73(10):1082-1086 (1990). cited by other .
Vakulenko et al., "The UPS-2.4 Modem", Originally published in Electrosryz, No. 3, 1989, pp. 45-49, Scripts Technica, Inc., pp. 23-27 (1990). cited by other .
Vin et al., "Hierarchical Conferencing Architectures for Inter-Group Multimedia Collaboration", Proceedings of the ACM Conference on Organizational Computing Systems, Atlanta, Georgia, Nov. 5-8, 1991, 12 pages (1991). cited by other .
Vin et al., "Multimedia Conferencing in the Etherphone Environment", Computer, 24(10):69-79 (1991). cited by other .
Wakerly, John, "A Voice/Data/Packet Switching Architecture", CH2135-2/85/0000/CT94301, IEEE, pp. 194-199 (1985). cited by other .
Wang et al., "Priority Statistical Multiplexer Design for SNA*/SDLC Access to a Virtual Circuit Packet Network", AT&T Technical Journal, 67(6):69-86 (1988). cited by other .
Wee, Jean, "Coming to S'pore: worldwide face-to-face contact through PCs", Business Times, 1 page (Mar. 23, 1991). cited by other .
Weinstein et al., "Experience with Speech Communication in Packet Networks", IEEE Journal on Selected Areas in Communications, SAC-1(6):963-980 (1983). cited by other .
Weiss, Christoph, "Connection of Video Communication Equipment to a Broadband Network", CH2298-9/86/0000-1428, IEEE, pp. 1428-1433 (1986). cited by other .
Weiss, Christoph, "Desk Top Video Conferencing --An Important Feature of future Visual Communications", ICC '90: IEEE International Conference on Communications, Conference Record, vol. 1 of 4, pp. 0134-0139 (1990). cited by other .
White, Patrick E., "The Role of the Broadband Integrated Services Digital Network", IEEE Communications Magazine, 29(3):116-119 (1991). cited by other .
Wieselthier et al., "A Movable-Boundary Channel-Access Scheme for Integrated Voice/Data Networks", IEEE Infocom '91: Tenth Annual Joint Conference of the IEEE Computer and Communications Societies, Apr. 7-11, 1991, Bal Harbour, Florida, pp. 0721-0731 (1991). cited by other .
Williams et al., "Maximum Utilisation of an Existing Optical Trunk Network", Third IEE Conference on Telecommunications, The Institution of electrical Engineers, pp. 202-295 (1991). cited by other .
Wong et al., "An Integrated Services Token-Controlled Ring Network", IEEE Journal on Selected Areas in Communications, 7:670-678 (1989). cited by other .
Wu et al., "Capacity Allocation for Integrated Voice/Data Transmission at a Packet Switched TDM", IEEE Transactions on Communications, 40(6):1059-1069 (1992). cited by other .
Yamada et al., "Multimedia High-Throughput X.25 Packet Switching System", NTT Review, 1(2):82-88 (1989). cited by other .
Yashiro et al., "High-Speed High-Throughput Packet Switching System for Multimedia Communications", IEEE Global Telecommunications Conference & Exhibition, Nov. 28-Dec. 1, 1988, Hollywood, Florida, Conference Record, vol. 1 of 3, pp. 0435-0439 (1988). cited by other .
Zellweger et al., "An Overview of the Etherphone System and its Applications", 2.sup.nd IEEE Conference on Computer Workstations, Mar. 7-10, 1988, Santa Clara, California, pp. 160-168 (1988). cited by other .
Ziegler et al., "Mechanisms for Integrated Voice and Data Conferencing", ACM 089791-405-8/90/0009/0101, IEEE, pp. 101-107 (1990). cited by other .
Ziegler et al., "Multimedia Conferencing on Local Area Networks", Computer, 23(9):52-61 (1990). cited by other .
Claiborne, "Natural MicroSystems Corp. Watson 1200; Hardware Review; one of three evaluations of voice-mail expansion boards in `Voice-mail boards link telephone and PC.`; evaluation", PC Week, 7(40):124 (1990). cited by other .
Cylink Secured Wireless Network. Cylink/Micom at Networld. Illustration. Cyclink Booth No. 3574. Micom Booth No. 1839. 1 page. Undated and Labeled MT46808. cited by other .
Data/Voice Internetwork Node. "The Integrated Network". Micom. 6 Pages. Undated and Labeled MT46900-905. cited by other .
Lewis James Enterprises, Nov. 29, 1993. Illustration. 1 page. Labeled MT46817. cited by other .
Micom Letter (Rev. 10/90). 1 Page. Undated and Labeled MT69478. cited by other .
Micom Product Information Sheets. 2 Pages. Undated and Labeled MT69479-80. cited by other .
Micom Product Information Sheets. 6 Pages. Undated and Labeled MT69481-86. cited by other .
Micom Product Information Sheets. 2 Pages. Undated and Labeled MT69487-88. cited by other .
Micom Product Information Sheets. 6 Pages. Undated and Labeled MT46877-82. cited by other .
Micom Communications Corp. Corporate Background. 6 pages. Undated and Labeled MT46779-84. cited by other .
Micom Customer Profile, "Continental Cablevision Tunes Into Cost-Savings with Marathon". 2 pages. Undated and Labeled MT69551-52. cited by other .
Micom Customer Profile, "Switch to Marathon Saves Thousands for Sager Electrical Supply Co.", 2 pages. Undated and Labeled MT69553-54. cited by other .
Micom Customer Profile, "Visone Motors Revs Up with Marathon". 2 pages. Undated and Labeled MT69555-56. cited by other .
Micom Brochure. "Warning: There May be an Embezzler in your Company!" Get Marathon and Stop Losing Money. 2 pages. Undated and Labeled MT46805-06. cited by other .
Micom Brochure "Hotline Application Assistance" . 4 pages. Undated and Labeled MT69543-46. cited by other .
Micom Reports on Data/VOice Products, Trends and Applications, Networking News, Nov. 1993. "Micom Announces New DVI Products for Multi-Site Networking", pp. 1-8. Labeled MT46791-800. cited by other .
Micom News Release, Sep. 25, 1990 Simi Valley, CA. "Micom Announces the Marathon 5K for Long Distance Networking", 4 pages. Labeled MT69474-77. cited by other .
Micom News Release, Oct. 5, 1993. Live at Networld Dallas, Booth #1839. "Multi-Site Integrated Internetworking for Seamless LAN/VOICE/FAX Connectivity, and Industry First". 3 pgs. Labeled MT69518-20. cited by other .
Micom News. Up-Close with Pete Oliver of PremierePage. Network. Mar./Apr. 1994. 4 pages. Labeled MT69513-16. cited by other .
Micom Product Sheet. Marathon 1K "Data/Voice Network Feeder". 1 page. Undated and Labeled MT69512. cited by other .
Micom Brochure. In the Last Year, Companies like your have saved millions of dollars. . . . Find out How Network Integratioon can Save you Big Money. 3 pages. Undated and Labeled MT69509-11. cited by other .
Micom Brochure. "Run Our Marathon" Marathon 5K.A Supplement to the Oct., 1990 Issue. 3 pages. Labeled MT46747-49. cited by other .
Micom Brochure-Marathon 5K. Locateions. 11 pages. Undated and Labeled MT69463-73. cited by other .
Micom Brochure. "Netrunner" 10 pages. Undated and Labeled MT69450, MT69453-61. cited by other .
Micom Brochure. "Cell Relay for Sub T1/E1" Data, Voice, Fax and Lan Integration. 2 pages. Undated and Labeled MT69597-98. cited by other .
Micom Brochure "Micom Digital Direct Series T1", Marathon and NetRunner Integration Products. 4 pages. Undated and Labeled MT69591-94. cited by other .
Micom Brochure, "Software Release 5.0" Voice, Fax, Data, LAN Integration. 2 pages. Undated and Labeled MT69595-96. cited by other .
Micom Brochure, "NetRunner Integration Routers" The Next Step in Multiprotocol Routing. 17 Pages. Undated and Labeled MT69576-601. cited by other .
Micom General Price List Domestic, Dec. 1993. Labeled MT47070-88. cited by other .
Micom- Marathon-Remote LAN Bridge User's Manual. Model 5000RLB/3, Stock No. 800-1768-30. Rev. A. Jun. 1993. Labeled MT68881-69000. cited by other .
Micom Brochure. A New Era of Wide Area Networking. "Free Voice and Fax Over Low-Cost Wans", 11 pages, Undated and Labeled MT69850-60. cited by other .
Micom Applications Guide. "Integrated Networking Over Low Cost Wans". 1991 Micom Communications Corporation. Labeled MT69774-804. cited by other .
Micom Brochure. "Voice over Frame Relay", 4 pages. Undated and Labeled MT69805-08. cited by other .
Micom. Marathon. Voice/Fax Module. Users Manual. Stock Number 800-1747, Rev. C. Dec. 1991. MT69359-448. cited by other .
Micon. M arathon. A Data/Voice Network Server. Network Management System Module User's Manual. Stock Number 800-1748, Rev. C. Dec. 1991. Labeled MT69287-332. cited by other .
Micom. Marathon 5K. A Data/Voice Network Server. Startup Guide. Stock Number 800-1745, Rev. B. Dec. 1991. Labeled MT64696-64793. cited by other .
Micom. Marathon 5K. A Data Voice Network Server. Users Manual. Stock Number 800-1705, Rev. A. Jan. 1991. Labeled MT69001-69282. cited by other .
Product Watch, "Micom Leads in the Multiport Bridging Race", 16. HP Professional. 1 page. Undated and Labeled MT46656. cited by other .
Netrunner Shoots High and Low. LAN Computing, Sep. 16, 1993, 1 page. Labeled MT-69602. cited by other .
Busse. "Marathon Integrates Traffic" Voice, fax, and LAN data can be sent over one line. Apr. 19, 1993 Infoworld. Labeled MT06141. cited by other .
Csenger. Network World. "Miicron boosts voice compression, adds fax feature to muxes". Nov. 7, 1994. 1 page. Labeled MT46914. cited by other .
Feltman. "A Free Ride for Data". Apr. 1994 LAN Magazine. pp. 67-77 Labeled MT69895. cited by other .
Guy. "Rapid Relay Technology". The Foundation of MICOM's Data/Voice Network Servers and Data/Voice Internetwork Nodes. Undated and Labeled MT69809-41. cited by other .
Guy. "Supplemental Technology Overview". MICOM Communications Corp. Simi Valley, CA. Undated and Labeled MT46725-35. cited by other .
Guy. "Fast Packet Multiplexing: A Technical Overview". MICOM Communications Corp. Simi Valley, CA. Undated and Labeled MT46708-16. cited by other .
Newton. Brochures. "Every Man's Network", including "The Network Goes Digital", "Understanding the New Networking", "Some Case Studies", Commonly Asked Networking Questions, "How to Manage a Network", "A Networking Glossary". pp. 4-19. Undated and Labeled MT69492-507. cited by other .
Phillips. "Branch-Office Muxes Make every Kbit Count". Multiplexers. Data Communications, Nov. 21, 1990. pp. 75-80. Labeled MT69882-887. cited by other.

Primary Examiner: Ton; Dang
Attorney, Agent or Firm: Mueting Raasch & Gebhardt, P.A.
Parent Case Text


REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. patent application Ser. No. 09/400,607 filed Sep. 20, 1999 (pending) which is a continuation of U.S. patent application Ser. No. 08/338,340 filed Nov. 10, 1994 (issued as U.S. Pat. No. 6,009,082) which is a continuation in part of U.S. patent application Ser. No. 08/002,467 filed Jan. 8, 1993 (issued as U.S. Pat. No. 5,452,289), wherein the complete applications of which, including any microfiche appendix, are incorporated herein by reference.
Claims


What is claimed is:

1. A communication apparatus comprising: a modem; one or more storage devices operable to store outgoing digital audio information and store other multi-media data components; and processing and control circuitry operable under control of a user interface program, wherein the processing and control circuitry is operable to: receive stored outgoing digital audio information, combine the stored outgoing digital audio information with one or more other stored multi-media data components resulting in combined outgoing multi-media, and provide the combined outgoing multi-media to the line interface for communication to a remote location via the modem for connection to at least one packet switched network.

2. The communication apparatus of claim 1, wherein the communication apparatus further comprises a voice interface apparatus for use in generating local analog voice signals from a local user representative of a voice message and for use in conveying audio signals to the local user, and wherein the processing and control circuitry is further operable to convert the local analog voice signals into outgoing digital voice information representative of the voice message, wherein the outgoing digital voice information representative of the voice message is provided for storage as the outgoing digital audio information, and further wherein the processing and control circuitry is operable to combine the stored outgoing digital voice information representative of the voice message with one or more of the other stored multi-media data components resulting in the combined outgoing multi-media for communication via the modem to the remote location.

3. The communication apparatus of claim 2, wherein the processing and control circuitry is further operable to compress the outgoing digital voice information representative of the voice message, wherein the compressed outgoing digital voice information representative of the voice message is provided for storage as the outgoing digital audio information, and further wherein the processing and control circuitry is operable to combine the stored compressed outgoing digital voice information representative of the voice message with one or more of the other stored multi-media data components resulting in the combined outgoing multi-media for communication via the modem to a remote location.

4. The communication apparatus of claim 1, wherein the one or more other stored multi-media data components comprise at least one of textual information and graphical information.

5. The communication apparatus of claim 4, wherein the graphical information comprises at least one of picture information and video information.

6. The communication apparatus of claim 1, wherein the outgoing digital audio information comprises at least one file of outgoing digital audio information.

7. The communication apparatus of claim 6, wherein the outgoing digital audio information comprises at least one file of outgoing compressed digital voice information.

8. The communication apparatus of claim 1, wherein the processing and control circuitry is operable to provide a display interface to a user comprising user selectable functions to allow the user to control the creation of the combined outgoing multi-media.

9. The communication apparatus of claim 8, wherein the user selectable functions comprise at least one of a composing function, a viewing function, an editing function, a playing function, a recording function, and a deleting function.

10. The communication apparatus of claim 1, wherein the processing and control circuitry is operable to provide a display interface to a user comprising user selectable functions to allow the user to select a destination for the combined outgoing multi-media.

11. The communication apparatus of claim 1, wherein the processing and control circuitry is farther operable to: receive incoming multi-media comprising incoming digital audio information and one or more other incoming multi-media data components via the modem; and store the incoming digital audio information and the one or more other incoming multi-media data components of the incoming multi-media in the one or more storage devices.

12. The communication apparatus of claim 11, wherein the processing and control circuitry is further operable for use in conveying the incoming multi-media to a local user.

13. The communication apparatus of claim 12, wherein conveying the incoming multi-media to the local user comprises displaying at least one of the one or more other incoming multi-media data components of the incoming multi-media to the local user.

14. The communication apparatus of claim 12, wherein conveying the incoming multi-media to the local user comprises converting the incoming digital audio information of the incoming multi-media to local analog audio signals for conveyance to the local user.

15. The communication apparatus of claim 12, wherein the processing and control circuitry is further operable for use in conveying at least one of the incoming digital audio information and the one or more other incoming multi-media data components of the incoming multi-media to another remote location.

16. The communication apparatus of claim 11, wherein the processing and control circuitry is further operable for use in modifying at least a portion of at least one of the incoming digital audio information and the one or more other incoming multi-media data components of the incoming multi-media.

17. The communication apparatus of claim 11, wherein the one or more other incoming multi-media data components comprise at least one of textual information and graphical information.

18. The communication apparatus of claim 17, wherein the graphical information comprises at least one of picture information and video information.

19. The communication apparatus of claim 11, wherein the incoming digital audio information comprises at least one file of incoming digital audio information.

20. The communication apparatus of claim 11, wherein the incoming digital audio information comprises at least incoming compressed digital voice information.

21. The communication apparatus of claim 11, wherein the processing and control circuitry is operable to provide a display interface to a user to notify the user of the receipt of the incoming multi-media.

22. The communication apparatus of claim 11, wherein the processing and control circuitry is operable to provide a display interface to a user comprising user selectable functions to allow the user to manipulate the received incoming multi-media.

23. The communication apparatus of claim 22, wherein the user selectable functions comprise at least one of a forwarding function, a viewing function, an editing function, a playing function, a recording function, a storing function, and a deleting function.

24. The communication apparatus of claim 1, wherein the modem is operable for full duplex communication.

25. A communication method implemented under control of a user interface program, the method comprising: providing a modem; providing stored outgoing digital audio information; combining, under control of a local user via input to the user interface program, the stored outgoing digital audio information with one or more other stored multi-media data components resulting in combined outgoing multi-media; and providing the combined outgoing multi-media for communication to a remote location via the modem for connection to at least one packet switched network.

26. The communication method of claim 25, wherein providing stored outgoing digital audio information comprises: generating local analog voice signals from a local user representative of a voice message; and converting the local analog voice signals into outgoing digital voice information representative of the voice message, wherein the outgoing digital voice information representative of the voice message is combined with one or more of the other stored multi-media data components resulting in the combined outgoing multi-media for communication to the remote location.

27. The communication method of claim 26, wherein providing stored outgoing digital audio information further comprises compressing the outgoing digital voice information representative of the voice message, wherein the compressed outgoing digital voice information representative of the voice message is combined with one or more of the other stored multi-media data components resulting in the combined outgoing multi-media for communication to the remote location.

28. The communication method of claim 25, wherein the one or more other stored multi-media data components comprise at least one of textual information and graphical information.

29. The communication method of claim 28, wherein the graphical information comprises at least one of picture information and video information.

30. The communication method of claim 25, wherein the outgoing digital audio information comprises at least one file of outgoing digital audio information.

31. The communication method of claim 30, wherein the outgoing digital audio information comprises at least one file of outgoing compressed digital voice information.

32. The communication method of claim 25, wherein the communication method further comprises providing a display interface to a user comprising user selectable functions to allow the user to control the creation of the combined outgoing multi-media.

33. The communication method of claim 32, wherein the user selectable functions comprise at least one of a composing function, a viewing function, an editing function, a playing function, a recording function, and a deleting function.

34. The communication method of claim 25, wherein the communication method further comprises providing a display interface to a user comprising user selectable functions to allow the user to select a destination for the combined outgoing multi-media.

35. The communication method of claim 25, wherein the communication method further comprises: receiving incoming multi-media comprising incoming digital audio information and one or more other incoming multi-media data components; and storing the incoming digital audio information and the one or more other incoming multi-media data components of the incoming multi-media.

36. The communication method of claim 35, wherein the communication method further comprises conveying the incoming multi-media to the local user.

37. The communication method of claim 36, wherein conveying the incoming multi-media to the local user comprises displaying at least one of the one or more other incoming multi-media data components of the incoming multi-media to the local user.

38. The communication method of claim 36, wherein conveying the incoming multi-media to the local user comprises converting the incoming digital audio information of the incoming multi-media to local analog audio signals for conveyance to the local user.

39. The communication method of claim 35, wherein the communication method further comprises conveying at least one of the incoming digital audio information and the one or more other incoming multi-media data components of the incoming multi-media to another remote location.

40. The communication method of claim 35, wherein the communication method further comprises modifying at least a portion of at least one of the incoming digital audio information and the one or more other incoming multi-media data components of the incoming multi-media.

41. The communication method of claim 35, wherein the one or more other incoming multi-media data components comprise at least one of textual information and graphical information.

42. The communication method of claim 41, wherein the graphical information comprises at least one of picture information and video information.

43. The communication method of claim 35, wherein the incoming digital audio information comprises at least one file of incoming digital audio information.

44. The communication method of claim 35, wherein the incoming digital audio information comprises at least incoming compressed digital voice information.

45. The communication method of claim 35, wherein the communication method further comprises providing a display interface to a user to notify the user of the receipt of the incoming multi-media.

46. The communication method of claim 35, wherein the communication method further comprises providing a display interface to a user comprising user selectable functions to allow the user to manipulate the received incoming multi-media.

47. The communication method of claim 46, wherein the us a selectable functions comprise at least one of a forwarding function, a viewing function, an editing function, a playing function, a recording function, a storing function, and a deleting function.

48. A communication apparatus comprising: a modem; one or more storage devices operable to store one or more outgoing multi-media data components, wherein each of the outgoing multi-media data components comprise one or more types of information, and further wherein the one or more types of information comprise at least one of textual information, graphical information, and audio information; and processing and control circuitry operable under control of a user interface program, wherein the processing and control circuitry is operable to: receive two or more stored outgoing multi-media data components, at least one of the two or more stored outgoing multi-media data components comprising a type of information different than another of the two or more stored outgoing multi-media data components, combine the two or more stored outgoing multi-media data components resulting in combined out going multi-media, and provide the combined outgoing multi-media to a remote location via the modem for connection to at least one packet switched network.

49. The communication apparatus of claim 48, wherein the communication apparatus further comprises an audio interface apparatus for use in generating local analog audio signals and for use in conveying audio signals to a local user, and wherein the processing and control circuitry is further operable to convert the local analog audio signals into outgoing digital audio information, wherein the outgoing digital audio information is provided fur storage as one of the two or more stored outgoing multi-media data components.

50. The communication apparatus of claim 49, wherein the audio interface apparatus comprises a voice interface apparatus for use in generating local analog voice signals from a local user and for use in conveying voice signals w the local user, and wherein the processing and control circuitry is further operable to convert the local analog voice signals into outgoing digital voice information, wherein the outgoing digital voice information is provided for storage as one of the two or more stored outgoing multi-media data components.

51. The communication apparatus of claim 49, wherein flue processing and control circuitry is further operable to compress the outgoing digital audio information, wherein the compressed outgoing digital audio information is provided for storage as one of the two or more stored outgoing multi-media data components.

52. The communication apparatus of claim 48, wherein the graphical information comprises at least one of picture information and video information.

53. The communication apparatus of claim 48, wherein the processing and control circuitry is operable to provide a display interface to a user comprising user selectable functions to allow the user to control the creation of the combined outgoing multi-media.

54. The communication apparatus of claim 53, wherein the user selectable functions comprise at least one of a composing function, a viewing function, an editing function, a playing function, a recording function, and a deleting function.

55. The communication apparatus of claim 48, wherein the processing and control circuitry is operable to provide a display interface to a user comprising user selectable functions to allow the user to select a destination for the combined outgoing multi-media.

56. The communication apparatus of claim 48, wherein the processing and control circuitry is further operable to: receive incoming multi-media comprising two or more incoming multi-media data components via the line interface, wherein each of die incoming multi-media data components comprises one or more types of information, wherein the one or more types of information comprise at least one of textual information, graphical information, and audio information, and further wherein at least one of the two or more incoming multi-media data components comprises a type of information different than another of the two or more incoming multi-media data components; and store the two or more incoming multi-media data components of the incoming multi-media in the one or more storage devices.

57. The communication apparatus of claim 56, wherein the processing and control circuitry is further operable for use in conveying the incoming multi-media to a local user.

58. The communication apparatus of claim 57, wherein conveying the incoming multi-media to the local user comprises displaying at least one of the incoming multi-media data components of the incoming multi-media to the local user.

59. The communication apparatus of claim 57, wherein conveying the incoming multi-media components to the local user comprises converting at least one of the incoming multi-media data components of the incoming multi-media to local analog audio signals for conveyance to the local user.

60. The communication apparatus of claim 56, wherein the processing and control circuitry is further operable for use in conveying at least one of the incoming multi-media data components of the incoming multi-media to another remote location.

61. The communication apparatus of claim 56, wherein the processing and control circuitry is further operable for use in modifying at least a portion of at least one of the incoming multi-media data components of the incoming multi-media.

62. The communication apparatus of claim 56, wherein the graphical information comprises at least one of picture information and video information.

63. The communication apparatus of claim 56, wherein the audio information comprises at least compressed digital audio information.

64. The communication apparatus of claim 63, wherein the audio information comprises at least compressed digital voice information.

65. The communication apparatus of claim 56, wherein the processing and control circuitry is operable to provide a display interface to a user to notify the user of the receipt of the incoming multi-media.

66. The communication apparatus of claim 56, wherein the processing and control circuitry is operable to provide a display interface to a user comprising user selectable functions to allow the user to manipulate the received incoming multi-media.

67. The communication apparatus of claim 66, wherein the user selectable functions comprise at least one of a forwarding function, a viewing function, an editing function, a playing function, a recording function, a storing function, and a deleting function.

68. The communication apparatus of claim 48, wherein the modem is operable for full duplex communication.

69. A communication method implemented under control of a user interface program, the method comprising: providing a modem; providing one or more outgoing multi-media data components, wherein each of the outgoing multi-media data components comprises one or more types of information, and further wherein the one or more types of information comprise at least one of textual information, graphical information, and audio information; combining, under control of a local user via input to the user interface program, two or more outgoing multi-media data components resulting in combined outgoing multi-media, wherein at least one of the two or more outgoing multi-media data components comprises a type of information different than another of the two or more outgoing multi-media data components; and providing the combined outgoing multi-media for communication to a remote location via the modem for connection to at least one packet switched network.

70. The communication method of claim 69, wherein providing the one or more outgoing multi-media data components comprises: generating local analog audio signals; and converting the local analog audio signals into outgoing digital audio information, wherein the outgoing digital audio information is provided as at least one of the two or more outgoing multi-media data components.

71. The communication method of claim 70, wherein generating local analog audio signals comprises generating local analog voice signals from a local user, and further wherein converting the local analog audio signals into outgoing digital audio information comprises converting the local analog voice signals into outgoing digital voice in Formation, wherein the outgoing digital voice information is provided as at least one of the two or more outgoing multi-media data components.

72. The communication method of claim 70, wherein converting the local analog audio signals into outgoing digital audio information further comprises compressing the outgoing digital audio information, wherein the compressed outgoing digital audio information is provided as at least one of the two or more outgoing multi-media data components.

73. The communication method of claim 69, wherein the graphical information comprises at least one of picture information and video information.

74. The communication method of claim 69, wherein the communication method further comprises providing a display interface to a user comprising user selectable functions to allow the user to control the creation of the combined outgoing multi-media.

75. The communication method of claim 74, wherein the user selectable functions comprise at least one of a composing function, a viewing function, an editing function, a playing function, a recording function, and a deleting function.

76. The communication method of claim 69, wherein the communication method further comprises providing a display interface to a user comprising user selectable functions to allow the user to select a destination for the combined outgoing multi-media.

77. The communication method of claim 69, wherein the communication method further comprises: receiving incoming multi-media comprising two or more incoming multi-media data components, wherein each of the incoming multi-media data components comprises one or more types of information, wherein the one or more types of information comprise at least one of textual information, graphical information, and audio information, and further wherein at least one of the two or more incoming multi-media data components comprises a type of information different than another of the two or more incoming multi-media data components; and storing the two or more incoming multi-media data components of the incoming multi-media.

78. The communication method of claim 77, wherein the communication method further comprises conveying the incoming multi-media to a local user.

79. The communication method of claim 78, wherein conveying the incoming multi-media to the local user comprises displaying at least one of the incoming multi-media data components of the incoming multi-media to the local user.

80. The communication method of claim 78, wherein conveying the incoming multi-media to the local user comprises converting at least one of the incoming multi-media data components of the incoming multi-media to local analog audio signals for conveyance to the local user.

81. The communication method of claim 77, wherein the communication method further comprises conveying at least one of the incoming multi-media data components of the incoming multi-media to another remote location.

82. The communication method of claim 77, wherein the communication method further comprises modifying at least a portion of at least one of the incoming multi-media data components o the incoming multi-media.

83. The communication method of claim 77, wherein the graphical information comprises at least one of picture information and video information.

84. The communication method of claim 77, wherein the communication method further comprises providing a display interface to a user to notify the user of the receipt of the incoming multi-media.

85. The communication method of claim 77, wherein the communication method further comprises providing a display interface to a user comprising user selectable functions to allow the user to manipulate the received incoming multi-media.

86. The communication method of claim 85, wherein the user selectable functions comprise at least one of a forwarding function, a viewing function, an editing function, a playing function, a recording function, a storing function, and a deleting function.

87. A method for receiving information from a remote location, the method comprising: providing a modem; providing a personal computer, the modem operable under control of a graphical user interface associated with the personal computer; receiving a plurality of data packets via the modem from at least one packet switched network, wherein each of the plurality of data packets comprises one type of information, and further wherein each of the plurality of data packets comprises at least one header, the at least one header for use in routing the one type of information; removing and buffering the one type of information from each of the plurality of data packets; and delivering sequentially the one type of information from each of the plurality of data packets to an output device associated with the personal computer based on the at least one header for each of the plurality of data packets.

88. The method of claim 87, wherein delivering sequentially the one type of information to the output device occurs prior to completing receipt of the plurality of data packets.

89. The method of claim 87, wherein the method further comprises controlling the output device associated with the personal computer with a windowing software application.

90. The method of claim 87, wherein the method further comprises determining whether one or more of the plurality of data packets is a corrupt data packet, and discarding the corrupt data packet.

91. The method of claim 87, wherein the method further comprises delivering the one type of information in a corrupt data packet to the output device without requesting retransmission of data of the corrupt data packet.

92. The method of claim 87, wherein the one type of information comprises one of audio, textual, and graphical information.

93. The method of claim 87, wherein the output device associated with the personal computer comprises at least one of a display device and a speaker.

94. A method for receiving information from a remote location, the method comprising: providing a modem; providing a personal computer, the modem operable under control of a graphical user interface associated with the personal computer; receiving a plurality of data packets via the modern from at least one packet switched network, wherein the plurality of data packets comprise two or more different types of information, and further wherein each of the plurality of data packets comprises at least one header, the at least one header for use in routing the two or more different types of information; removing the two or more different types of information from the plurality of data packets; forming two or more data flows corresponding to the two or more different types of information; buffering the two or more data flows; and delivering the two or more data flows to at least one output device associated with the personal computer based on the at least one header for each of the plurality of data packets.

95. The method of claim 94, wherein the two or more different types of information comprise audio, textual, and graphical information.

96. The method of claim 94, wherein the graphical information comprises at least one of picture information and video information.

97. The method of claim 94, wherein the at least one output device associated with the personal computer comprises one or more of a display device and a speaker.

98. The method of claim 94, wherein the method further comprises determining that one or more of the plurality of data packets is a corrupt data packet, and discarding the corrupt data packet.

99. The method of claim 94, wherein delivering the two or more data flows comprises delivering the two or more data flows without requesting retransmission of data contained within a corrupt data packet of the plurality of data packets.

100. A system for receiving multi-media information, the system comprising: a modem; processing circuitry operable under control of a graphical user interface program executable on a personal computer, wherein the processing circuitry is operable to: receive a plurality of incoming data packets comprising one or more types of information via the modem from at least one packet switched network, wherein each of the plurality of incoming data packets comprises at least one header, the at least one header operable for use in routing the one or more types of information; remove the one or more types of information from the plurality of incoming data packets; store the one or more types of information from the plurality of incoming data packets; assemble the one or more types of information from the plurality of incoming data packets to form at least one data flow; and deliver the at least one data flow to one or more output devices associated with the personal computer based on the at least one header for each of the plurality of data packets.

101. The system of claim 100, wherein the system is operable to begin delivery of the at least one data flow to the one or more output devices prior to completing receipt of the plurality of incoming data packets.

102. The system of claim 100, wherein the one or more types of information comprise one or more of audio, textual, and graphical information.

103. The system of claim 102, wherein the graphical information comprises at least one of picture information and video information.

104. The system of claim 100, wherein the graphical user interface program comprises a windowing software application.

105. The system of claim 100, wherein the graphical user interface program comprises a display interface comprising user selectable functions to control communication of the one or more types of information to a user via the one or more output devices associated with the personal computer.

106. A system for receiving information, the system comprising: a modem; and processing circuitry operable under control of a graphical user interface program executable on a personal computer, wherein the processing circuitry is operable to: receive a plurality of data packets via the modem from at least one packet switched network, wherein the plurality of data packets comprise two or more different types of information, and further wherein each of the plurality of data packets comprises at least one header for use in routing the two or more different types of information; remove and buffer the two or more different types of information from the plurality of data packets; and deliver sequentially the two or more different types of information from the plurality of data packets to an output device associated with the personal computer based on the at least one header for each of the plurality of data packets.

107. The system of claim 106, wherein the system is operable to begin delivery of the two or more different types of information to the output device prior to completing receipt of the plurality of data packets.

108. The system of claim 106, wherein the two or more different types of information comprise two or more of audio, textual, and graphical information.

109. A method for receiving multi-media information on a personal computer from a remote location, the method comprising: providing a modem, the modem operable under control of a graphical user interface associated with a personal computer; receiving a plurality of data packets via the modem from at least one packet switched network, wherein the plurality of data packets comprise at least two different types of information, and further wherein each of the plurality of data packets comprises a header for use in routing the at least two different types of information; removing and buffering the at least two different types of information from the plurality of data packets; and delivering sequentially the at least two different types of information from the plurality of data packets to one or more output devices associated with the personal computer based on the header for each of the plurality of data packets, wherein delivering sequentially the at least two different types of information to the output device occurs prior to completing receipt of the plurality of data packets.

110. The method of claim 109, wherein the at least two different types of information comprise at least two of audio, textual, and graphical information.

111. The method of claim 109, wherein the method further comprises determining whether one or more of the plurality of data packets is a corrupt data packet, and discarding the corrupt data packet.

112. The method of claim 109, wherein delivering sequentially the at least two different types of information comprises delivering sequentially the at least two different types of information without requesting retransmission of data contained within a corrupt data packet of the plurality of data packets.

113. The method of claim 109, wherein the method further comprises controlling the one or more output devices with a windowing software application.

114. The method of claim 109, wherein the one or more output devices comprise at least one speaker.

115. The method of claim 109, wherein the one or more output devices comprise at least a video display device.
Description


FIELD OF THE INVENTION

The present invention relates to communications systems and, in particular, to computer-assisted multi-media communications.

BACKGROUND OF THE INVENTION

A wide variety of communications alternatives are currently available to telecommunications users. For example, facsimile transmission of printed matter is available through what is commonly referred to as a stand-alone fax machine. Alternatively, fax-modem communication systems are currently available for personal computer users which combine the operation of a facsimile machine with the word processor of a computer to transmit documents held on computer disk. Modem communication over telephone lines in combination with a personal computer is also known in the art, where file transfers can be accomplished from one computer to another. Also, simultaneous voice and modem data transmitted over the same telephone line has been accomplished in several ways.

There is a need in the art, however, for a personal communications system which combines a wide variety of communication functions into an integrated hardware-software product such that the user can conveniently choose a mode of communication and have that communication automatically invoked from a menu-driven selection system.

SUMMARY OF THE INVENTION

The present disclosure describes a complex computer-assisted communications system which contains multiple inventions. The subject of the present multiple inventions is a personal communications system which includes components of software and hardware operating in conjunction with a personal computer. The user interface control software operates on a personal computer, preferably within the Microsoft Windows.RTM. environment. The software control system communicates with hardware components linked to the software through the personal computer serial communications port. The hardware components include telephone communication equipment, digital signal processors, and hardware to enable both fax and data communication with hardware components at a remote site connected through a standard telephone line. The functions of the hardware components are controlled by control software operating within the hardware components and from the software components operating within the personal computer.

Communications between the software components running on the personal computer and the local hardware components over the serial communications link is by a special packet protocol for digital data communications. This bi-directional communications protocol allows uninterrupted bi-directional full-duplex transfer of both control information and data communication.

The major functions of the present system are a telephone function, a voice mail function, a fax manager function, a multi-media mail function, a show-and-tell function, a terminal function, and an address book function. The telephone function allows the present system to operate, from the user's perspective, as a conventional telephone using either hands-free, headset, or handset operation. The telephone function is more sophisticated than a standard telephone in that the present system converts the voice into a digital signal which can be processed with echo cancellation, compressed, stored as digital data for later retrieval, and transmitted as digital voice data concurrent with the transfer of digital information data.

The voice mail function of the present system operates as a telephone answering machine which can receive, compress, and store voice messages for later retrieval or reuse in response messaging.

The fax manager function of the present system allows the transmission and reception of facsimile information. The software component of the present system operates in conjunction with other commercially-available software programs, such as word processors and the like, to transmit and receive facsimile pages of digital data stored on a computer system.

The multi-media mail component of the present system allows the operator to create documents that include text, graphics, and voice mail messages which can be sent as a combined package over conventional telephone lines for receipt at a like-configured site using the present system.

The show-and-tell component of the present system enables the operator to simultaneously transmit voice and data communication to a remote site. This voice over data function dynamically allocates data bandwidth over the telephone line depending on the demands of the voice grade digitized signal.

The terminal feature of the present system allows the user to establish a data communications session with another computer system allowing the user's local computer system to operate as a dumb terminal.

The address book function of the present system is a versatile database that is built by the user and operates in conjunction with the other components of the present system to dial and establish communication links with remote sites to enable data communication, voice mail, facsimile, and file transfer all in an automated mode without user intervention.

The hardware components of the present system include circuitry to enable digital data communication and facsimile communication over standard telephone lines. The hardware components also include circuitry to convert the voice to digital data and compress that data for transfer to the software component on the personal computer or transfer it over the telephone lines to a remote site.

Many of the functions of the present system are accomplished by including a voice control digital signal processor (DSP) to operate in conjunction with a data/fax modem implemented with a data pump DSP. The data pump DSP and the voice control DSP accomplish the following functions in an integrated hardware arrangement.

A sophisticated telephone apparatus with its attached handset, headset, and a built-in hands-free telephone operation using the integrated microphone and speaker system. The hands-free telephone works in full-duplex mode through the use of voice echo cancellation performed by the voice control DSP.

The voice control DSP, in conjunction with a telephone CODEC, provides voice compression which can be sent to the computer system that is attached to the RS232 port for storage and later retrieval. The compressed voice from the voice control DSP can also be multiplexed with the input data stream from the personal computer with dynamic time allocation. Whereas, the input data from the attached computer is transmitted using the error control protocol like MNP or V.42 with or without data compression (e.g., V.42bis), the speech is packetized using a different header defining it as a speech packet and then transmitted through a controller. The speech packets, like the data packets, have the attached CRC codes. However, the speech packets are not sequenced, and the like hardware at the receiving end ignores the accompanying CRC codes for voice packets and passes the voice packets to the voice control DSP for decompression. The decompressed speech is played through one of the telephone receiving units, i.e., the headset, handset, or the built-in speaker.

The voice control DSP allows the compressed speech to be recorded on a recording media, e.g., the hard disk drive of the attached computer system. This provides the function of an answering machine. In addition to the answering machine function, the recorded speech can be provided for the voice mail functions.

The special packet protocol over the RS232 interface between the software component and the hardware component that governs the operation of the hardware component is so designed that it allows various control functions to be intermixed with data over the RS232 serial port. The software component of the present system accepts the generic AT modem commands when not in the special packet mode. When the hardware component is configured to accept the packet level protocol over the RS232 port, it can be made to switch to the generic command mode through the use of a break sequence.

The hardware components of the present system function as a data/fax modem when the speech compression or telephone mode is not invoked. The packet mode or the generic AT command mode may be used for this purpose.

The hardware components of the present system incorporate a provision for a special link integrity packet to facilitate the device to work over cellular networks. This scheme allows the modem in one of its plurality of modes to ignore the carrier drop-outs (selective fading) inherent in the cellular networks. Such a scheme does not use carrier detect circuitry of the modem. The disconnect of the cellular connection is done through a negotiation scheme using packet interchange between the two ends of the link.

In cellular networks, the multiplexed voice data technology of the present system allows a single apparatus to function as a smart telephone, an intelligent data modem, as well as a fax modem. These features, along with the voice data multiplex mode, provide a traveling user complete freedom to use his or her moving vehicle as a true traveling office.

These features of the hardware component of the present system, along with the features of the software component of the present system running on a PC, provides a user with a complete range of telecommunications functions of a modern office, be it stationary or mobile.

DESCRIPTION OF THE DRAWINGS

In the drawings, where like numerals describe like components throughout the several views,

FIG. 1 shows the telecommunications environment within which the present system may operate in several of the possible modes of communication;

FIG. 2 is the main menu icon for the software components operating on the personal computer;

FIG. 3 is a block diagram of the hardware components of the present system;

FIG. 4 is a key for viewing the detailed electrical schematic diagrams of

FIGS. 5A 10C to facilitate understanding of the interconnect between the drawings;

FIGS. 5A 5C, 6A 6C, 7A 7C, 8A 8B, 9A 9C, and 10A 10C are detailed electrical schematic diagrams of the circuitry of the hardware components of the present system;

FIG. 11 is a signal flow diagram of the speech compression algorithm;

FIG. 12 is a detailed function flow diagram of the speech compression algorithm;

FIG. 13 is a detailed function flow diagram of the speech decompression algorithm;

FIG. 14 is a detailed function flow diagram of the echo cancellation algorithm;

FIG. 15 is a detailed function flow diagram of the voice/data multiplexing function;

FIG. 16 is a perspective view of the components of a digital computer compatible with the present invention;

FIG. 17 is a block diagram of the software structure compatible with the present invention;

FIG. 18 is a block diagram of the control structure of software compatible with the present invention;

FIG. 19 is a block diagram of the main menu structure of software compatible with the present invention;

FIG. 20 is a flow diagram of answer mode software compatible with the present invention;

FIG. 21 is a flow diagram of telephone software compatible with the present invention;

FIG. 22 is a flow diagram of voice mail software compatible with the present invention;

FIG. 23 is a flow diagram of fax manager software compatible with the present invention;

FIG. 24 is a flow diagram of multi-media mail software compatible with the present invention;

FIG. 25 is a flow diagram of a timing loop compatible with the present invention;

FIG. 26 is a flow diagram of telephone control software compatible with the present invention;

FIG. 27 is a flow diagram of voice mail control software compatible with the present invention;

FIG. 28 is a flow diagram of high resolution fax driver software compatible with the present invention;

FIG. 29 is a flow diagram of low resolution fax driver software compatible with the present invention;

FIG. 30 is a flow diagram of multi-media mail control software compatible with the present invention;

FIG. 31 is a flow diagram of multi-media mail editor software compatible with the present invention;

FIG. 32 is a flow diagram of multi-medial mail transmit software compatible with the present invention;

FIG. 33 is a flow diagram of multi-medial mail receive software compatible with the present invention;

FIG. 34 is a flow diagram of show-and-tell transmit software compatible with the present invention;

FIG. 35 is a flow diagram of show-and-tell receive software compatible with the present invention;

FIG. 36 is a flow diagram of voice mail transmit software compatible with the present invention;

FIG. 37 is a flow diagram of voice mail receive software compatible with the present invention;

FIG. 38 is a flow diagram of an outgoing timer loop compatible with the present invention;

FIG. 39 is a flow diagram of an incoming timer loop compatible with the present invention;

FIG. 40 is an initialization screen display compatible with the present invention;

FIG. 41 is a communication port setup screen display compatible with the present invention;

FIG. 42 is an answer mode setup screen display compatible with the present invention;

FIG. 43 is a hold call setup screen display compatible with the present invention;

FIG. 44 is a voice mail setup screen display compatible with the present invention;

FIG. 45 is a PBX setup screen display compatible with the present invention;

FIG. 46 is a fax setup screen display compatible with the present invention;

FIG. 47 is a multi-media mail set-up screen display compatible with the present invention;

FIG. 48 is a show-and-tell set-up screen display compatible with the present invention;

FIG. 49 is a telephone control screen display compatible with the present invention;

FIG. 50 is a voice mail control screen display compatible with the present invention;

FIG. 51 is a voice editor screen display compatible with the present invention;

FIG. 52 is a fax manager control screen display compatible with the present invention;

FIG. 53 is a multi-media mail control screen display compatible with the present invention;

FIG. 54 is a show-and-tell control screen display compatible with the present invention;

FIG. 55 is an address book control screen display compatible with the present invention;

FIG. 56 is a voice message destination screen display compatible with the present invention; and

FIG. 57 is a message composer screen display compatible with the present invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

The specifications for the multiple inventions described herein includes the present description, the drawings, and a microfiche appendix. In the following detailed description of the preferred embodiment, reference is made to the accompanying drawings which form a part hereof, and in which is shown by way of illustration specific embodiments in which the inventions may be practiced. These embodiments are described in sufficient detail to enable those skilled in the art to practice the invention, and it is to be understood that other embodiments may be utilized and that structural changes may be made without departing from the spirit and scope of the present inventions. The following detailed description is, therefore, not to be taken in a limiting sense, and the scope of the present inventions is defined by the appended claims.

FIG. 1 shows a typical arrangement for the use of the present system. Personal computer 10 is running the software components of the present system while the hardware components 20 include the data communication equipment and telephone headset. Hardware components 20 communicate over a standard telephone line 30 to one of a variety of remote sites. One of the remote sites may be equipped with the present system, including hardware components 20a and software components running on personal computer 10a. In one alternative use, the local hardware components 20 may be communicating over standard telephone line 30 to facsimile machine 60. In another alternative use, the present system may be communicating over a standard telephone line 30 to another personal computer 80 through a remote modem 70. In another alternative use, the present system may be communicating over a standard telephone line 30 to a standard telephone 90. Those skilled in the art will readily recognize the wide variety of communication interconnections possible with the present system by reading and understanding the following detailed description.

The ornamental features of the hardware components 20 of FIG. 1 are claimed as part of Design patent application Ser. No. 29/001368, filed Nov. 12, 1992, entitled, "Telephone/Modem case for a Computer-Based Multifunction Personal Communications System," assigned to the same assignee of the present inventions and hereby incorporated by reference.

General Overview

The present inventions are embodied in a commercial product by the assignee, Multi-Tech Systems, Inc. The software component operating on a personal computer is sold under the commercial trademark of MultiExpress PCS.TM. personal communications software, while the hardware component of the present system is sold under the commercial name of MultiModemPCS.TM., Intelligent Personal Communications System Modem. In the preferred embodiment, the software component runs under Microsoft.RTM. Windows.RTM., however, those skilled in the art will readily recognize that the present system is easily adaptable to run under any single- or multi-user, single- or multi-window operating system.

The present system is a multi-function communication system which includes hardware and software components. The system allows the user to connect to remote locations equipped with a similar system or with modems, facsimile machines, or standard telephones over a single analog telephone line. The software component of the present system includes a number of modules which are described in more detail below.

FIG. 2 is an example of the Windows.RTM.-based main menu icon of the present system operating on a personal computer. The functions listed with the icons used to invoke those functions are shown in the preferred embodiment. Those skilled in the art will readily recognize that a wide variety of selection techniques may be used to invoke the various functions of the present system. The icon of FIG. 2 is part of Design patent application Ser. No. 29/001397, filed Nov. 12, 1992, entitled, "Icons for a Computer-Based Multifunction Personal Communications System," assigned to the same assignee of the present inventions and hereby incorporated by reference.

The telephone module allows the system to operate as a conventional or sophisticated telephone system. The system converts voice into a digital signal so that it can be transmitted or stored with other digital data, like computer information. The telephone function supports PBX and Centrex features such as call waiting, call forwarding, caller ID, and three-way calling. This module also allows the user to mute, hold, or record a conversation. The telephone module enables the handset, headset, or hands-free speaker telephone operation of the hardware component. It includes on-screen pushbutton dialing, speed-dial of stored numbers, and digital recording of two-way conversations.

The voice mail portion of the present system allows this system to operate as a telephone answering machine by storing voice messages as digitized voice files along with a time/date voice stamp. The digitized voice files can be saved and sent to one or more destinations immediately or at a later time using a queue scheduler. The user can also listen to, forward, or edit the voice messages which have been received with a powerful digital voice editing component of the present system. This module also creates queues for outgoing messages to be sent at pre-selected times and allows the users to create outgoing messages with the voice editor.

The fax manager portion of the present system is a queue for incoming and outgoing facsimile pages. In the preferred embodiment of the present system, this function is tied into the Windows "print" command once the present system has been installed. This feature allows the user to create faxes from any Windows.RTM.-based document that uses the "print" command. The fax manager function of the present system allows the user to view queued faxes which are to be sent or which have been received. This module creates queues for outgoing faxes to be sent at pre-selected times and logs incoming faxes with time/date stamps.

The multi-media mail function of the present system is a utility which allows the user to compose documents that include text, graphics, and voice messages using the message composer function of the present system, described more fully below. The multi-media mail utility of the present system allows the user to schedule messages for transmittal and queues up the messages that have been received so that they can be viewed at a later time.

The show-and-tell function of the present system allows the user to establish a data over voice (DOV) communications session. When the user is transmitting data to a remote location similarly equipped, the user is able to talk to the person over the telephone line while concurrently transferring the data. This voice over data function is accomplished in the hardware components of the present system. It digitizes the voice and transmits it in a dynamically changing allocation of voice data and digital data multiplexed in the same transmission. The allocation at a given moment is selected depending on amount of voice digital information required to be transferred. Quiet voice intervals allocate greater space to the digital data transmission.

The terminal function of the present system allows the user to establish a data communications session with another computer which is equipped with a modem but which is not equipped with the present system. This feature of the present system is a Windows.RTM.-based data communications program that reduces the need for issuing "AT" commands by providing menu-driven and "pop-up" window alternatives.

The address book function of the present system is a database that is accessible from all the other functions of the present system. This database is created by the user inputting destination addresses and telephone numbers for data communication, voice mail, facsimile transmission, modem communication, and the like. The address book function of the present system may be utilized to broadcast communications to a wide variety of recipients. Multiple linked databases have separate address books for different groups, and different destinations may be created by the users. The address book function includes a textual search capability which allows fast and efficient location of specific addresses, as described more fully below.

Hardware Components

FIG. 3 is a block diagram of the hardware components of the present system corresponding to reference number 20 of FIG. 1. These components form the link between the user, the personal computer running the software component of the present system, and the telephone line interface. As will be more fully described below, the interface to the hardware components of the present system is via a serial communications port connected to the personal computer. The interface protocol is well ordered and defined such that other software systems or programs running on the personal computer may be designed and implemented which would be capable of controlling the hardware components shown in FIG. 3 by using the control and communications protocol defined below.

In the preferred embodiment of the present system, three alternate telephone interfaces are available: the telephone handset 301, a telephone headset 302, and a hands-free microphone 303 and speaker 304. Regardless of the telephone interface, the three alternative interfaces connect to the digital telephone coder-decoder (CODEC) circuit 305.

The digital telephone CODEC circuit 305 interfaces with the voice control digital signal processor (DSP) circuit 306 which includes a voice control DSP and CODEC. This circuit does digital-to-analog (D/A) conversion, analog-to-digital (A/D) conversion, coding/decoding, gain control, and is the interface between the voice control DSP circuit 306 and the telephone interface. The CODEC of the voice control circuit 306 transfers digitized voice information in a compressed format to multiplexor circuit 310 and to analog telephone line interface 309.

The CODEC of the voice control circuit 306 is actually an integral component of a voice control digital signal processor integrated circuit, as described more fully below. The voice control DSP of circuit 306 controls the digital telephone CODEC circuit 305, performs voice compression, and echo cancellation.

Multiplexor (MUX) circuit 310 selects between the voice control DSP circuit 306 and the data pump DSP circuit 311 for transmission of information on the telephone line through telephone line interface circuit 309.

The data pump circuit 311 also includes a digital signal processor (DSP) and a CODEC for communicating over the telephone line interface 309 through MUX circuit 310. The data pump DSP and CODEC of circuit 311 performs functions such as modulation, demodulation, and echo cancellation to communicate over the telephone line interface 309 using a plurality of telecommunications standards, including FAX and modem protocols.

The main controller circuit 313 controls the DSP data pump circuit 311 and the voice control DSP circuit 306 through serial input/output and clock timer control (SIO/CTC) circuits 312 and dual port RAM circuit 308, respectively. The main controller circuit 313 communicates with the voice control DSP 306 through dual port RAM circuit 308. In this fashion, digital voice data can be read and written simultaneously to the memory portions of circuit 308 for high speed communication between the user (through interfaces 301, 302, or 303/304) and the personal computer connected to serial interface circuit 315 and the remote telephone connection connected through the telephone line attached to line interface circuit 309.

As described more fully below, the main controller circuit 313 includes, in the preferred embodiment, a microprocessor which controls the functions and operation of all of the hardware components shown in FIG. 3. The main controller is connected to RAM circuit 316 and a programmable and electrically erasable read-only memory (PEROM) circuit 317. The PEROM circuit 317 includes non-volatile memory in which the executable control programs for the voice control DSP circuits 306 and the main controller circuits 313 operate.

The RS232 serial interface circuit 315 communicates to the serial port of the personal computer which is running the software components of the present system. The RS232 serial interface circuit 315 is connected to a serial input/output circuit 314 with main controller circuit 313. SIO circuit 314 is in the preferred embodiment, a part of SIO/CTC circuit 312.

Functional Operation of the Hardware Components

Referring once again to FIG. 3, the multiple and selectable functions described in conjunction with FIG. 2 are all implemented in the hardware components of FIG. 3. Each of these functions will be discussed in turn.

The telephone function 115 is implemented by the user either selecting a telephone number to be dialed from the address book 127 or manually selecting the number through the telephone menu on the personal computer. The telephone number to be dialed is downloaded from the personal computer over the serial interface and received by main controller 313. Main controller 313 causes the data pump DSP circuit 311 to seize the telephone line and transmit the DTMF tones to dial a number. Main controller 313 configures digital telephone CODEC circuit 305 to enable either the handset 301 operation, the microphone 303 and speaker 304 operation, or the headset 302 operation. A telephone connection is established through the telephone line interface circuit 309 and communication is enabled. The user's analog voice is transmitted in an analog fashion to the digital telephone CODEC 305 where it is digitized. The digitized voice patterns are passed to the voice control circuit 306 where echo cancellation is accomplished, and the digital voice signals are reconstructed into analog signals and passed through multiplexor circuit 310 to the telephone line interface circuit 309 for analog transmission over the telephone line. The incoming analog voice from the telephone connection through telephone connection circuit 309 is passed to the integral CODEC of the voice control circuit 306 where it is digitized. The digitized incoming voice is then passed to digital telephone CODEC circuit 305 where it is reconverted to an analog signal for transmission to the selected telephone interface (either the handset 301, the microphone/speaker 303/304, or the headset 302). Voice control DSP circuit 306 is programmed to perform echo cancellation to avoid feedback and echoes between transmitted and received signals, as is more fully described below.

In the voice mail function mode of the present system, voice messages may be stored for later transmission or the present system may operate as an answering machine receiving incoming messages. For storing digitized voice, the telephone interface is used to send the analog speech patterns to the digital telephone CODEC circuit 305. Circuit 305 digitizes the voice patterns and passes them to voice control circuit 306 where the digitized voice patterns are digitally compressed. The digitized and compressed voice patterns are passed through dual port RAM circuit 308 to the main controller circuit 313 where they are transferred through the serial interface to the personal computer using a packet protocol defined below. The voice patterns are then stored on the disk of the personal computer for later use in multi-media mail, for voice mail, as a pre-recorded answering machine message, or for later predetermined transmission to other sites.

For the present system to operate as an answering machine, the hardware components of FIG. 3 are placed in answer mode. An incoming telephone ring is detected through the telephone line interface circuit 309, and the main controller circuit 313 is alerted which passes the information off to the personal computer through the RS232 serial interface circuit 315. The telephone line interface circuit 309 seizes the telephone line to make the telephone connection. A pre-recorded message may be sent by the personal computer as compressed and digitized speech through the RS232 interface to the main controller circuit 313. The compressed and digitized speech from the personal computer is passed from main controller circuit 313 through dual port RAM circuit 308 to the voice control DSP circuit 306 where it is uncompressed and converted to analog voice patterns. These analog voice patterns are passed through multiplexor circuit 310 to the telephone line interface 309 for transmission to the caller. Such a message may invite the caller to leave a voice message at the sound of a tone. The incoming voice messages are received through telephone line interface 309 and passed to voice control circuit 306. The analog voice patterns are digitized by the integral CODEC of voice control circuit 306 and the digitized voice patterns are compressed by the voice control DSP of the voice control circuit 306. The digitized and compressed speech patterns are passed through dual port RAM circuit 308 to the main controller circuit 313, where they are transferred using packet protocol described below through the RS232 serial interface 315 to the personal computer for storage and later retrieval. In this fashion, the hardware components of FIG. 3 operate as a transmit and receive voice mail system for implementing the voice mail function 117 of the present system.

The hardware components of FIG. 3 may also operate to facilitate the fax manager function 119 of FIG. 2. In fax receive mode, an incoming telephone call will be detected by a ring detect circuit of the telephone line interface 309 which will alert the main controller circuit 313 to the incoming call. Main controller circuit 313 will cause line interface circuit 309 to seize the telephone line to receive the call. Main controller circuit 313 will also concurrently alert the operating programs on the personal computer through the RS232 interface using the packet protocol described below. Once the telephone line interface seizes the telephone line, a fax carrier tone is transmitted and a return tone and handshake is received from the telephone line and detected by the data pump circuit 311. The reciprocal transmit and receipt of the fax tones indicates the imminent receipt of a facsimile transmission, and the main controller circuit 313 configures the hardware components of FIG. 3 for the receipt of that information. The necessary handshaking with the remote facsimile machine is accomplished through the data pump 311 under control of the main controller circuit 313. The incoming data packets of digital facsimile data are received over the telephone line interface and passed through data pump circuit 311 to main controller circuit 313, which forwards the information on a packet basis (using the packet protocol described more fully below) through the serial interface circuit 315 to the personal computer for storage on disk. Those skilled in the art will readily recognize that the fax data could be transferred from the telephone line to the personal computer using the same path as the packet transfer except using the normal AT stream mode. Thus, the incoming facsimile is automatically received and stored on the personal computer through the hardware components of FIG. 3.

A facsimile transmission is also facilitated by the hardware components of FIG. 3. The transmission of a facsimile may be immediate or queued for later transmission at a pre-determined or pre-selected time. Control packet information to configure the hardware components to send a facsimile are sent over the RS232 serial interface between the personal computer and the hardware components of FIG. 3 and are received by main controller circuit 313. The data pump circuit 311 then dials the recipient's telephone number using DTMF tones or pulse dialing over the telephone line interface circuit 309. Once an appropriate connection is established with the remote facsimile machine, standard facsimile handshaking is accomplished by the data pump circuit 311. Once the facsimile connection is established, the digital facsimile picture information is received through the data packet protocol transfer over serial line interface circuit 315, passed through main controller circuit 313 and data pump circuit 311 onto the telephone line through telephone line interface circuit 309 for receipt by the remote facsimile machine.

The operation of the multi-media mail function 121 of FIG. 2 is also facilitated by the hardware components of FIG. 3. A multi-media transmission consists of a combination of picture information, digital data, and digitized voice information. For example, the type of multi-media information transferred to a remote site using the hardware components of FIG. 3 could be the multi-media format of the Microsoft.RTM. Multimedia Wave.RTM. format with the aid of an Intelligent Ser. Interface (ISI) card added to the personal computer. The multi-media may also be the type of multi-media information assembled by the software component of the present system, which is described more fully below.

The multi-media package of information including text, graphics, and voice messages (collectively called the multi-media document) may be transmitted or received through the hardware components shown in FIG. 3. For example, the transmission of a multi-media document through the hardware components of FIG. 3 is accomplished by transferring the multi-media digital information using the packet protocol described below over the RS232 serial interface between the personal computer and the serial line interface circuit 315. The packets are then transferred through main controller circuit 313 through the data pump circuit 311 on to the telephone line for receipt at a remote site through telephone line interface circuit 309. In a similar fashion, the multi-media documents received over the telephone line from the remote site are received at the telephone line interface circuit 309, and passed through the data pump circuit 311 for receipt and forwarding by the main controller circuit 313 over the serial line interface circuit 315.

The show-and-tell function 123 of the present system allows the user to establish a data over voice communication session. In this mode of operation, full-duplex data transmission may be accomplished simultaneously with the voice communication between both sites. This mode of operation assumes a like configured remote site. The hardware components of the present system also include a means for sending voice/data over cellular links. The protocol used for transmitting multiplexed voice and data include a supervisory packet described more fully below to keep the link established through the cellular link. This supervisory packet is an acknowledgment that the link is still up. The supervisory packet may also contain link information to be used for adjusting various link parameters when needed. This supervisory packet is sent every second when data is not being sent, and if the packet is not acknowledged after a specified number of attempts, the protocol would then give an indication that the cellular link is down and then allow the modem to take action. The action could be, for example, change speeds, retrain, or hang up. The use of supervisory packets is a novel method of maintaining inherently intermittent cellular links when transmitting multiplexed voice and data.

The voice portion of the voice over data transmission of the show-and-tell function is accomplished by receiving the user's voice through the telephone interface 301, 302, or 303, and the voice information is digitized by the digital telephone circuit 305. The digitized voice information is passed to the voice control circuit 306, where the digitized voice information is compressed using a voice compression algorithm described more fully below. The digitized and compressed voice information is passed through dual port RAM circuit 308 to the main controller circuit 313. During quiet periods of the speech, a quiet flag is passed from voice control circuit 306 to the main controller 313 through a packet transfer protocol described below by a dual port RAM circuit 308.

Simultaneous with the digitizing compression and packetizing of the voice information is the receipt of the packetized digital information from the personal computer over interface line circuit 315 by main controller circuit 313. Main controller circuit 313 in the show-and-tell function of the present system must efficiently and effectively combine the digitized voice information with the digital information for transmission over the telephone line via telephone line interface circuit 309. As described above, and as described more fully below, main controller circuit 313 dynamically changes the amount of voice information and digital information transmitted at any given period of time, depending upon the quiet times during the voice transmissions. For example, during a quiet moment where there is no speech information being transmitted, main controller circuit 313 ensures that a higher volume of digital data information be transmitted over the telephone line interface in lieu of digitized voice information.

Also, as described more fully below, the packets of digital data transmitted over the telephone line interface with the transmission packet protocol described below requires 100% accuracy in the transmission of the digital data, but a lesser standard of accuracy for the transmission and receipt of the digitized voice information. Since digital information must be transmitted with 100% accuracy, a corrupted packet of digital information received at the remote site must be retransmitted. A retransmission signal is communicated back to the local site and the packet of digital information which was corrupted during transmission is retransmitted. If the packet transmitted contained voice data, however, the remote site uses the packets whether they were corrupted or not, as long as the packet header was intact. If the header is corrupted, the packet is discarded. Thus, the voice information may be corrupted without requesting retransmission, since it is understood that the voice information must be transmitted on a real-time basis and the corruption of any digital information of the voice signal is not critical. In contrast to this, the transmission of digital data is critical and retransmission of corrupted data packets is requested by the remote site.

The transmission of the digital data follows the CCITT V.42 standard, as is well known in the industry and as described in the CCITT Blue Book, Volume VIII, entitled Data Communication over the Telephone Network, 1989. The CCITT V.42 standard is hereby incorporated by reference. The voice data packet information also follows the CCITT V.42 standard but uses a different header format so the receiving site recognizes the difference between a data packet and a voice packet. The voice packet is distinguished from a data packet by using undefined bits in the header (80 hex) of the V.42 standard. The packet protocol for voice over data transmission during the show-and-tell function of the present system is described more fully below.

Since the voice over data communication with the remote site is full-duplex, incoming data packets and incoming voice packets are received by the hardware components of FIG. 3. The incoming data packets and voice packets are received through the telephone line interface circuit 309 and passed to the main controller circuit 313 via data pump DSP circuit 311. The incoming data packets are passed by the main controller circuit 313 to the serial interface circuit 315 to be passed to the personal computer. The incoming voice packets are passed by the main controller circuit 313 to the dual port RAM circuit 308 for receipt by the voice control DSP circuit 306. The voice packets are decoded and the compressed digital information therein is uncompressed by the voice control DSP of circuit 306. The uncompressed digital voice information is passed to digital telephone CODEC circuit 305 where it is reconverted to an analog signal and retransmitted through the telephone line interface circuits. In this fashion, full-duplex voice and data transmission and reception is accomplished through the hardware components of FIG. 3 during the show-and-tell functional operation of the present system.

Terminal operation 125 of the present system is also supported by the hardware components of FIG. 3. Terminal operation means that the local personal computer simply operates as a "dumb" terminal including file transfer capabilities. Thus, no local processing takes place other than the handshaking protocol required for the operation of a dumb terminal. In terminal mode operation, the remote site is assumed to be a modem connected to a personal computer, but the remote site is not necessarily a site which is configured according to the present system. In terminal mode of operation, the command and data information from the personal computer is transferred over the RS232 serial interface circuit 315 and forwarded by main controller circuit 313 to the data pump circuit 311 where the data is placed on the telephone line via telephone line interface circuit 309.

In a reciprocal fashion, data is received from the telephone line over telephone line interface circuit 309 and simply forwarded by the data pump circuit 311, the main controller circuit 313 over the serial line interface circuit 315 to the personal computer.

As described above, and more fully below, the address book function of the present system is primarily a support function for providing telephone numbers and addresses for the other various functions of the present system.

Detailed Electrical Schematic Diagrams

The detailed electrical schematic diagrams comprise FIGS. 5A C, 6A C, 7A C, 8A B, 9A C, and 10A C. FIG. 4 shows a key on how the schematic diagrams may be conveniently arranged to view the passing of signals on the electrical lines between the diagrams. The electrical connections between the electrical schematic diagrams are through the designators listed next to each wire. For example, on the right side of FIG. 5A, address lines A0 A19 are attached to an address bus for which the individual electrical lines may appear on other pages as A0 A19 or may collectively be connected to other schematic diagrams through the designator "A" in the circle connected to the collective bus. In a like fashion, other electrical lines designated with symbols, such as RNGL on the lower left-hand side of FIG. 5A, may connect to other schematic diagrams using the same signal designator RNGL.

Beginning with the electrical schematic diagram of FIG. 7C, the telephone line connection in the preferred embodiment is through connector J2, which is a standard 6-pin modular RJ-11 jack. In the schematic diagram of FIG. 7C, only the tip and ring connections of the first telephone circuit of the RJ-11 modular connector are used. Ferrite beads FB3 and FB4 are placed on the tip and ring wires of the telephone line connections to remove any high frequency or RF noise on the incoming telephone line. The incoming telephone line is also over-voltage protected through SIDACTOR R4. The incoming telephone line may be full wave rectified by the full wave bridge comprised of diodes CR27, CR28, CR29, and CR31. Switch S4 switches between direct connection and full wave rectified connection depending upon whether the line is a non-powered leased line or a standard telephone line. Since a leased line is a "dead" line with no voltage, the full wave rectification is not needed.

Also connected across the incoming telephone line is a ring detect circuit. Optical isolator U32 (part model number CNY17) senses the ring voltage threshold when it exceeds the breakdown voltages on zener diodes CR1 and CR2. A filtering circuit shown in the upper right corner of FIG. 7C creates a long RC delay to sense the constant presence of an AC ring voltage and buffers that signal to be a binary signal out of operational amplifier U25 (part model number TLO82). Thus, the RNGL and J1RING signals are binary signals for use in the remaining portions of the electrical schematic diagrams to indicate a presence of a ring voltage on the telephone line.

The present system is also capable of sensing the caller ID information which is transmitted on the telephone line between rings. Between the rings, optically isolated relays U30, U31 on FIG. 7C and optically isolated relay U33 on FIG. 7B all operate in the period between the rings so that the FSK modulated caller ID information is connected to the CODEC and data pump DSP in FIGS. 8A and 8B, as described more fully below.

Referring now to FIG. 7B, more of the telephone line filtering circuitry is shown. Some of the telephone line buffering circuitry, such as inductor L1 and resistor R1, are optional and are connected for various telephone line standards used around the world to meet local requirements. For example, Switzerland requires a 22 millihenry inductor and 1K resistor in series with the line. For all other countries, the 1K resistor is replaced with a 0 ohm resistor.

Relay U29, shown in FIG. 7B, is used to accomplish pulse dialing by opening and shorting the tip and ring wires. Optical relay X2 is engaged during pulse dialing so that the tip and ring are shorted directly. Transistors Q2 and Q3, along with the associated discrete resistors, comprise a holding circuit to provide a current path or current loop on the telephone line to grab the line.

FIG. 7A shows the telephone interface connections between the hardware components of the present system and the handset, headset, and microphone.

The connections T1 and T2 for the telephone line from FIG. 7B are connected to transformer TR1 shown in the electrical schematic diagram of FIG. 8B. Only the AC components of the signal pass through transformer TR1. The connection of signals attached to the secondary of TR1 is shown for both transmitting and receiving information over the telephone line.

Incoming signals are buffered by operational amplifiers U27A and U27B. The first stage of buffering using operational amplifier U27B is used for echo suppression so that the transmitted information being placed on the telephone line is not fed back into the receive portion of the present system. The second stage of the input buffering through operational amplifier U27A is configured for a moderate amount of gain before driving the signal into CODEC U35.

CODEC chip U35 on FIG. 8B, interface chip U34 on FIG. 8A, and digital signal processor (DSP) chip U37 on FIG. 8A comprise a data pump chip set manufactured and sold by AT&T Microelectronics. A detailed description of the operation of these three chips in direct connection and cooperation with one another is described in the publication entitled, "AT&T V.32bis/V.32/FAX High-Speed Data Pump Chip Set Data Book," published by AT&T Microelectronics, December 1991, which is hereby incorporated by reference. This AT&T data pump chip set comprises the core of an integrated, two-wire full-duplex modem which is capable of operation over standard telephone lines or leased lines. The data pump chip set conforms to the telecommunications specifications in CCITT recommendations V.32bis, V.32, V.22bis, V.22, V.23, V.21, and is compatible with the Bell 212A and 103modems. Speeds of 14400, 9600, 4800, 2400, 1200, 600, and 300 bits per second are supported. This data pump chip set consists of a ROM-coded DSP16A digital signal processor U37, an interface chip U34, and an AT&T T7525 linear CODEC U35. The AT&T V.32 data pump chip set is available from AT&T Microelectronics.

The chip set U34, U35, and U37 on FIGS. 8A and 8B perform all A/D, D/A, modulation, demodulation, and echo cancellation of all signals placed on or taken from the telephone line. The CODEC U35 performs DTMF tone generation and detection, signal analysis of call progress tones, etc. The transmission of information on the telephone line from CODEC U35 is through buffer U28A, through CMOS switch U36, and through line buffer U25. The CMOS switch U36 is used to switch between the data pump chip set CODEC of circuit 310 (shown in FIG. 3) and the voice control CODEC of circuit 306 (also shown in FIG. 3). The signal lines AOUTN and AOUTP correspond to signals received from the voice control CODEC of circuit 306. CODEC U35 is part of circuit 311 of FIG. 3.

The main controller of controller circuit 313 and the support circuits 312, 314, 316, 317, and 308 are shown in FIGS. 5A 5C. In the preferred embodiment of the present system, the main controller is a Z80180 8-bit microprocessor chip. In the preferred implementation, microcontroller chip U17 is a Z80180 microprocessor, part number Z84C01 by Zilog, Inc. of Campbell, Calif. (also available from Hitachi Semiconductor as part number HD64180Z). The Zilog Z80180 8-bit microprocessor operates at 12 MHz internal clock speed by means of an external crystal XTAL, which, in the preferred embodiment, is a 24.576 MHz crystal. The crystal circuit includes capacitors C4 and C5, which are 20 pf capacitors, and resistor R28, which is a 33 ohm resistor. The crystal and support circuitry is connected according to manufacturer's specifications found in the Zilog Intelligent Peripheral Controllers Data Book published by Zilog, Inc. The product description for the Z84C01 Z80180 CPU from the Z84C01 Z80 CPU Product Specification pgs. 43 73 of the Zilog 1991 Intelligent Peripheral Controllers databook is hereby incorporated by reference.

The Z80180 microprocessor in microcontroller chip U17 is intimately connected to a serial/parallel I/O counter timer chip U15 which is, in the preferred embodiment, a Zilog 84C90 CMOS Z80 KIO serial/parallel/counter/timer integrated circuit available from Zilog, Inc. This multi-function I/O chip U15 combines the functions of a parallel input/output port, a serial input/output port, a bus control circuitry, and a clock timer circuit in one chip. The Zilog Z84C90 product specification describes the detailed internal operations of this circuit in the Zilog Intelligent Peripheral Controllers 1991 Handbook available from Zilog, Inc. Z84C90 CMOS Z80KIO Product Specification pgs. 205 224 of the Zilog 1991 Intelligent Peripheral Controllers Data Book is hereby incorporated by reference.

Data and address buses A and B shown in FIG. 5A connect the Z80180 microprocessor in microcontroller U17 with the Z80 KIO circuit U15 and a gate array circuit U19, and to other portions of the electrical schematic diagrams. The gate array U19 includes miscellaneous latch and buffer circuits for the present system which normally would be found in discrete SSI or MSI integrated circuits. By combining a wide variety of miscellaneous support circuits into a single gate array, a much reduced design complexity and manufacturing cost is achieved. A detailed description of the internal operations of gate array U19 is described more fully below in conjunction with schematic diagrams of FIGS. 10A 10C.

The memory chips which operate in conjunction with the Z80 microprocessor in microcontroller chip U17 are shown in FIG. 5C. The connections A, B correspond to the connections to the address and data buses, respectively, found on FIG. 5A. Memory chips U16 and U13 are read-only memory (ROM) chips which are electrically alterable in place. These programmable ROMs, typically referred to as flash PROMs or Programmable Erasable Read-Only Memories (PEROMs), hold the program code and operating parameters for the present system in a non-volatile memory. Upon power-up, the programs and operating parameters are transferred to the voice control DSP RAM U12, shown in FIG. 9B.

In the preferred embodiment, RAM chip U14 is a pseudostatic RAM which is essentially a dynamic RAM with a built-in refresh. Those skilled in the art will readily recognize that a wide variety of memory chips may be used and substituted for pseudo-static RAM U14 and flash PROMs U16 and U13.

Referring once again to FIG. 3, the main controller circuit 313 communicates with the voice control DSP of circuit 306 through dual port RAM circuit 308. The digital telephone CODEC circuit 305, the voice control DSP and CODEC circuit 306, the DSP RAM 307, and the dual port RAM 308 are all shown in detailed electrical schematic diagrams of FIGS. 9A 9C.

Referring to FIG. 9A, the DSP RAM chips U6 and U7 are shown with associated support chips. Support chips U1 and U2 are in the preferred embodiment part 74HCT244 which are TTL-level latches used to capture data from the data bus and hold it for the DSP RAM chips U6 and U7. Circuits U3 and U4 are also latch circuits for also latching address information to control DSP RAM chips U6 and U7. Once again, the address bus A and data bus B shown in FIG. 9A are multi-wire connections which, for the clarity of the drawing, are shown as a thick bus wire representing a grouping of individual wires.

Also in FIG. 9A, the DSP RAMs U6 and U7 are connected to the voice control DSP and CODEC chip U8 as shown split between FIGS. 9A and 9B. DSP/CODEC chip U8 is, in the preferred embodiment, part number WE.RTM. DSP16C, digital signal processor and CODEC chip manufactured and sold by AT&T Microelectronics. This is a 16-bit programmable DSP with a voice band sigma-delta CODEC on one chip. Although the CODEC portion of this chip is capable of analog-to-digital and digital-to-analog signal acquisition and conversion system, the actual D/A and A/D functions for the telephone interface occur in digital telephone CODEC chip U12 (corresponding to digital telephone CODEC circuit 305 of FIG. 3). Chip U8 includes circuitry for sampling, data conversion, anti-aliasing filtering, and anti-imaging filtering. The programmable control of DSP/CODEC chip U8 allows it to receive digitized voice from the telephone interface (through digital telephone CODEC chip U12) and store it in a digitized form in the dual port RAM chip U11. The digitized voice can then be passed to the main controller circuit 313, where the digitized voice may be transmitted to the personal computer over the RS232 circuit 315. In a similar fashion, digitized voice stored by the main controller circuit 313 in the dual port RAM U11 may be transferred through voice control DSP chip U8, converted to analog signals by telephone CODEC U12, and passed to the user. Digital telephone CODEC chip U12 includes a direct telephone handset interface on the chip.

The connections to DSP/CODEC chip U8 are shown split across FIGS. 9A and 9B. Address/data decode chips U9 and U10 on FIG. 9A serve to decode address and data information from the combined address/data bus for the dual port RAM chip U11 of FIG. 9B. The interconnection of the DSP/CODEC chip U8 shown on FIGS. 9A and 9B is described more fully in the WE.RTM. DSP16C Digital Signal Processor/CODEC Data Sheet published May, 1991, by AT&T Microelectronics, which is hereby incorporated by reference.

The Digital Telephone CODEC chip U12 is also shown in FIG. 9B which, in the preferred embodiment, is part number T7540 Digital Telephone CODEC manufactured and sold by AT&T Microelectronics. A more detailed description of this telephone CODEC chip U12 is described in the T7540 Digital Telephone CODEC Data Sheet and Addendum published July, 1991, by AT&T Microelectronics, which is hereby incorporated by reference.

Support circuits shown on FIG. 9C are used to facilitate communication between CODEC chip U12, DSP/CODEC chip U8, and dual port RAM U11. For example, an 8 kHz clock is used to synchronize the operation of CODEC U12 and DSP/CODEC U8.

The operation of the dual port RAM U11 is controlled both by DSP U8 and main controller chip U17. The dual port operation allows writing into one address while reading from another address in the same chip. Both processors can access the exact same memory locations with the use of a contention protocol such that when one is reading, the other cannot be writing. In the preferred embodiment, dual port RAM chip U11 is part number CYZC131 available from Cyprus Semiconductor. This chip includes built-in contention control so that if two processors try to access the same memory location at the same time, the first one making the request gets control of the address location and the other processor must wait. In the preferred embodiment, a circular buffer is arranged in dual port RAM chip U11 comprising 24 bytes. By using a circular buffer configuration with pointers into the buffer area, both processors will not have a contention problem.

The DSP RAM chips U6 and U7 are connected to the DSP chip U8 and also connected through the data and address buses to the Zilog microcontroller U17. In this configuration, the main controller can download the control programs for DSP U8 into DSP RAMs U6 and U7. In this fashion, DSP control can be changed by the main controller or the operating programs on the personal computer, described more fully below. The control programs stored in DSP chips U6 and U7 originate in the flash PEROM chips U16 and U17. The power-up control routine operating on controller chip U17 downloads the DSP control routines into DSP RAM chips U6 and U7.

The interface between the main controller circuit 313 and the personal computer is through SIO circuit 314 and RS232 serial interface 315. These interfaces are described more fully in conjunction with the detailed electrical schematic diagrams of FIGS. 6A 6C. RS232 connection J1 is shown on FIG. 6A with the associated control circuit and interface circuitry used to generate and receive the appropriate RS232 standard signals for a serial communications interface with a personal computer. FIG. 6B is a detailed electrical schematic diagram showing the generation of various voltages for powering the hardware components of the electrical schematic diagrams of hardware components 20. The power for the present hardware components is received on connector J5 and controlled by power switch S34. From this circuitry of FIG. 6B, plus and minus 12 volts, plus five volts and minus five volts are derived for operating the various RAM chips, controller chips, and support circuitry of the present system. FIG. 6C shows the interconnection of the status LEDs found on the front display of the box 20.

Finally, the "glue logic" used to support various functions in the hardware components 20 are described in conjunction with the detailed electrical schematic diagrams of FIGS. 10A 10C. The connections between FIGS. 10A and 10C and the previous schematic diagrams is made via the labels for each of the lines. For example, the LED status lights are controlled and held active by direct addressing and data control of latches GA1 and GA2. For a more detailed description of the connection of the glue logic of FIGS. 10A 10C, the gate array U19 is shown connected in FIGS. 5A and 5B.

Packet Protocol Between the PC and the Hardware Component

A special packet protocol is used for communication between the hardware components 20 and the personal computer (PC) 10. The protocol is used for transferring different types of information between the two devices such as the transfer of DATA, VOICE, and QUALIFIED information. The protocol also uses the BREAK as defined in CCITT X.28 as a means to maintain protocol synchronization. A description of this BREAK sequence is also described in the Statutory Invention Registration entitled, "Escape Methods for Modem Communications," to Timothy D. Gunn, filed Jan. 8, 1993, which is hereby incorporated by reference.

The protocol has two modes of operation. One mode is packet mode, and the other is stream mode. The protocol allows mixing of different types of information into the data stream without having to physically switch modes of operation. The hardware component 20 will identify the packet received from the computer 10 and perform the appropriate action according to the specifications of the protocol. If it is a data packet, then the controller 313 of hardware component 20 would send it to the data pump circuit 311. If the packet is a voice packet, then the controller 313 of hardware component 20 would distribute that information to the Voice DSP 306. This packet transfer mechanism also works in the reverse, where the controller 313 of hardware component 20 would give different information to the computer 10 without having to switch into different modes. The packet protocol also allows commands to be sent to either the main controller 313 directly or to the Voice DSP 306 for controlling different options without having to enter a command state.

Packet mode is made up of 8-bit asynchronous data and is identified by a beginning synchronization character (01 hex) followed by an ID/LI character and then followed by the information to be sent. In addition to the ID/LI character codes defined below, those skilled in the art will readily recognize that other ID/LI character codes could be defined to allow for additional types of packets, such as video data, or alternate voice compression algorithm packets, such as Codebook Excited Linear Predictive Coding (CELP) algorithm, GSM, RPE, VSELP, etc.

Stream mode is used when large amounts of one type of packet (VOICE, DATA, or QUALIFIED) is being sent. The transmitter tells the receiver to enter stream mode by a unique command. Thereafter, the transmitter tells the receiver to terminate stream mode by using the BREAK command followed by an "AT" type command. The command used to terminate the stream mode can be a command to enter another type of stream mode or it can be a command to enter back into packet mode.

Currently, there are three types of packets used: DATA, VOICE, and QUALIFIED. Table 1 shows the common packet parameters used for all three packet types. Table 2 shows the three basic types of packets with the sub-types listed.

TABLE-US-00001 TABLE 1 Packet Parameters 1. Asynchronous transfer 2. 8 bits, no parity 3. Maximum packet length of 128 bytes IDentifier byte = 1 InFormation = 127 4. SPEED variable from 9600 to 57600 default to 19200

TABLE-US-00002 TABLE 2 Packet Types 1. Data 2. Voice 3. Qualified: a. COMMAND b. RESPONSE c. STATUS d. FLOW CONTROL e. BREAK f. ACK g. NAK h. STREAM

A Data Packet is shown in Table 1 and is used for normal data transfer between the controller 313 of hardware component 20 and the computer 10 for such things as text, file transfers, binary data, and any other type of information presently being sent through modems. All packet transfers begin with a synch character 01 hex (synchronization byte). The Data Packet begins with an ID byte which specifies the packet type and packet length. Table 3 describes the Data Packet byte structure, and Table 4 describes the bit structure of the ID byte of the Data Packet. Table 5 is an example of a Data Packet with a byte length of 6. The value of the LI field is the actual length of the data field to follow, not counting the ID byte.

TABLE-US-00003 TABLE 3 Data Packet Byte Structure byte 1 = 01h (sync byte) byte 2 = ID/LI (ID byte/length indicator) bytes 3 127 = data (depending on LI) ##STR00001##

TABLE-US-00004 TABLE 4 ID Byte of Data Packet Bit 7 identifies the type of packet Bits 6-0 contain the LI or length indicator portion of the ID byte ##STR00002##

TABLE-US-00005 TABLE 5 Data Packet Example LI (length indicator) = 6 ##STR00003##

The Voice Packet is used to transfer compressed VOICE messages between the controller 313 of hardware component 20 and the computer 10. The Voice Packet is similar to the Data Packet except for its length, which is, in the preferred embodiment, currently fixed at 23 bytes of data. Once again, all packets begin with a synchronization character chosen in the preferred embodiment to be 01 hex (01H). The ID byte of the Voice Packet is completely a zero byte: all bits are set to zero. Table 6 shows the ID byte of the Voice Packet and Table 7 shows the Voice Packet byte structure.

TABLE-US-00006 TABLE 6 ID Byte of Voice Packet ##STR00004##

TABLE-US-00007 TABLE 7 Voice Packet Byte Structure LI (length indicator) = 0 23 bytes of data ##STR00005##

The Qualified Packet is used to transfer commands and other non-data/voice-related information between the controller 313 of hardware component 20 and the computer 10. The various species or types of the Qualified Packets are described below and are listed above in Table 2. Once again, all packets start with a synchronization character chosen in the preferred embodiment to be 01 hex (01H). A Qualified Packet starts with two bytes, where the first byte is the ID byte and the second byte is the QUALIFIER type identifier. Table 8 shows the ID byte for the Qualified Packet, Table 9 shows the byte structure of the Qualified Packet, and Tables 10 12 list the Qualifier Type byte bit maps for the three types of Qualified Packets.

TABLE-US-00008 TABLE 8 ID Byte of Qualified Packet ##STR00006##

The Length Identifier of the ID byte equals the amount of data which follows, including the QUALIFIER byte (QUAL byte+DATA). If LI=1, then the Qualifier Packet contains the Q byte only.

TABLE-US-00009 TABLE 9 Qualifier Packet Byte Structure ##STR00007##

The bit maps of the Qualifier Byte (QUAL BYTE) of the Qualified Packet are shown in Tables 10 12. The bit map follows the pattern, whereby if the QUAL byte=0, then the command is a break. Also, bit 1 of the QUAL byte designates ack/nak, bit 2 designates flow control, and bit 6 designates stream mode command. Table 10 describes the Qualifier Byte of Qualified Packet, Group 1, which are immediate commands. Table 11 describes the Qualifier Byte of Qualified Packet, Group 2, which are stream mode commands in that the command is to stay in the designated mode until a BREAK+INIT command string is sent. Table 12 describes the Qualifier Byte of Qualified Packet, Group 3, which are information of status commands.

TABLE-US-00010 TABLE 10 Qualifier Byte of Qualified Packet: Group 1 7 6 5 4 3 2 1 0 x x x x x x x x 0 0 0 0 0 0 0 0 = break 0 0 0 0 0 0 1 0 = ACK 0 0 0 0 0 0 1 1 = NAK 0 0 0 0 0 1 0 0 = xoff or stop sending data 0 0 0 0 0 1 0 1 = xon or resume sending data 0 0 0 0 1 0 0 0 = cancel fax

TABLE-US-00011 TABLE 11 Qualifier Byte of Qualified Packet: Group 2 7 6 5 4 3 2 1 0 x x x x x x x x 0 1 0 0 0 0 0 0 = stream command mode 0 1 0 0 0 0 1 0 = stream data 0 1 0 0 0 0 1 1 = stream voice 0 1 0 0 0 1 0 0 = stream video 0 1 0 0 0 1 0 1 = stream A 0 1 0 0 0 1 1 0 = stream B 0 1 0 0 0 1 1 1 = stream C

The Qualifier Packet indicating stream mode and BREAK attention is used when a large amount of information is sent (voice, data, . . . ) to allow the highest throughput possible. This command is mainly intended for use in DATA mode but can be used in any one of the possible modes. To change from one mode to another, break-init sequence would be given. A break "AT . . . <cr>" type command would cause a change in state and set the serial rate from the "AT" command.

TABLE-US-00012 TABLE 12 Qualifier Byte of Qualified Packet: Group 3 7 6 5 4 3 2 1 0 x x x x x x x x 1 0 0 0 0 0 0 0 = commands 1 0 0 0 0 0 0 1 = responses 1 0 0 0 0 0 1 0 = status

Cellular Supervisory Packet

In order to determine the status of the cellular link, a supervisor packet (shown in Table 13) is used. Both sides of the cellular link will send the cellular supervisory packet every 3 seconds. Upon receiving the cellular supervisory packet, the receiving side will acknowledge it using the ACK field of the cellular supervisory packet. If the sender does not receive an acknowledgement within one second, it will repeat sending the cellular supervisory packet up to 12 times. After 12 attempts of sending the cellular supervisory packet without an acknowledgement, the sender will disconnect the line. Upon receiving an acknowledgment, the sender will restart its 3 second timer. Those skilled in the art will readily recognize that the timer values and wait times selected here may be varied without departing from the spirit or scope of the present invention.

TABLE-US-00013 TABLE 13 Cellular Supervisory Packet Byte Structure ##STR00008##

Speech Compression

The Speech Compression algorithm described above for use in the voice mail function, the multi-media mail function, and the show-and-tell function of the present system is all accomplished via the voice control circuit 306. Referring once again to FIG. 3, the user is talking either through the handset, the headset, or the microphone/speaker telephone interface. The analog voice signals are received and digitized by the telephone CODEC circuit 305. The digitized voice information is passed from the digital telephone CODEC circuit 305 to the voice control circuits 306. The digital signal processor (DSP) of the voice control circuit 306 is programmed to do the voice compression algorithm. The source code programmed into the voice control DSP is attached in the microfiche appendix. The DSP of the voice control circuit 306 compresses the speech and places the compressed digital representations of the speech into special packets described more fully below. As a result of the voice compression algorithm, the compressed voice information is passed to the dual port RAM circuit 308 for either forwarding and storage on the disk of the personal computer via the RS232 serial interface or for multiplexing with conventional modem data to be transmitted over the telephone line via the telephone line interface circuit 309 in the voice over data mode of operation show-and-tell function 123.

Speech Compression Algorithm

To multiplex high-fidelity speech with digital data and transmit both over the over the telephone line, a high available bandwidth would normally be required. In the present invention, the analog voice information is digitized into 8-bit PCM data at an 8 kHz sampling rate producing a serial bit stream of 64,000 bps serial data rate. This rate cannot be transmitted over the telephone line. With the Speech Compression algorithm described below, the 64 kbs digital voice data is compressed into a 9200 bps encoding bit stream using a fixed-point (non-floating point) DSP such that the compressed speech can be transmitted over the telephone line using a 9600 baud modem transmission. This is an approximately 7:1 compression ratio. This is accomplished in an efficient manner such that enough machine cycles remain during real-time speech compression to allow real-time acoustic and line echo cancellation in the same fixed-point DSP.

Even at 9200 bps serial data rate for voice data transmission, this bit rate leaves little room for concurrent conventional data transmission. A silence detection function is used to detect quiet intervals in the speech signal and substitute conventional data packets in lieu of voice data packets to effectively time multiplex the voice and data transmission. The allocation of time for conventional data transmission is constantly changing depending upon how much silence is on the voice channel.

The voice compression algorithm of the present system relies on a model of human speech which shows that human speech contains redundancy inherent in the voice patterns. Only the incremental innovations (changes) need to be transmitted. The algorithm operates on 160 digitized speech samples (20 milliseconds), divides the speech samples into time segments of 5 milliseconds each, and uses predictive coding on each segment. With this algorithm, the current segment is predicted as best as possible, based on the past recreated segments, and a difference signal is determined. The difference value is compared to the stored difference values in a look-up table or code book, and the address of the closest value is sent to the remote site along with the predicted gain and pitch values for each segment. In this fashion, four 5 ms speech segments can be reduced to a packet of 23 bytes or 184 bits (46 bits per sample segment). By transmitting 184 bits every 20 milliseconds, an effective serial data transmission rate of 9200 bps is accomplished.

To produce this compression, the present system includes a unique Vector Quantization (VQ) speech compression algorithm designed to provide maximum fidelity with minimum compute power and bandwidth. The VQ algorithm has two major components. The first section reduces the dynamic range of the input speech signal by removing short-term and long-term redundancies. This reduction is done in the waveform domain, with the synthesized part used as the reference for determining the incremental "new" content. The second section maps the residual signal into a code book optimized for preserving the general spectral shape of the speech signal.

FIG. 11 is a high level signal flow block diagram of the speech compression algorithm used in the present system to compress the digitized voice for transmission over the telephone line in the voice over data mode of operation or for storage and use on the personal computer. The transmitter and receiver components are implemented using the programmable voice control DSP/CODEC circuit 306 shown in FIG. 3.

The DC removal stage 1101 receives the digitized speech signal and removes the DC bias by calculating the long-term average and subtracting it from each sample. This ensures that the digital samples of the speech are centered about a zero mean value. The pre-emphasis stage 1103 whitens the spectral content of the speech signal by balancing the extra energy in the low band with the reduced energy in the high band.

The system finds the innovation in the current speech segment by subtracting 1109 the prediction from reconstructed past samples synthesized from synthesis stage 1107. This process requires the synthesis of the past speech samples locally (analysis by synthesis). The synthesis block 1107 at the transmitter performs the same function as the synthesis block 1113 at the receiver. When the reconstructed previous segment of speech is subtracted from the present segment (before prediction), a difference term is produced in the form of an error signal. This residual error is used to find the best match in the code book 1105. The code book 1105 quantizes the error signal using a code book generated from a representative set of speakers and environments. A minimum mean squared error match is determined in 5 ms segments. In addition, the code book is designed to provide a quantization error with spectral roll-off (higher quantization error for low frequencies and lower quantization error for higher frequencies). Thus, the quantization noise spectrum in the reconstructed signal will always tend to be smaller than the underlying speech signal.

The channel corresponds to the telephone line in which the compressed speech bits are multiplexed with data bits using a packet format described below. The voice bits are sent in 100 ms packets of 5 frames each, each frame corresponding to 20 ms of speech in 160 samples. Each frame of 20 ms is further divided into 4 sub-blocks or segments of 5 ms each. In each sub-block of the data consists 7 bits for the long-term predictor, 3 bits for the long-term predictor gain, 4 bits for the sub-block gain, and 32 bits for each code book entry, for a total 46 bits each 5 ms. The 32 bits for code book entries consists of four 8-bit table entries in a 256 long code book of 1.25 ms duration. In the code book block, each 1.25 ms of speech is looked up in a 256 word code book for the best match. The 8-bit table entry is transmitted rather than the actual samples. The code book entries are pre-computed from representative speech segments. (See the DSP Source Code in the microfiche appendix.)

On the receiving end 1200, the synthesis block 1113 at the receiver performs the same function as the synthesis block 1107 at the transmitter. The synthesis block 1113 reconstructs the original signal from the voice data packets by using the gain and pitch values and code book address corresponding to the error signal most closely matched in the code book. The code book at the receiver is similar to the code book 1105 in the transmitter. Thus, the synthesis block recreates the original pre-emphasized signal. The de-emphasis stage inverts the pre-emphasis operation by restoring the balance of original speech signal.

The complete speech compression algorithm is summarized as follows: a. Remove any DC bias in the speech signal. b. Pre-emphasize the signal. c. Find the innovation in the current speech segment by subtracting the prediction from reconstructed past samples. This step requires the synthesis of the past speech samples locally (analysis by synthesis) such that the residual error is fed back into the system. d. Quantize the error signal using a code book generated from a representative set of speakers and environments. A minimum mean squared error match is determined in 5 ms segments. In addition, the code book is designed to provide a quantization error with spectral roll-off (higher quantization error for lower frequencies and lower quantization error for higher frequencies). Thus, the quantization noise spectrum in the reconstructed signal will always tend to be smaller than the underlying speech signal. e. At the transmitter and the receiver, reconstruct the speech from the quantized error signal fed into the inverse of the function in step c above. Use this signal for analysis by synthesis and for the output to the reconstruction stage below. f. Use a de-emphasis filter to reconstruct the output.

The major advantages of this approach over other low bit rate algorithms are that there is no need for any complicated calculation of reflection coefficients (no matrix inverse or lattice filter computations). Also, the quantization noise in the output speech is hidden under the speech signal and there are no pitch tracking artifacts: the speech sounds "natural," with only minor increases of background hiss at lower bit rates. The computational load is reduced significantly compared to a VSELP algorithm, and variations of the same algorithm provide bit rates of 8, 9.2, and 16 Kbits/s. The total delay through the analysis section is less than 20 milliseconds in the preferred embodiment. The present algorithm is accomplished completely in the waveform domain and there is no spectral information being computed and there is no filter computations needed.

Detailed Description of the Speech Compression Algorithm

The speech compression algorithm is described in greater detail with reference to FIGS. 11 through 13, and with reference to the block diagram of the hardware components of the present system shown at FIG. 3. Also, reference is made to the detailed schematic diagrams in FIGS. 9A 9C. The voice compression algorithm operates within the programmed control of the voice control DSP circuit 306. In operation, the speech or analog voice signal is received through the telephone interface 301, 302, or 303 and is digitized by the digital telephone CODEC circuit 305. The CODEC for circuit 305 is a companding .mu.-law CODEC. The analog voice signal from the telephone interface is band-limited to about 3,500 Hz and sampled at 8 kHz by digital telephone CODEC 305. Each sample is encoded into 8-bit PCM data producing a serial 64 kb/s signal. The digitized samples are passed to the voice control DSP/CODEC of circuit 306. There, the 8-bit .mu.-law PCM data is converted to 13-bit linear PCM data. The 13-bit representation is necessary to accurately represent the linear version of the logarithmic 8-bit .mu.-law PCM data. With linear PCM data, simpler mathematics may be performed on the PCM data.

The voice control DSP/CODEC of circuit 306 correspond to the single integrated circuit U8 shown in FIGS. 9A and 9B as a WE.RTM. DSP16C Digital Signal Processor/CODEC from AT&T Microelectronics, which is a combined digital signal processor and a linear CODEC in a single chip, as described above. The digital telephone CODEC of circuit 305 corresponds to integrated circuit U12, shown in FIG. 9B as a T7540 companding .mu.-law CODEC.

The sampled and digitized PCM voice signals from the telephone .mu.-law CODEC U12 shown in FIG. 9B are passed to the voice control DSP/CODEC U8 via direct data lines clocked and synchronized to an 8 kHz clocking frequency. The digital samples are loaded into the voice control DSP/CODEC U8 one at a time through the serial input and stored into an internal queue held in RAM and converted to linear PCM data. As the samples are loaded into the end of the queue in the RAM of the voice control DSP U8, the samples at the head of the queue are operated upon by the voice compression algorithm. The voice compression algorithm then produces a greatly compressed representation of the speech signals in a digital packet form. The compressed speech signal packets are then passed to the dual port RAM circuit 308 shown in FIG. 3 for use by the main controller circuit 313 for either transferring in the voice over data mode of operation or for transfer to the personal computer for storage as compressed voice for functions, such as telephone answering machine message data, for use in the multi-media documents, and the like.

In the voice over data mode of operation, voice control DSP/CODEC circuit 306 of FIG. 3 will be receiving digital voice PCM data from the digital telephone CODEC circuit 305, compressing it, and transferring it to dual port RAM circuit 308 for multiplexing and transfer over the telephone line. This is the transmit mode of operation of the voice control DSP/CODEC circuit 306 corresponding to transmitter block 1100 of FIG. 11 and corresponding to the compression algorithm of FIG. 12.

Concurrent with this transmit operation, the voice control DSP/CODEC circuit 306 is receiving compressed voice data packets from dual port RAM circuit 308, uncompressing the voice data, and transferring the uncompressed and reconstructed digital PCM voice data to the digital telephone CODEC 305 for digital-to-analog conversion and eventual transfer to the user through the telephone interface 301, 302, or 304. This is the receive mode of operation of the voice control DSP/CODEC circuit 306 corresponding to receiver block 1200 of FIG. 11 and corresponding to the decompression algorithm of FIG. 13. Thus, the voice control DSP/CODEC circuit 306 is processing the voice data in both directions in a full-duplex fashion.

The voice control DSP/CODEC circuit 306 operates at a clock frequency of approximately 24.576 Mhz, while processing data at sampling rates of approximately 8 kHz in both directions. The voice compression/decompression algorithms and packetization of the voice data is accomplished in a quick and efficient fashion to ensure that all processing is done in real-time without loss of voice information. This is accomplished in an efficient manner such that enough machine cycles remain in the voice control DSP circuit 306 during real-time speech compression to allow real-time acoustic and line echo cancellation in the same fixed point DSP.

In programmed operation, the availability of an 8-bit sample of PCM voice data from the .mu.-law digital telephone CODEC circuit 305 causes an interrupt in the voice control DSP/CODEC circuit 306 where the sample is loaded into internal registers for processing. Once loaded into an internal register, it is transferred to a RAM address, which holds a queue of samples. The queued PCM digital voice samples are converted from 8-bit .mu.-law data to a 13-bit linear data format using table look-up for the conversion. Those skilled in the art will readily recognize that the digital telephone CODEC circuit 305 could also be a linear CODEC.

Referring to FIG. 11, the digital samples are shown as speech entering the transmitter block 1100. The transmitter block, of course, is the mode of operation of the voice control DSP/CODEC circuit 306 operating to receive local digitized voice information, compress it, and packetize it for transfer to the main controller circuit 313 for transmission on the telephone line. The telephone line connected to telephone line interface 309 of FIG. 3 corresponds to the channel 1111 of FIG. 11.

A frame rate for the voice compression algorithm is 20 milliseconds of speech for each compression. This correlates to 160 samples to process per frame. When 160 samples are accumulated in the queue of the internal DSP RAM, the compression of that sample frame is begun.

The voice control DSP/CODEC circuit 306 is programmed to first remove the DC component 1101 of the incoming speech. The DC removal is an adaptive function to establish a center base line on the voice signal by digitally adjusting the values of the PCM data. The formula for removal of the DC bias or drift is as follows:

.function..function..function..alpha..function..times..times..alpha. ##EQU00001##

The removal of the DC is for the 20 millisecond frame of voice, which amounts to 160 samples. The selection of a is based on empirical observation to provide the best result.

Referring to FIG. 12, the voice compression algorithm in a control flow diagram is shown, which will assist in the understanding of the block diagram of FIG. 11. The analysis and compression begin at block 1201, where the 13-bit linear PCM speech samples are accumulated until 160 samples representing 20 milliseconds of voice or one frame of force is passed to the DC removal portion of code operating within the programmed voice control DSP/CODEC circuit 306. The DC removal portion of the code described above approximates the base line of the frame of voice by using an adaptive DC removal technique.

A silence detection algorithm 1205 is also included in the programmed code of the DSP/CODEC 306. The silence detection function is a summation of the square of each sample of the voice signal over the frame. If the power of the voice frame falls below a pre-selected threshold, this would indicate a silent frame. The detection of a silence frame of speech is important for later multiplexing of the V-data and C-data described below. During silent portions of the speech, the main controller circuit 313 will transfer conventional digital data (C-data) over the telephone line in lieu of voice data (V-data). The formula for computing the power is:

.times..times..times. ##EQU00002##

If the power PWR is lower than a pre-selected threshold, then the present voice frame is flagged as containing silence (see Table 15). The 160-sample silent frame is still processed by the voice compression algorithm; however, the silent frame packets are discarded by the main controller circuit 313 so that digital data may be transferred in lieu of voice data.

The rest of the voice compression is operated upon in segments where there are four segments per frame amounting to 40 samples of data per segment. It is only the DC removal and silence detection which is accomplished over an entire 20 millisecond frame. The pre-emphasis 1207 of the voice compression algorithm, shown in FIG. 12, is the next step. The formula for the pre-emphasis is: S(n)=S(n)-.tau.*S(n-1), where.tau.=0.55

Each segment thus amounts to five milliseconds of voice, which is equal to 40 samples. Pre-emphasis then is done on each segment. The selection of T is based on empirical observation to provide the best result.

The pre-emphasis essentially flattens the signal by reducing the dynamic range of the signal. By using pre-emphasis to flatten the dynamic range of the signal, less of a signal range is required for compression, making the compression algorithm operate more efficiently.

The next step in the speech compression algorithm is the long-term predictor (LTP). The long-term prediction is a method to detect the innovation in the voice signal. Since the voice signal contains many redundant voice segments, we can detect these redundancies and only send information about the changes in the signal from one segment to the next. This is accomplished by comparing the linear PCM data of the current segment on a sample by sample basis to the reconstructed linear PCM data from the previous segments to obtain the innovation information and an indicator of the error in the prediction.

The first step in the long-term prediction is to predict the pitch of the voice segment, and the second step is to predict the gain of the pitch. For each segment of 40 samples, a long-term correlation lag PITCH and associated LTP gain factor .beta..sub.j (where j=0, 1, 2, 3 corresponding to each of the four segments of the frame) are determined at 1209 and 1211, respectively. The computations are done as follows.

From MINIMUM PITCH (40) to MAXIMUM PITCH (120) for indices 40 through 120 (the pitch values for the range of previous speech viewed), the voice control DSP circuit 306 computes the cross-correlation between the current speech segment and the previous speech segment by comparing the samples of the current speech segment against the reconstructed speech samples of the previous speech segment using the following formula:

.times..times..times.'.times. ##EQU00003## where j=40, . . . 120 S=current sample of current segment S'=past sample of reconstructed previous segment n.sub.k=0, 40, 80, 120(the subframe index) and where the best fit is: Sxy=MAX{Sxy(j)}, where j=40, . . . 120.

The value of j for which the peak occurs is the PITCH. This is a 7-bit value for the current segment calculated at 1209. The value of j is an indicator of the delay or lag at which the cross-correlation matches the best between the past reconstructed segment and the current segment. This indicates the pitch of the voice in the current frame. The maximum computed value of j is used to reduce the redundancy of the new segment compared to the previous reconstructed segments in the present algorithm, since the value of j is a measure of how close the current segment is to the previous reconstructed segments.

Next, the voice control DSP circuit 306 computes the LTP gain factor .beta. at 1211 using the following formula, in which Sxy is the current segment and Sxx is the previous reconstructed segment:

.beta..function..function. ##EQU00004## ##EQU00004.2## .times.'.function. ##EQU00004.3##

The value of the LTP gain factor .beta. is a normalized quantity between zero and unity for this segment, where .beta. is an indicator of the correlation between the segments. For example, a perfect sine wave would produce a .beta. which would be close to unity, since the correlation between the current segments and the previous reconstructed segments should be almost a perfect match so the LTP gain factor is one.

The LTP gain factor is quantized from a LTP Gain Table. This table is characterized in Table 14.

TABLE-US-00014 TABLE 14 LTP Gain Quantization ##STR00009##

The gain value of .beta. is then selected from this table depending upon which zone or range .beta..sub.segment was found, as depicted in Table 14. For example, if .beta..sub.segment is equal to 0.45, then .beta. is selected to be 2. This technique quantizes the .beta. into a 3-bit quantity.

Next, the LTP (Long-Term Predictor) filter function 1213 is computed. The pitch value computed above is used to perform the long-term analysis filtering to create an error signal e(n). The normalized error signals will be transmitted to the other site as an indicator of the original signal on a per sample basis. The filter function for the current segment is as follows: e(n)=S(n)-.beta.*S'(n-pitch) where n=0, 1, . . . 39

Next, the code book search and vector quantization function 1215 is performed. First, the voice control DSP circuit 306 computes the maximum sample value in the segment with the formula: GAIN=MAX{|e(n)|} where n=0, 1, . . . 39

This gain is different than the LTP gain. This gain is the maximum amplitude in the segment. This gain is quantized using the GAIN table described in the DSP Source Code attached in the microfiche appendix. Next, the voice control DSP circuit 306 normalizes the LTP filtered speech by the quantized GAIN value by using the maximum error signal |e(n)| (absolute value for e(n)) for the current segment and dividing this into every sample in the segment to normalize the samples across the entire segment. Thus, the e(n) values are all normalized to have values between zero and one, using the following: e(n)=e(n)/GAIN n=0 . . . 39

Each segment of 40 samples is comprised of four subsegments of 10 samples each. The voice control DSP circuit 306 quantizes 10 samples of e(n) with an index into the code book. The code book consists of 256 entries (256 addresses) with each code book entry consisting of 10 sample values. Every entry of 10 samples in the code book is compared to the 10 samples of each subsegment. Thus, for each subsegment, the code book address or index is chosen based on a best match between the 10-sample subsegment and the closest 10-sample code book entry. The index chosen has the least difference, according to the following minimization formula:

.times..times. ##EQU00005## where x.sub.l=the input vector of 10 samples, and y.sub.l=the code book vector of 10 samples

This comparison to find the best match between the subsegment and the code book entries is computationally intensive. A brute force comparison may exceed the available machine cycles if real-time processing is to be accomplished. Thus, some shorthand processing approaches are taken to reduce the computations required to find the best fit. The above formula can be computed in a shorthand fashion by pre-computing and storing some of the values of this equation. For example, by expanding out the above formula, some of the unnecessary terms may be removed and some fixed terms may be pre-computed:

.times..times..times..times. ##EQU00006## where x.sub.l.sup.2 is a constant so it may be dropped from the formula, and where the value of 1/2.SIGMA.y.sub.l.sup.2 may be pre-computed and stored as the eleventh value in the code book so that the only real-time computation involved is the following formula:

.times..times..times..times. ##EQU00007##

Thus, for a segment of 40 samples, we will transmit 4 code book indexes corresponding to 4 subsegments of 10 samples each. After the appropriate index into the code book is chosen, the LTP filtered speech samples are replaced with the code book samples. These samples are then multiplied by the quantized GAIN in block 1217.

Next, the inverse of the LTP filter function is computer at 1219: e(n)=e(n)=.beta.*S'(n-pitch)n=0, . . . , 39 S'(i)=S'(n)n=40, . . . 120; i=0, . . . (120 40) S'(i)=e(i)i=0, . . . 40

The voice is reconstructed at the receiving end of the voice over data link according to the reverse of the compression algorithm as shown as the decompression algorithm in FIG. 13. The synthesis of FIG. 13 is also performed in the compression algorithm of FIG. 12 since the past segment must be synthesized to predict the gain and pitch of the current segment.

Echo Cancellation Algorithm

The use of the speaker 304 and the microphone 303 necessitates the use of an acoustical echo cancellation algorithm to prevent feedback from destroying the voice signals. In addition, a line echo cancellation algorithm is needed no matter which telephone interface 301, 302, or 303/304 is used. The echo cancellation algorithm used is an adaptive echo canceller which operates in any of the modes of operation of the present system whenever the telephone interface is operational. In particular, the echo canceller is operational in a straight telephone connection and it is operational in the voice over data mode of operation.

In the case of a straight telephone voice connection between the telephone interface 301, 302, 303/304, and the telephone line interface 309 in communication with an analog telephone on the other end, the digitized PCM voice data from digital telephone CODEC 305 is transferred through the voice control DSP/CODEC circuit 306 where it is processed in the digital domain and converted back from a digital form to an analog form by the internal linear CODEC of voice control DSP/CODEC circuit 306. Since digital telephone CODEC circuit 305 is a .mu.-law CODEC and the internal CODEC to the voice control DSP/CODEC circuit 306 is a linear CODEC, a .mu.-law to linear conversion must be accomplished by the voice control DSP/CODEC circuit 306.

In addition, the sampling rate of digital telephone CODEC 305 is slightly less than the sampling rate of the linear CODEC of voice control DSP/CODEC circuit 306, so a slight sampling conversion must also be accomplished. The sampling rate of digital telephone .mu.-law CODEC 305 is 8000 samples per second, and the sampling rate of the linear CODEC of voice control DSP/CODEC circuit 306 is 8192 samples per second.

Referring to FIG. 14, in conjunction with FIG. 3, the speech or analog voice signal is received through the telephone interface 301, 302, or 303 and is digitized by the digital telephone CODEC circuit 305 in an analog-to-digital conversion 1401. The CODEC for circuit 305 is a companding .mu.-law CODEC. The analog voice signal from the telephone interface is band-limited to about 3,500 Hz and sampled at 8 kHz with each sample encoded into 8-bit PCM data producing a serial 64 kb/s signal. The digitized samples are passed to the voice control DSP of circuit 306 where they are immediately converted to 13-bit linear PCM samples.

Referring again to FIG. 14, the PCM digital voice data y(n) from telephone CODEC circuit 305 is passed to the voice control DSP/CODEC circuit 306 where the echo estimate signal y(n) in the form of digital data is subtracted from it. The subtraction is done on each sample on a per sample basis.

Block 1405 and 1421 are gain control blocks g.sub.m and g.sub.s, respectively. These digital gain controls are derived from tables for which the gain of the signal may be set to different levels depending upon the desired level for the voice signal. These gain levels can be set by the user through the level controls in the software, as shown in FIG. 49. The gain on the digitized signal is set by multiplying a constant to each of the linear PCM samples.

In an alternate embodiment, the gain control blocks g.sub.m and g.sub.s may be controlled by sensing the level of the speaker's voice and adjusting the gain accordingly. This automatic gain control facilitates the operation of the silence detection described above to assist in the time allocation between multiplexed data and voice in the voice over data mode of operation.

In voice over data mode, the output of gain control block g.sub.m is placed in a buffer for the voice compression/decompression algorithm 1425 instead of sample rate converter 1407. The samples in this mode are accumulated, as described above, and compressed for multiplexing and transmission by the main controller 313. Also in voice over data mode, the gain control block 1421 receives decompressed samples from the voice compression/decompression algorithm 1425 instead of sample rate converter 1423 for output.

The echo canceller of FIG. 14 uses a least mean square (LMS) method of adaptive echo cancellation. The echo estimate signal subtracted from the incoming signal at 1403 is determined by function 1411. Function 1411 is a FIR (finite impulse response) filter having in the preferred embodiment an impulse response which is approximately the length of delay through the acoustic path. The coefficients of the FIR filter are modeled and tailored after the acoustic echo path of the echo, taking into account the specific physical attributes of the box that the speaker 304 and microphone 303 are located in and the proximity of the speaker 304 to the microphone 303. Thus, any signal placed on to the speaker is sent through the echo cancellation function 1411 to be subtracted from the signals received by the microphone 303 after an appropriate delay to match the delay in the acoustic path. The formula for echo replication of function box 1411 is:

.times..times..times..times..times. ##EQU00008## and the result of the subtraction of the echo cancellation signal y(n) from the microphone signal y(n) is: e(n)=y(n)-y(n).

The LMS coefficient function 1413 provides adaptive echo cancellation coefficients for the FIR filter of 1411. The signal is adjusted based on the following formula:

.function..function..beta..times..function..times..times..function..times- ..function. ##EQU00009## .times..times..times. .times..times..times..times..times..times..times..times..beta. ##EQU00009.2##

The echo cancellation of functions 1415 and 1417 are identical to the functions of 1413 and 1411, respectively. The functions 1407 and 1423 of FIG. 14 are sample rate conversions, as described above, due to the different sampling rates of the digital telephone CODEC circuit 305 and the voice control CODEC of circuit 306.

Voice Over Data Packet Protocol

As described above, the present system can transmit voice data and conventional data concurrently by using time multiplex technology. The digitized voice data, called V-data, carries the speech information. The conventional data is referred to as C-data. The V-data and C-data multiplex transmission is achieved in two modes at two levels: the transmit and receive modes, and data service level and multiplex control level. This operation is shown diagrammatically in FIG. 15.

In transmit mode, the main controller circuit 313 of FIG. 3 operates in the data service level 1505 to collect and buffer data from both the personal computer 10 (through the RS232 port interface 315) and the voice control DSP 306. In multiplex control level 1515, the main controller circuit 313 multiplexes the data and transmits that data out over the phone line 1523. In the receive mode, the main controller circuit 313 operates in the multiplex control level 1515 to de-multiplex the V-data packets and the C-data packets, and then operates in the data service level 1505 to deliver the appropriate data packets to the correct destination: the personal computer 10 for the C-data packets, or the voice control DSP circuit 306 for the V-data.

Transmit Mode

In transmit mode, there are two data buffers--the V-data buffer 1511 and the C-data buffer 1513--implemented in the main controller RAM 316 and maintained by main controller 313. When the voice control DSP circuit 306 engages voice operation, it will send a block of V-data every 20 ms to the main controller circuit 313 through dual port RAM circuit 308. Each V-data block has one sign byte as a header and 23 bytes of V-data, as described in Table 15 below.

TABLE-US-00015 TABLE 15 Compressed Voice Packet Structure ##STR00010## ##STR00011## Where P.sub.n = pitch (7 bits) where n = subframe number .beta..sub.n.sup.m = Beta (3 bits) G.sub.n = Gain (4 bits) Vd = Voice data (4 .times. 8 bits) Effective Bit Rate = 184 bits/20 msec = 9200 bps

The sign byte header is transferred every frame from the voice control DSP to the controller 313. The sign byte header contains the sign byte which identifies the contents of the voice packet. The sign byte is defined as follows: 00 hex=the following V-data contains silent sound 01 hex=the following V-data contains speech information

If the main controller 313 is in transmit mode for V-data/C-data multiplexing, the main controller circuit 313 operates at the data service level to perform the following tests. When the voice control DSP circuit 306 starts to send the 23-byte V-data packet through the dual port RAM to the main controller circuit 313, the main controller will check the V-data buffer to see if the buffer has room for 23 bytes. If there is sufficient room in the V-data buffer, the main controller will check the sign byte in the header preceding the V-data packet. If the sign byte is equal to one (indicating voice information in the packet), the main controller circuit 313 will put the following 23 bytes of V-data into the V-data buffer and clear the silence counter to zero. Then, the main controller 313 sets a flag to request that the V-data be sent by the main controller at the multiplex control level.

If the sign byte is equal to zero (indicating silence in the V-data packet), the main controller circuit 313 will increase the silence counter by 1 and check if the silence counter has reached 5. When the silence counter reaches 5, the main controller circuit 313 will not put the following 23 bytes of V-data into the V-data buffer and will stop increasing the silence counter. By this method, the main controller circuit 313 operating at the service level will only provide non-silence V-data to the multiplex control level, while discarding silence V-data packets and preventing the V-data buffer from being overwritten.

The operation of the main controller circuit 313 in the multiplex control level is to multiplex the V-data and C-data packets and transmit then through the same channel. At this control level, both types of data packets are transmitted by the HDLC protocol in which data is transmitted in synchronous mode and checked by CRC error checking. If a V-data packet is received at the remote end with a bad CRC, it is discarded, since 100% accuracy of the voice channel is not ensured. If the V-data packets were resent in the event of corruption, the real-time quality of the voice transmission would be lost. In addition, the C-data is transmitted following a modem data communication protocol, such as CCITT V.42.

In order to identify the V-data block to assist the main controller circuit 313 to multiplex the packets for transmission at this level and to assist the remote site in recognizing and de-multiplexing the data packets, a V-data block is defined, which includes a maximum of five V-data packets. The V-data block size and the maximum number of blocks are defined as follows: The V-data block header=80h; The V-data block size=23; The maximum V-data block size=5;

The V-data block has higher priority to be transmitted than C-data to ensure the integrity of the real-time voice transmission. Therefore, the main controller circuit 313 will check the V-data buffer first to determine whether it will transmit V-data or C-data blocks. If V-data buffer has V-data of more than 69 bytes, a transmit block counter is set to 5 and the main controller circuit 313 starts to transmit V-data from the V-data buffer through the data pump circuit 311 onto the telephone line. Since the transmit block counter indicates 5 blocks of V-data will be transmitted in a continuous stream, the transmission will stop either at finish of the 115 bytes of V-data, or if the V-data buffer is empty. If V-data buffer has V-data with a number more than 23 bytes, the transmit block counter is set to 1 and starts transmitting V-data. This means that the main controller circuit will only transmit one block of V-data. If the V-data buffer has V-data with less than 23 bytes, the main controller circuit services the transmission of C-data.

During the transmission of a C-data block, the V-data buffer condition is checked before transmitting the first C-data byte. If the V-data buffer contains more than one V-data packet, the current transmission of the C-data block will be terminated in order to handle the V-data.

Receive Mode

On the receiving end of the telephone line, the main controller circuit 313 operates at the multiplex control level to de-multiplex received data to V-data and C-data. The type of block can be identified by checking the first byte of the incoming data blocks. Before receiving a block of V-data, the main controller circuit 313 will initialize a receive V-data byte counter, a back-up pointer, and a temporary V-data buffer pointer. The value of the receiver V-data byte counter is 23, the value of the receive block counter is 0, and the back-up pointer is set to the same value as the V-data receive buffer pointer. If the received byte is not equal to 80 hex (80h indicating a V-data packet), the receive operation will follow the current modem protocol since the data block must contain C-data. If the received byte is equal to 80h, the main controller circuit 313 operating in receive mode will process the V-data. For a V-data block received, when a byte of V-data is received, the byte of V-data is put into the V-data receive buffer, the temporary buffer pointer is increased by 1, and the receive V-data counter is decreased by 1. If the V-data counter is down to zero, the value of the temporary V-data buffer pointer is copied into the back-up pointer buffer. The value of the total V-data counter is added with 23, and the receive V-data counter is reset to 23. The value of the receive block counter is increased by 1. A flag to request service of V-data is then set. If the receive block counter has reached 5, the main controller circuit 313 will not put the incoming V-data into the V-data receive buffer but throw it away. If the total V-data counter has reached its maximum value, the receiver will not put the incoming V-data into the V-data receive buffer but throw it away.

At the end of the block, which is indicated by receipt of the CRC check bytes, the main controller circuit 313 operating in the multiplex control level will not check the result of the CRC but instead will check the value of the receive V-data counter. If the value is zero, the check is finished; otherwise, the value of the back-up pointer is copied back into the current V-data buffer pointer. By this method, the receiver is insured to de-multiplex the V-data from the receiving channel 23 bytes at a time. The main controller circuit 313 operating at the service level in the receive mode will monitor the flag of request service of V-data. If the flag is set, the main controller circuit 313 will get the V-data from the V-data buffer and transmit it to the voice control DSP circuit 306 at a rate of 23 bytes at a time. After sending a block of V-data, it decreases 23 from the value in the total V-data counter.

User Interface Description

The hardware components of the present system are designed to be controlled by an external computing device, such as a personal computer. As described above, the hardware components of the present system may be controlled through the use of special packets transferred over the serial line interface between the hardware components and the personal computer. Those skilled in the art will readily recognize that the hardware components of the present systems may be practiced independent of the software components of the present systems and that the preferred software description described below is not to be taken in a limiting sense.

The combination of the software components and hardware components described in the present patent application may conveniently be referred to as a Personal Communications System (PCS). The present system provides for the following functions:

1. The control and hands-off operation of a telephone with a built-in speaker and microphone.

2. Allowing the user to create outgoing voice mail messages with a voice editor, and logging incoming voice mail messages with a time and date stamp.

3. Creating queues for outgoing faxes, including providing the ability for a user to send faxes from unaware applications through a print command; also allowing the user to receive faxes and log incoming faxes with a time and date stamp.

4. Allowing a user to create multi-media messages with the message composer. The message can contain text, graphics, pictures, and sound segments. A queue is created for the outgoing multi-media messages, and any incoming multi-media messages are logged with a time and date stamp.

5. Providing a way for a user to have a simultaneous data and voice connection over a single communication line.

6. Providing terminal emulation by invoking an external terminal emulation program.

7. Providing address book databases for all outbound calls and queues for the telephone, voice mail, fax manager, multi-media mail, and show-and-tell functions. A user may also search through the database using a dynamic pruning algorithm keyed on order insensitive matches.

FIG. 16 shows the components of a computer system that may be used with the PCS. The computer includes a keyboard 101 by which a user may input data into a system, a computer chassis 103 which holds electrical components and peripherals, a screen display 105 by which information is displayed to the user, and a pointing device 107, typically a mouse, with the system components logically connected to each other via internal system bus within the computer. The PCS software runs on a central processing unit 109 within the computer.

FIG. 17 reveals the high-level structure of the PCS software. A main menu function 111 is used to select the following subfunctions: set-up 113, telephone 115, voice mail 117, fax manager 119, multi-media mail 121, show-and-tell 123, terminal 125, and address book 127.

The preferred embodiment of the present system currently runs under Microsoft Windows.RTM. software running on an IBM.RTM. personal computer or compatible. However, it will be recognized that other implementations of the present inventions are possible on other computer systems and windowing software without loss of scope or generality.

FIG. 18 describes the control structure of the main menu 111 in greater detail. A timer 131 sends a timing signal to a control block 129 in order to make the control block 129 active substantially once every 10 seconds. It will be recognized that other timing intervals may be used, as appropriate to the windowing system being used, without loss of generality. A status 133 is used to preclude other applications or program blocks from taking control of a communications port by indicating that the port is currently being used by the PCS. The controller 129 looks at all outbound queues in voice mail 117, fax manager 119, and multi-media mail 121, and if there is an outgoing message in one of the outbound queues, initiates a dispatch. A signal is then sent to the status box in order to preclude other applications or program blocks from using the serial communications port.

The control block 129 also monitors incoming calls and invokes the appropriate program block, either voice mail 117, fax manager 119, multi-media mail 121, or show-and-tell 123, in order to further process the incoming call. Additionally, the control block 129 is used to invoke telephone functions 115, terminal emulation functions 125, and allow users to edit the database of addresses with the address book function 127. The control block 129 further provides for the initialization of PCS parameters via the set-up function 113. The main menu, as it is displayed to the user, is shown in FIG. 2.

FIG. 19 illustrates the structure of control block 129. Upon selecting the set-up function 113, the user has access to initialization functions 135 which include serial port, answer mode, hold call, voice mail, PBX, fax, multi-media mail, and show-and-tell initializations. Upon selecting telephone 115, the user has access to telephone functions 137 which include equipment select, volume control, and call functions, as shown in the screen display of FIG. 49. Upon selecting voice mail 117, voice mail functions 139 are provided which include a voice editor, voice messages to be sent, and voice messages received, as shown in the screen display of FIG. 50. Upon selecting fax manager 119, fax manager functions 141 are provided which include set-up functions, faxes to be sent, and faxes to be received, as shown in the screen display of FIG. 52. If multi-media mail 121 is selected, multi-media mail functions 143 are provided which include the set-up function, multi-media messages to be sent, and multi-media messages received functions, as illustrated by the screen display shown in FIG. 53. If show-and-tell 123 is selected, show-and-tell functions 145 are provided to the user which include open, select new, and help, as illustrated in the screen display of FIG. 54. If the terminal function 125 is selected by the user, the terminal emulation function 147 is provided to the user via a terminal emulation block. If address book 127 is selected, address book functions 149 are provided to the user which include file functions, edit functions, and help functions, as shown in the screen display of FIG. 55.

The set-up functions 135 are accessed by an initialization menu, as shown in the screen display of FIG. 40. The PCS software provides support for any communications port that is contained within the personal computer. FIG. 41 shows the screen display shown to a user to enable a user to select a specific communications port to be used by the PCS software. A user may also specify what action is to be taken by the PCS software when an incoming call arrives. The screen display of FIG. 42 is shown to the user, while FIG. 20 describes the full control of the answer mode set-up initialization procedure. Upon selecting answer mode set-up 151, the user is presented with eight answer mode choices 153 and described as follows:

1. PCS does not answer. The PCS software does not answer an incoming call and the telephone equipment acts as normal.

2. Voice mail. The PCS software answers the incoming call and acts as an answering machine to record messages.

3. Fax. The PCS software answers the incoming calls and acts as a fax machine.

4. Multi-media mail. The PCS software answers the incoming call and receives multi-media mail that is being sent by a remote caller.

5. Show-and-tell. The PCS software enables simultaneous data and voice communication using the same communication line.

6. Terminal. The PCS provides terminal emulation through a terminal emulation block, or optionally transfers control to a third-party terminal emulation program.

7. Automatic. The incoming call is analyzed and the appropriate mode is automatically entered.

The user may additionally enter a numeric value which represents the number of rings to wait before the PCS software answers an incoming call.

If, from the set-up functions 135, the hold call function is selected, the hold call display, as illustrated in FIG. 43, is shown to the user, who may then enter a numeric value to specify the number of hours all outgoing calls are to be held. If, at set-up functions 135, the voice mail set-up option is selected, the screen display, as illustrated in FIG. 44, is displayed to the user, who may then enter a file name to be used as a greeting file for all incoming calls. If, at set-up functions 135, the PBX set-up function is selected, the screen display of FIG. 45 is shown to the user, who may then enter a dialing prefix to be applied to any outgoing telephone number. This provides for an easy way to use a list of telephone numbers with an in-house PBX system without making modifications to the telephone number list. If, at set-up functions 135, the fax set-up function is selected, the screen display of FIG. 46 is displayed to the user, who may then enter a numeric value which represents the number of times to attempt to send a fax before timing out. If, at set-up functions 135, the multi-media mail set-up function is selected, the display of FIG. 47 is shown to the user, who may then enter the name of a file to be used for user information. If, at set-up functions 135, the show-and-tell function is selected, the screen display of FIG. 48 is shown to the user, who may then enter show-and-tell user information.

FIG. 49 shows the telephone control function display as shown to the user. FIG. 21 further illustrates the steps and options 155 available when the telephone control function 115 is selected. The user may use the mouse to select between a speaker phone, a handset, or a headset to be used with the communications device, and may adjust the volume of the speaker with a first logical slider switch, and the gain of the microphone with a second logical slider switch (both slider switches being displayed on the screen). During a call, the user may select to save a telephone number, redial a telephone number, record a call, flash between calls, mute a call, or place a call on hold.

FIG. 22 shows the voice mail functions 157 that are available upon selecting voice mail 117. A user may set-up a voice mail greeting file or may edit a voice mail message by selecting the voice mail editor, as shown in FIG. 50. The PCS software provides for two voice mail queues: the first, for voice mail messages to be sent, and the second, for voice mail messages received. In the send queue, a user may add messages to the queue, listen to messages in the queue, delete messages from the queue, or refresh the send queue display. With the receive queue, a user may listen to messages in the queue, store messages in the queue, delete messages from the queue, forward messages from the queue to another voice mail user, or refresh the queue. The voice mail editor, as shown to the user in FIG. 51, allows the user to select file functions to open or save a voice mail message; edit functions for editing a voice mail message; playing, recording, and pausing a voice mail message; and adjusting the play volume and the record volume. The user may also optionally select between a speaker phone, headset, and handset.

FIG. 52 illustrates the fax manager display as shown to the user. FIG. 23 illustrates the fax manager functions 159 that are available to a user after selecting the fax manager function 119 from the main menu 111. The fax manager function provides for two queues: the first, for faxes to be sent; and the second, for faxes that are received. When reviewing the first queue of faxes to be sent, the user may preview a fax, print a fax, delete a fax from the queue, refresh the send fax queue, or forward a fax to another user. When reviewing the second queue of received faxes, the user may view a fax, print a fax, delete a fax, refresh the received fax queue, forward a fax to another user, or store a fax.

FIG. 53 describes the multi-media mail display that is shown to the user. FIG. 24 describes the multi-media mail functions 161 that are available to a user upon selecting the multi-media mail function 121. Upon selecting multi-media mail, the PCS software provides for set-up, composing a message with the message composer, or allowing the user to view and edit multi-media messages in two queues: the first queue, having multi-media messages to be sent; and the second queue, having multi-media messages that have been received. When reviewing the first queue of send messages, a user may add messages to the queue, preview messages in the queue, delete messages from the queue, or change attributes of messages in the queue. When a user is accessing the second queue of multi-media messages that have been received, the user may view messages, store messages, delete messages, forward messages to another user, or refresh the queue.

FIG. 54 illustrates the display shown to the user upon selecting the show-and-tell function 123 from the main menu 111. FIG. 55 illustrates the display that is shown to the user upon selecting the address book function 127 from the main menu 111. A user may open a previously stored address book file or may edit an existing address book file by adding, deleting, or changing entries that are in the file. Additionally, the PCS software provides for a user to search through the database by using a dynamic pruning algorithm keyed on order insensitive matches. As the user enters a search string at a dialogue box, the list of matches displayed is automatically updated in real-time to correspond to as much of the search string that has already been entered. The list continues to be updated until the search string has been completely entered.

Software Control Description

The preferred embodiment of the software control system of the present invention runs under Microsoft Windows software on an IBM PC or compatible. It will be recognized that other software implementations are available on other types of computers and windowing systems without loss of generality.

FIG. 25 shows the timing loop 131 of FIG. 18 in greater detail. In order to process pending actions, the timer 131 checks the fax out queue at 163, the multi-media out queue at 165, the voice mail out queue at 167, and the communications port at 169. The three output queues and the communications port are checked substantially once every 10 seconds to determine if there are any pending actions to be performed. If the timer does find a pending action in one of the queues, or if there is information coming in from the communications port, a secondary timer of duration 100 milliseconds is spawned in order to handle each pending action. This polling of the output queues continues as long as the main PCS software is active and running. The polling interval rate of substantially 10 seconds is short enough such that there is no significant time delay in handling any pending action in any of the output queues. It will be recognized that short timing intervals other than substantially 10 seconds may be used without loss of generality.

FIG. 38 illustrates the flow control for the outgoing queue timer. The incoming timer is stopped at 3701 and the communications port is initialized at 3703. A job record 3707 is used to determine the destination for this message and dial the telephone at 3705. After a protocol handshake at 3709, an output page is sent at 3711, which may include a fax code file 3713. The remaining number of pages to be sent is checked at 3715, and if there are more pages, control returns to 3709 so that the additional page or pages can be sent to the destination. Otherwise, if at 3715 there are no more pages to be sent, the job record 3707 is updated at 3717, and the incoming time is restarted at 3719.

FIG. 39 illustrates the flow control for the incoming queue timer. At 3801, the communications port is initialized and the software waits for a ring indicator from an incoming call at 3803. After receiving a ring, the software answers, establishes a connection via a protocol handshake at 3805, and receives input data at 3807. If at 3809 there is more data to be received, control passes back to 3805 so that the pending data may be received. Otherwise, if at 3809 there is no more data to be received, the call is terminated at 3811.

FIG. 26 shows the control software used for a hands-off telephone. The telephone control software is invoked upon selection of the telephone option 115, shown in FIG. 2. Upon selection of the telephone option at 2501, any timers that have been spawned or are currently running are disabled, and at 2503, the communications port for the personal computer is initialized. At 2505, the telephone control software handles any options that have been selected by the user as displayed to the user by FIG. 49. At 2507, the user may logically select between a headset, handset, or speaker phone; at 2509, the user may adjust the volume level of the speaker or the gain of the microphone; and at 2511, the user may select between mute, hold, record, and redial functions for the hands-off telephone. After the user selects options at 2513, a telephone number is dialed and the call initiated. After the call is complete, at 2515 the user hangs up, and at 2517 any timers that had been stopped at 2503 are restarted and at 2519 the communications port is restored to its previous state.

FIG. 27 shows the voice mail control software that is invoked by option 117 from FIG. 2. The user may select either a new file or record at 2601, open an existing file at 2613, or abort at 2625. Upon selecting a file or record at 2603, the file may be saved at 2611, or the user may select options at 2605. The selectable options include setting the volume or record levels at 2607, or selecting between a handset, a headset, or a speaker phone at 2609, as shown to the user by the voice mail editor display given in FIG. 51. If an existing file is opened at 2613, the file name is selected at 2615, whereupon a user may then play the file at 2617 or select options at 2619. The selectable options from 2619 include setting the volume or record levels at 2621 or selecting between a headset, handset, or speaker phone at 2623. Once a voice mail message has been recorded or opened from a previous session, a graphical representation of the voice mail message is displayed in a window with x and y dimensions, where the x dimension represents time and the y dimension represents the volume of the voice mail message at that point in time. The pointing device may be used to modify the voice message by graphically changing the two-dimensional voice message plot. The cursor is placed within the two-dimensional voice message plot in order to indicate the portion of the voice message to be modified. A scroll button beneath the two-dimensional voice message plot may be used to select the time portion of the message to be displayed within the two-dimensional plot window. Time is shown increasing form the left to the right, corresponding to the x axis of the plot. As the scroll button is moved to the right, later portions of the voice message are displayed. As the scroll button is moved to the left, earlier portions of the voice mail message are displayed. A numeric value is shown substantially on the left side of the two-dimensional plot windows, which is updated continuously and corresponds to the time value of the current location on the x axis.

Upon the recording of a voice mail message, the voice mail may be added to the voice mail send queue, as displayed to the user in FIG. 50 and described at 159 in FIG. 23. Upon adding a voice mail message to the voice mail queue, the user is prompted, as shown in FIG. 56, to enter a name and a telephone number to whom the voice mail message must be sent. The user may select from a pre-determined list of voice mail recipients previously set up in the address book, as shown to the user in FIG. 55.

FIGS. 28 and 29 show how the fax code drivers of the PCS software typically work. The fax capability is tied to the windowing system print command so that facsimile transmissions may be sent by any software that can print through the windowing environment. At 2701 in FIG. 28, a high resolution fax driver examines a print file at 2703 that has been printed by a windowing system application. The print file is then converted and imaged at 2705 into a PCX bit-mapped format which is of the correct horizontal and vertical resolution in dots per inch (dpi) for high resolution facsimile devices. FIG. 29 shows an equivalent process used by a low resolution fax driver at 2801. A print file 2803 is converted at 2805 to a low resolution PCX bit-mapped format for use with low resolution facsimile devices. Upon converting a print file to a facsimile document, the facsimile document may be added to the fax out queue, as displayed to the user in FIG. 52. The user may select the name and telephone number of a person to send the fax to through the address book function, as shown to the user in FIG. 55. Received faxes may be viewed or printed from the fax in queue.

FIG. 30 shows the multi-media control software that is invoked by a user selecting multi-media option 121 shown to a user in FIG. 2. The user may initialize multi-media mail settings at 2901, select various multi-media mail options at 2905, or invoke a message composer at 2903, as shown to the user by the screen display of FIG. 57. In the message composer 2903, the user may open a new file at 2907 or cancel the message composer at 2911 or open an existing file at 2909. Upon opening an existing file, the file name is selected at 2913, and the multi-media mail editor 2917 is invoked at 2915. While editing a file at 2919 a user may select to alternately play a message at 2921 or record a message at 2923. The user may edit in line mode either inserting, deleting, joining, or splitting lines; edit in block mode by moving, copying, deleting, or highlighting blocks of text; changing the font used to display the text; or changing the indent and justification attributes of paragraphs of text. Standard search features such as forward and backward search and global search and replace are also available through an "other" menu. FIG. 31 further describes the options available that are shown to the user by the multi-media edit display of FIG. 57 (at 3001, file, line edit, block edit, fonts, paragraph, voice, "other," and help options are available). If at 3001 the user selects "file," the options shown in 3003 (save, save as, page lay-out, printer set-up, and print) are available to the user. If at 3001 the user selects "voice," the options available at 3005 (record voice, stop recording, play voice, stop play, store voice to disk, and get voice from disk) are available to the user. After selecting from edit controls 3001, at 3007 the appropriate action is taken by the PCS software, the voice icon is displayed, and any additional graphics are also displayed.

After a multi-media message has been created, at 3101 the user may add a message to the send queue. Upon adding the message, at 3103 the user selects the name and telephone number of a person to send a message to, or at 3105 selects a name from the address book and at 3107 selects a destination from the address book to send the message to. The message is then added to the job list at 3109, and, at 3111 the job scroll list is updated.

FIG. 33 shows the options available when a multi-media message has been received. At 3201, the user may select and view the message. At 3203, the software selects the appropriate message, opens the job list at 3205, loads the received message at 3207, and invokes the message composer at 3209 in order to display the received multi-media message.

FIG. 34 shows the software control procedure used with the show-and-tell feature to provide data over voice capability when selected at 123 from FIG. 2. At 3301, any existing timers are disabled and the communications port is initialized at 3303. The destination for any messages is accessed from the address book at 3305, whereupon the telephone number is dialed and the connection is set up at 3307. Upon detecting a successful connection, the data over voice mode is initialized at 3309 while a message is being transmitted at 3311 the user may select options of either quitting PCS at 3315, or invoking a terminal emulation at 3313. The data over voice connection is accomplished by multiplexing the bandwidth of the connection, as described elsewhere in this specification.

FIG. 35 shows the control procedure used when receiving data over voice messages. At 3401, any existing timers are disabled and the communications port is initialized at 3403. At 3405, the software waits for a ring indicator. If, after a pre-determined amount of time, no ringing indicator is detected, a time-out occurs at 3407, whereupon the PCS software aborts are returns at 3409. Otherwise, at 3411 a ring indicator is received, the data over voice connection is established at 3413, and the user may select options at 3415 either to quit the PCS software at 3419 and close the communications port at 3421, or invoke terminal emulation at 3417.

FIG. 36 shows the control procedure used when transmitting voice mail messages. At 3501, a voice mail message is added to the send queue, whereupon at 3503, the user specifies the message file name and, at 3505, indicates a name from a previously entered address book entry 3505 and destination telephone number 3507 to send the message to. The message is then added to the job list at 3509, and the queue display is updated at 3511.

FIG. 37 shows the control procedure used when receiving voice mail messages. At 3601, a voice mail message is received and recorded by the PCS software. At 3603, the user selects a message to be reviewed or edited, after which the software opens the job list at 3605, loads the selected recorded message into memory at 3607, and invokes the voice editor at 3609 with the selected recorded message. The user may then select various functions from within the voice editor to review or edit the message, as previously described >above.

Data Structures Description

Descriptions of the data structures and variable names and types are given below for the preferred embodiment of the present invention. The preferred embodiment is written in the C programming language and runs under Microsoft Windows software on an IBM PC or compatible system; however, it will be recognized that these data structures and methods are generic and potentially useful for a wide variety of other windowing software, systems, and programming languages.

Address Book

Address key types are used to indicate how information within the address book should be displayed to the user:

TABLE-US-00016 KEY_NAME 0 list addresses by name KEY_AFFILIATION 1 by affiliation KEY_STREETADRS 2 by streetadrs code KEY_CITYSTATE 3 by citystate code KEY_ZIP 4 by zip code KEY_PHONE 5 by phone code KEY_FAX 6 by fax code KEY_MISC 7 by misc code KEY_TYPE_COUNT 8 number of key types

The address entry structure defines what fields are associated with each address book entry:

TABLE-US-00017 s_address_entry_struct unsigned int delFlag; delete flag unsigned int caName; Name field unsigned int caAffiliation; Affiliation field unsigned int caStreetAdrs; Street Address field unsigned int caCityState; City State field unsigned int caZip; Zip field unsigned int caPhone; Phone Number field unsigned int caFax; Fax Number field unsigned int caMisc; Miscellaneous field Key Names[KEY_TYPE_COUNT] = "Name" , "Affiliation" , "StreetAdrs" , "CityState" , "Zip" , "Phone" , "Fax" , "Misc"

The following character strings are used to hold address book information:

TABLE-US-00018 char g_adrsbooktemp[MAX_FILE_NAME_LEN] = "temp.adr" ; static char g_ClipboardFormat[ ] = "CF_FLOCOM" ; static char g_adrsbookDlgName[ ] = "AdrsBkDlg" ; static char g_adrsbookFileName[MAX_FILE_NAME_LEN] ; static char g_FaxStr[ADDRESS_ENTRY_FIELD_SIZE] ; static char g_PhoneStr[ADDRESS_ENTRY_FIELD_SIZE] ; static char g_NameStr[ADDRESS_ENTRY_FIELD_SIZE] ;

Fax Send and Receive Queues

A structure definition is used to indicate how information within the fax send and receive queues is stored:

TABLE-US-00019 struct s_jobDesc { union { unsigned char phoneNum[32]; short nbrOf Jobs; } u1; union { short nextJobNum; unsigned char date[16]; } u2; unsigned char title[64]; unsigned char receiver[32]; unsigned char coverFile[32]; unsigned char telNum[32]; unsigned char faxCdFile[8]; unsigned char time[8]; WORD zState; time_t zTime; short zPages; short zResult; int zTries; char zType; char zDummy[19]; } t_jobDesc, *tp_jobDesc, far *tpl_jobDesc; static t_jobDesc g_curJob, g_bufJob;

Multi-Media Send and Receive Queues

The following structure definition is used to indicate how information within the multi-media send and receive queues is stored:

TABLE-US-00020 struct_sjobDesc { union { unsigned char phoneNum[32]; short nbrOfJobs; } u1; union { short nextJobNum; unsigned char date[16]; } u2; unsigned char title[64]; unsigned char receiver[32]; unsigned char coverFile[32]; unsigned char telNum[32]; unsigned char MMCdFile[8]; unsigned char time[8]; WORD zState; time_t zTime; short zPages; short zResult; int zTries; char zType; char zDummy[19]; } tjobDesc, *tp_jobDesc, far *tpl_jobDesc;

Show-and-Tell

The show-and-tell structure definition is the same as that used in the address book. The static variables to define field sizes are given below:

TABLE-US-00021 /* Static Variables Used For Address Book Proc. The Variable Names Are Same As The Ones In "Adrsbook", So That Same Modules Could Be Used */ #define ADDRESS_ENTRY_FIELD_SIZE 64 #define NUM_MEMBERS_ADRS_STRUCT 8 #define NUM_ADRS_FIELDS 8 #define VAL_PAUSE 2000 #define PCKT_COM_TIME 20 #define WAIT_RING_TIME 120 #define DIAL_TIME 60 //1029 vasanth for delay before %p1 command #define DELAYTIME 10 /* Address Key Types */ #define KEY_NAME 0 /* list addresses by name */ #define KEY_AFFILIATION 1 /* by affiliation */ #define KEY_STREETADRS 2 /* by streetadrs code */ #define KEY_CITYSTATE 3 /* by citystate code */ #define KEY_ZIP 4 /* by zip code */ #define KEY_PHONE 5 /* by phone code */ #define KEY_FAX 6 /* by fax code */ #define KEY_MISC 7 /* by misc code */ #define KEY_TYPE_COUNT 8 /* number of key types */

Voice Mail Send and Receive Queues

The voice mail structure definition and static variables to define field sizes are given below:

TABLE-US-00022 #define READCOUNT 24000 #define FRAME_SIZE 2000 #define SCROLL_STEP 4 #define POS_IN_PAGE 100 #define T_POSITION 0.02 #define COMP_FRAME_SZ 24 #define BYTES_IN_FRAME 12000 #define BYTE_NUMBER 6 #define WAVE_COEF 1 #define FORMAT_STR1 "%s -> %s # %s Schedule: %s: %s" #define FORMAT_STR2 "%s -> %s # %s Sent: %s: %s" #define OUT_JOBS_FILE_NAME "jobs" #define IN_JOBS_FILE_NAME "jobs" #define MAX_JOBS 10 struct s_jobDesc { union { unsigned char phoneNum[32]; short nbrOfJobs; } u1; union { short nextJobNum; unsigned char date[16]; } u2; unsigned char title[64]; unsigned char receiver[32]; unsigned char coverFile[32]; unsigned char telNum[32]; unsigned char faxCdFile[8]; unsigned char time[8]; WORD zState; time_t zTime; short zPages; short zResult; int zTries; char zType; char zDummy[19]; } t_jobDesc, *tp_jobDesc, far *tpl_jobDesc; static HWND hwndVMD1g; static int g_cxWave; /* width of waveform window */ static int g_cyWave; /* height of waveform window */ static int g_nSamples; /* sample counter */ static char g_outVMDir[MAX_FILE_NAME_LEN]; static char g_inVMDir[MAX_FILE_NAME_LEN]; static HANDLE g_outJobsHndl = 0; static HANDLE g_inJobsHndl = 0; static HANDLE g_jobFileHndl = 0; static t_jobDesc g_outJobs0; static t_jobDesc g_outJobsn; static t_jobDesc g_inJobs0; static t_jobDesc g_inJobsn; static short g_numOfOutJobs = 0; static short g_numbOfInJobs = 0; static int g_VMOutIx = -1; static int g_VMInIx = -1; static OFSTRUCT jOfStruct; static FILE *vdata; static OFSTRUCT OfStruct; static int hFile; static OFSTRUCT oOfStruct; static int hoFile; static char g_inFileName[MAX_FILE_NAME_LEN]; static int hjFile; static char g_jFileName[MAX_FILE_NAME_LEN]; static char g_uFileName[MAX_FILE_NAME_LEN]; static char g_oFileName[MAX_FILE_NAME_LEN]; static char g_sendListenFile[MAX_FILE_NAME_LEN] static char g_recListenFile[MAX_FILE_NAME_LEN];

The following are variables to be used to write the playback or record level into the pcs.ini volume field names:

TABLE-US-00023 char *g_PlayVolume = "Play Volume"; char *g_RecVolume = "Record Volume"; Default volume levels for record and play: int g_recPos = 5; int g_playPos = 5; char g_MicroVol[10] = ">MV0"; Microphone level to be used for recording. char g_PlayVol[10] = ">SV0"; Speaker level to be used for play. int g_offhk = FALSE; Off hook flag indicating that either record or play is in progress. If flag is set only, then send packet commands to increase/decrease volume level.

Below are given Static Variables Used For Address Book Proc. The Variable Names Are The Same As The Ones In "Adrsbook.c", So That Same Modules Could Be Used:

TABLE-US-00024 #define MAX_REC_VOL 13 #define MIN_VOL_LEVEL 0 #define MAX_PLAY_VOL 9 #define MAX_ADRS_ENTRIES 512 #define ADDRESS_ENTRY_FIELD_SIZE 64 #define NUM_MEMBERS_ADRS_STRUCT 8 #define NUM_ADRS_FIELDS 8

The present inventions are to be limited only in accordance with the scope of the appended claims, since others skilled in the art may devise other embodiments still within the limits of the claims.

Microfiche Appendix

The microfiche appendix of U.S. patent application Ser. No. 08/002,467 filed Jan. 8, 1993 (issued as U.S. Pat. No. 5,452,289), incorporated herein by reference, contains the source code for the software running on the personal computer and the source code for the software running on the voice control DSP/CODEC.

* * * * *
[Image]
[View Shopping Cart] [Add to Shopping Cart]
[Top]
[Home] [Boolean Search] [Manual Search] [Number Search] [Help]