Abstract:
A system for transmitting a wide variety of communications between a wide variety of end user devices via IP networks, including delivering voice quality calls over IP networks between conventional telephones. The system allows end users to change or augment the format for communication, for example between email, chat, voice and video, by simply selecting an icon or button. The communications may also be redirected between the various end user devices and to a message storage center that can store the message in a variety of formats, providing various messaging options. One or more packet managers that receive, transmit and process digital signals control the various communications, providing for users a seamless communication experience traversing the PSTN and IP networks.

Description:
MICROFICHE APPENDIX  
         [0001]    A Microfiche Appendix is included herewith, the Appendix containing computer program code.  
         COPYRIGHT NOTICE  
         [0002]    A portion of the disclosure of this patent document contains material that is subject to copyright protection. The copyright owner has no objection to the reproduction of the patent document or the patent disclosure in exactly the form it appears in the Patent and Trademark Office patent file or records, but otherwise reserves all copyright rights whatsoever.  
         BACKGROUND  
         [0003]    The present invention relates to communication systems, such as telecommunication networks, data networks, and the Internet, and to message transfer over such systems.  
           [0004]    Telephone systems have traditionally set up a fixed connection path at the outset of a call on between individual telephones over a telecommunications network such as the public switched telephone network (PSTN). The fixed path may include plural transmission lines that are connected by plural switches, the switches locked in place for the duration of the call to hold the path for the use of those individual telephones for that duration. In addition to ensuring that communication between users of the phones can occur while the phones are connected, such a fixed path also makes delays in hearing the voice from the other end typically imperceptible, even for the situation in which the path is several thousand kilometers or more in length. Telephone systems such as the PSTN also traditionally have employed analog signals that are created by a microphone in the phone of a user that is speaking and interpreted by a speaker in the phone a user that is listening.  
           [0005]    Although the analog PSTN lets people located in different states or nations sound like they are next door, a disadvantage with this system is an inefficient use of bandwidth for the fixed path. Analogous to this fixed path is a highway that only allows one car to travel at a time, until the car has reached its destination. During pauses in speech the allocated bandwidth is entirely wasted, and during times when only one side is speaking half the bandwidth may be wasted. To reduce this waste of bandwidth, multiple analog conversations can be transmitted over such lines and switches with frequency division multiplexing (FDM). On the other hand, point-to-point dedicated, digital circuits that ensure high bandwidth, such as T-1 and T-3 lines, may employ time division multiplexing (TDM) to interleave a number of voice or data channels, with unused bandwidth wasted.  
           [0006]    Data networks have traditionally offered variable paths for communications between individual computers connected to the networks. Data, such as a digital file that is to be transferred between the computers, is divided into packets that each are affixed with control information such as the addresses of the sending and receiving computers. When the packets are transmitted to a local network by the sending computer, they may be routed through various networks according to their addresses to eventually arrive at the receiving computer. In order to use networks and routers more efficiently, different packets from the same file transfer may be routed over very different paths, especially when the packets are transferred over the Internet. Thus the packets from a file being transferred may arrive at different times or in a different order than that with which they were sent, or in bursts, while some packets may not arrive at all and need to be resent.  
           [0007]    Individual routers that are designed for data networks can be programmed to operate in a fast switching mode as opposed to a process switching mode, so that after a router has sent a first packet for a message to a next hop router, subsequent packets for the same message are sent to the same next hop router. While this may be beneficial for packets that need to be sent as quickly as possible, the efficiency of those routers is diminished for data messages, since the opportunity to choose a low cost route is lost. A message sent over the Internet, which may pass through many routers before delivery, may encounter routers operating in the fast switching mode as well as routers operating in the more common, process switching mode.  
           [0008]    Internet protocol (IP), the most common internet layer protocol used on data networks such as the Internet, is connectionless. Without dedicated connections, IP only uses bandwidth when sending packets containing data. Transport control protocol (TCP), which runs over IP, provides a reliable connection only via end-user activities, such as checking that all data has been received, requesting retransmission of missing packets and rearranging packets into the correct order. Thus voice over TCP suffers from delays in receiving messages, as the data may need to be retransmitted and/or rearranged before being available to an end user.  
           [0009]    User datagram protocol (UDP), which can also run over IP, is a connectionless transport layer protocol. Thus there may not be long delays while processing a message, but the message may not all be received. Further, the packets may be received out of order due to varying paths over the Internet, so that message quality may be low. These problems may be exacerbated with video, which has more data packets and requires more bandwidth.  
           [0010]    Asynchronous Transfer Mode (ATM) packet switching networks use switches that establish a logical circuit from end to end, which guarantees a quality of service (QOS) for that transmission. However, unlike telephone switches that dedicate circuits end to end, unused bandwidth in ATM&#39;s logical circuits can be appropriated whenever available. For example, idle bandwidth in a videoconference circuit can be used to transfer data. ATM works by transmitting all traffic as fixed-length, 53-byte cells. This fixed unit allows very fast switches to be built, because it is much faster to process a known packet size than to determine the start and end of variable length packets. The small ATM packet also ensures that voice and video can be inserted into the stream often enough for real-time transmission. Most networks do not have ATM capabilities, however, and conversion between IP packets and ATM packets can more than negate ATMs advantages.  
           [0011]    Frame relay protocol, runs over ATM networks, provides permanent and switched logical connections. A Permanent Virtual Circuit (PVC) is a logical connection that is provisioned ahead of time, while a Switched Virtual Circuit (SVC) is a logical connection provisioned on demand. The connections are identified by a Data Link Connection Identifier (DLCI) number that is significant to the local frame relay switch, which will change the number as it passes the packet on to its destination, because the receiving switch uses a different DLCI for its end of the same connection. Every DLCI requires a Committed Information Rate (CIR), which is a pledge on the part of the network to provide a certain amount of transmission capacity for the connection. Voice over frame relay enables voice to be packetized and travel over a frame relay network, but suffers from the need for IP translation, may be too slow when implemented as a SVC and too limited when implemented as a PVC.  
           [0012]    Realtime transport protocol (RTP) is a protocol that runs over IP and supports “realtime” transmission of voice and video. An RTP packet rides on top of UDP and includes timestamping and synchronization information in its header for proper reassembly at the receiving end. Realtime transport control protocol (RTCP) is a companion protocol that is used to maintain QOS. RTP nodes analyze network conditions and periodically send each other RTCP packets that report on network congestion.  
           [0013]    Variations in packet reception may not be harmful for file transfers that are not urgent, such as TCP packets that can be rearranged into the correct order by the receiving computer, with missing or damaged packets retransmitted within a few minutes. Such delays and variations, however, are not tolerable for realtime voice or video communications. Even for the situation in which no packets are lost or out of order, bursts of received packets or variations in the Internet paths taken by digital voice packets may cause the packets to be received with a different timing than that with which they were sent, resulting in “jitter” that is heard by the listener. In general, human perception sets limits on the quality of such communications, such that errors that are imperceptible are generally tolerable, while those that are perceptible are not. For instance, people generally cannot detect delays in speech that are less than about 150 milliseconds, so that a telecommunications system that has a one-way delay of less than that amount, and lacks other perceptible errors, may be said to be “voice quality.” 
           [0014]    [0014]FIG. 1 provides a diagram of prior art communication systems including a pair of telephones  20  and  22  that may be connected via local exchange carrier (LEC)  25  and LEC  28  via the PSTN  30 . LECs  25  and  28  may for example be local telephone companies such as regional Bell Operating Companies, and may each include a switch that maintains a connection with the PSTN  30 , which also maintains a fixed path for the duration of a call between telephones  20  and  22 . Personal computer (PC)  33  and PC  35  can also be connected to LECs  25  and  28 , sending and receiving analog signals that are directed to gateway routers  37  and  39 , for conversion to digital packets that are routed over the Internet  40 . In this case communications between PCs  33  and  35  are sent as IP packets that may traverse the Internet over variable paths.  
           [0015]    It is also possible to send messages from PC  33  to telephone  22  over the Internet  40 , or from PC  33  to telephone  22  over the Internet  40 . PCs  33  and  35  may in this case each be equipped with a microphone that turns sound into analog electrical signals, the analog signals then being converted to digital signals by the PC. For the situation in which the PC is connected to an analog LEC line, the digital signals may again be converted back into analog signals, which are directed to a gateway router that converts the signals back into digital packets for transfer over the Internet  40 . Due to the variable paths that packets may take over the Internet, the voice quality may be very poor. For example, an average packet sent via the Internet from one coast of the U.S. to the other in 1999 may have encountered more than twenty routers, each of which adds delay and may send the packet to a different node than related packets.  
           [0016]    On the other hand, telephones  42  and  44  are equipped with specialized jitter buffers  46  and  48 , respectively, that include mechanisms for rearranging the timing of received packets to correspond to the timing with which those packets were transmitted. Since rearranging the timing of the packets adds to the overall delay, a tradeoff between perceptible delays and jitter exists that has caused voice communications over the Internet to be of generally poorer quality than traditional telephone calls.  
           [0017]    [0017]FIG. 2 shows another prior art communication system as described in U.S. Pat. No. 6,069,890, which is incorporated by reference herein. A pair of telephones  50  and  52  may be connected via LEC  55  and LEC  58 , which may each include a switch that affords a connection with the PSTN, not shown in this figure. PC  53  and PC  55  can also be connected to LECs  25  and  28 , sending and receiving analog signals that are directed to gateway routers  57  and  59 , for conversion to digital packets that are routed over the Internet  40 . In this case communications between PCs  53  and  55  are sent as IP packets that may traverse the Internet  40  over different paths.  
           [0018]    Gateway router  57  includes an Internet address database  77  that can be used by telephone  50  to route a telephone call over the Internet  40 . To do this, a user of telephone  50  dials a special number that is directed to gateway router  57 , alerting that gateway router that a telephone call over the Internet is to follow. The user then dials the telephone number of telephone  52 , to initiate a telephone call between phones  50  and  52 , which call is also directed to gateway router  57 . Internet address database  77  has a table associating various telephone numbers with Internet addresses of adjacent gateway routers. Using this table, gateway router  57  directs the number dialed to another gateway router  59  that is adjacent to an LEC  58  providing service to telephone  52 , and the call is connected, assuming telephone  52  is not off-hook. Although this system is said to allow a conventional telephone to call another convention telephone via the Internet, the quality may be poor, given the delay and jitter inherent in such an Internet routed call.  
           [0019]    Entering another special number allows telephone  50  to call PC  55  or telephone  52  to call PC  53  via the Internet  40 , via a similar mechanism. Switching systems  80  and  82  are connected to LECs  55  and  58 , respectively, providing switching mechanisms for telephones  72  and  75 , respectively. A voice mail system  85  is connected to switching system  80  and another voice mail system  88  is connected to switching system  82 . Voice mail systems  85  and  88  can store messages of various formats, such as voice, fax and text. Switching systems  80  and  82  and voice mail systems  85  and  88  all use analog signaling. While this may be adequate for adjacent telephones, the availability of voice mail system  85  to distal telephones  75 , for example, may be limited.  
         SUMMARY  
         [0020]    In accordance with the present invention, voice quality calls may be transmitted over IP networks between conventional telephones. In addition, a wide variety of communications may be routed between a wide variety of end user devices via the IP networks. Users may change or augment the format for communication, for example between email, chat, voice and video, by simply selecting an icon or button. The communications may also be redirected between the various end user devices and to a message storage center that can store the message in a variety of formats, providing various messaging options. One or more packet managers that receive, transmit and process digital signals control the various communications, providing for users a seamless communication experience traversing the PSTN and IP networks. 
       
    
    
     DESCRIPTION OF THE FIGURES  
       [0021]    [0021]FIG. 1 provides a diagram of some prior art communication systems including a traditional telephone system and the Internet, along with a way of routing a call from a PC to a telephone over the Internet.  
         [0022]    [0022]FIG. 2 provides a diagram of another prior art communication system including a way of routing a call from a first telephone to a second telephone over the Internet.  
         [0023]    [0023]FIG. 3 shows a communication system  100  in accordance with the present invention, including a packet manager that controls various communications of the system.  
         [0024]    [0024]FIG. 4 shows some components of a first packet manager of FIG. 3.  
         [0025]    [0025]FIG. 5 shows some components of a second packet manager of FIG. 3.  
         [0026]    [0026]FIG. 6 shows a database component of a packet manager of FIG. 4 or FIG. 5.  
         [0027]    [0027]FIG. 7 shows a communication device displaying various communication options provided by the packet manager of FIG. 4.  
         [0028]    [0028]FIG. 8 shows a protocol-processing stack contained in a gatekeeper of the packet manager of FIG. 4. 
     
    
     DESCRIPTION OF THE PREFERRED EMBODIMENTS  
       [0029]    [0029]FIG. 3 shows a communication system  100  in accordance with the present invention, including a packet manager  101  that controls various communications of the system. An optional second packet manager  102  is shown that may provide additional controls for communications of this system  100  or others, not shown. First and second conventional telephones  105  and  107  are connected to conventional telephone lines  110  and  112 , which are respectively connected to switches  115  and  118 . Similarly, PC  120  and PC  122  are connected by conventional transmission lines  125  and  128  to switches  115  and  118 , respectively. Transmission lines  110 ,  112 ,  125  and  128  may be conventional twisted pairs of copper wires or other transmission lines, such as coaxial cable or fiber optic lines, or may represent a wireless connection for example with a cellular telephone. Also connected to switch  115  are facsimile machine  106  and videophone  108 , which may be connected by lines  110  and  128  or separately as shown. Similarly, facsimile machine  114  and videophone  116  are connected to switch  118 . Advantageously, the present invention will work with a wide variety of devices, including common equipment such as conventional telephones and lines, without the need for upgrading or replacing existing equipment.  
         [0030]    Switches  115  and  118  are connected to the PSTN  30  for traditional analog telephone calls, and are also connected to gateways  130  and  133  by transmission lines  135  and  138 , respectively. Switches  115  and  118  also have a direct connection to packet managers  101  and  102  via transmission lines  136  and  137 , respectively. Gateways  130  and  133  are coupled to packet managers  101  and  102  by transmission lines  140  and  142 , respectively, and may also be coupled to the Internet  40  separately from packet managers  101  and  102 , those connections not shown in this figure for clarity. Transmission lines  140  and  142  are private networks allocating sufficient bandwidth to IP packets transferred between packet managers  101  and  102  and respective gateways  130  and  133  to ensure that the latency therebetween does not exceed  30  milliseconds. Gateways  130  and  133  translate between analog signals employed by switches  115  and  118  and digital signals employed by packet managers  101  and  102 . Gateways  130  and  133  may optionally be contained in respective packet managers  101  and  102 . A signaling system 7 (SS7) database  131  is provided by the PSTN  30  for control functions such as connecting, routing and disconnecting telephone calls, the SS7 database accessible by packet managers  101  and  102 . Transmission lines that carry primarily analog signals, such as lines  135  and  138 , are shown as solid lines in this figure, whereas transmission lines that carry primarily digital signals, such as lines  140  and  142 , are shown as dashed lines.  
         [0031]    Packet managers  101  and  102  provide controls for various communications devices illustrated in FIG. 3, via signals appropriate for each such device. Although packet managers  101  and  102  operate on, send and receive digital signals, the packet managers can control analog devices such as telephones  105  and  107  via analog signals over lines such as  110  and  112 , the analog signals converted from digital signals sent by the packet managers  101  and  102  to gateways  130  and  133 . Stated differently, the packet managers  101  and  102  control and provide communication between various and sundry devices shown in this figure with IP protocol signals generated, received, translated and transferred by the packet managers.  
         [0032]    Packet managers  101  and  102  are coupled to the Internet  40  via routers  144  and  146 , with web site  148  coupled to or part of the Internet. As described below, conventional analog telephone  105  as well as videophone  108  or PC  120  are enabled by packet manager  101  to access web site  148 . A high-speed trunk  150 , which may for example include optical fibers, employ dense wave division multiplexing (DWDM) and traverse states or connect continents, directly connects packet managers  101  and  102 . As described below, the combination of high-speed trunk  150  with dedicated transmission lines  140  and  144  allows packet manager  101  provide voice quality communications over a wide area IP network between conventional analog telephones  105  and  107 .  
         [0033]    Packet manager  101  is also coupled to a message center  155  that stores digital messages of various types for the various devices. Message center  155  can convert the messages from digital to analog signals and provide the analog signals via switch  115  to phone  105 , fax  106 , video phone  108  or PC  120 , in the appropriate form for the selected device and under control of the packet manager  101 . Packet manager affords various options for routing, receiving and storing communications, which may be selected after initiation of a call, such as call forwarding, call waiting, call transfer, finding a device where a person can be reached, connection of multiple parties, and connection of multiple adjacent devices.  
         [0034]    Packet manager  101  can also provide identification and preferences to emergency services calls  160  such as 911, police or fire calls. Note that these emergency services  160  are available via conventional analog telephone  120  even if a power outage were to render specialized Internet phones useless. Since many emergencies may include a loss of power, this feature may provide a tremendous advantage to a communication system of the present invention, especially compared to competing voice over IP technologies.  
         [0035]    In addition, packet manager  101  may be configured to provide support for the Communications Assistance for Law Enforcement Act (CALEA)  166 , unlike some competing voice over IP technologies. CALEA essentially mandates that new forms of technology allow law enforcement officials to conduct authorized electronic surveillance. In accordance with the present invention, upon receiving the required authorization from a court to conduct a wiretap of telephone  105 , for instance, packet manager  101  can simply copy packets headed to or from that telephone and send the copied packets to the appropriate law enforcement agency.  
         [0036]    A local area network (LAN)  170  is also coupled to packet manager  101  in this example, with a PC  172  and telephone  175  coupled to LAN  170 . Telephone  175  may be a videophone, and other devices may be coupled to LAN  170 , which are not shown for clarity. In this configuration digital signals (IP packets) may be sent between devices such as PC  172  and phone  175  and LAN  170 , and between LAN and packet manager  101 .  
         [0037]    Another gateway  180  is shown coupled to packet manager  101  and a PC  184 , phone  185  or other devices may be connected to gateway  180  via switch  182 . In this manner, voice quality communications are available between conventional analog telephones  105  and  185  via IP networks connected by packet manager  101 . Gateway  190  is coupled to high-speed trunk  150  and switch  192 , converting analog signals sent from switch  192  to digital signals that are directed to trunk  150  and vice-versa. PC  194  and telephone  195  are connected to switch  192 , with packet manager  101  and trunk  150  affording voice quality communications over an IP network that may span or connect continents.  
         [0038]    [0038]FIG. 4 shows that packet manager  101  contains a gatekeeper  200 , database  202 , and web site  208 . Gatekeeper  200  is coupled to a gateway  205 , which may be similar to gateway  130  or  133 , gateway  205  in turn being coupled to a telephone  210 , optionally via a switch that determines whether the phone  210  is connected to the PSTN or the packet manager  101 . Web site  208  provides communications, such as hypertext markup language (HTML) or extensible markup language (XML) pages, that are interpreted by a web browser of a user interface device such as PC  212  to be accessible by a graphical user interface of such a device. PC  212  has multimedia capabilities such as a speaker, microphone, display and optional camera.  
         [0039]    A computer program that runs packet manager  101  on a Windows® NT operating system is included in the Microfiche Appendix submitted with this application. A similar program may be used to run packet manager  101  on other operating systems, such as Unix or Linux.  
         [0040]    The gatekeeper  200  and gateway  205  each include a processor and may be connected to each other by an Ethernet cable, with the gatekeeper serving as master and gateway as slave. The database  202  may be contained in a disk drive or plurality of disk drive, such as a redundant array of inexpensive disks (RAID) system, which may also be connected to the I/O bus or may be coupled via a network such as Ethernet. User data from database  202  is also cached in a short-term memory of gatekeeper  200  for users that are currently connected.  
         [0041]    Gateway  205  provides conversion between analog telephone signals that may be used by telephone  210  and the PSTN  30 , and digital signals used by the packet manager, including control signals and packets containing compressed voice data. Gateway  205  includes codec hardware or software that converts analog sound, speech or video to digital code (analog to digital) and vice versa (digital to analog). The gateway  205  in this embodiment provides a node for both the analog PSTN  30  and a digital IP network including the packet manager  101 , the gateway  205  being connected to and controlled by gatekeeper  202 . In another embodiment, packet manager  101  may contain gateway  205 , instead being coupled to a gateway that communicates with other IP nodes in addition to packet manager  101 .  
         [0042]    [0042]FIG. 5 shows that packet manager  102  contains a gatekeeper  220 , database  222 , and gateway  225 . Gateway  225  is coupled to a telephone  230 , optionally via a switch that determines whether the phone  230  is connected to the PSTN or the packet manager  102 . Unlike packet manager  101 , packet manager  102  does not contain a web site, with a web server  218  instead being coupled to packet manager  101  by transmission lines. Web server  218  provides communications accessible by a graphical user interface of a device such as cellular telephone or PDA  232 . Cell/PDA  232  includes a web browser and has multimedia capabilities such as a speaker, microphone, display and optional camera.  
         [0043]    [0043]FIG. 6 shows an embodiment of database  202  with examples of data contained therein. A user name  300  or alias is shown in a first column, which in this example contains the names Sue, Bob and Terry. The database can contain information corresponding to many more users than the few that are shown for brevity. At least one IP address  303  is associated with each user name, as shown in a second column. The user name may be associated with a password or other identifier, such as a biometric identification, that restricts access to the database  202 . The IP addresses  303  may be static or temporary dynamic host configuration protocol (DHCP) addresses. In the illustrated example, user name Sue has an IP address of 623.996.058.103, user name Bob corresponds to IP addresses 051.693.789.456 and 193.004.628.551, and user name Terry has a DHCP IP address assigned as needed. At least one telephone number  305  is associated with each user name, as shown in a second column. In accordance with the current invention, a user can maintain a telephone number that is available for user communication irrespective of the physical location of the user.  
         [0044]    Voicemail messages  310  or other audio files for each user can be stored in digital form in the database  202 , for example using the WAV format. The examples include a call from Terry in Bob&#39;s voicemail box, and no voice messages in Sue or Terry&#39;s voicemail box. Fax messages  313  or other image files for each user can also be stored in digital form in the database  202 , for example using TIF or JPEG formats. Similarly, although not shown, database  202  can contain digital video files for a user that may be stored, for example, in MPEG format.  
         [0045]    A call forwarding  315  capability is also included in the database  202  for each user, the call forwarding programmable according to the user&#39;s preferences. The call forwarding  315  can be set up to perform “find me” and/or “follow me” functions. In the “find me” mode, the call forwarding  315  can be programmed to sequentially connect with various devices corresponding to the user name  300  in an attempt to reach a device immediately accessible to the user. If access to the user is not successful through any device, a message can be left for the user in the voicemail box. In the “follow me” mode, the call forwarding  315  can be programmed to attempt to direct a communication initially to a communication device that was most recently used by a user, or to a device that has been preselected by the user. Thus for user name Sue, call forwarding is set up to initially try her work telephone, then her personal digital assistant (PDA), then her cellular telephone, and then her home telephone, after which a message may be stored in her voicemail box. For user name Bob, call forwarding  315  is programmed to poll Bob&#39;s cellular telephone, home telephone, work PC and PDA, while user name Terry&#39;s call forwarding  315  is programmed to poll Terry&#39;s PDA, work telephone, home telephone and work PC.  
         [0046]    An integrated voice response (IVR) function is also included in database  202 , which offers at least one prerecorded message providing callers various options on directing their call. In the example shown in FIG. 6, Sue has her IVR  320  to offer callers the option of having the communication system poll Sue&#39;s various communication devices in an attempt to find Sue, or to simply leave a message for her. Bob&#39;s IVR  320  offers callers a selection of options, including voice mail, email, and paging services, as does Terry&#39;s IVR  320 .  
         [0047]    A 911 emergency call recognition and forwarding feature  322  is also included in database  202 . That is, upon dialing 911 or another known emergency telephone number, the caller is connected to an emergency services telephone number near the telephone from which they called. The SS7 database is used to find the emergency services telephone number that is near the caller. At the same time, the call may be recorded, and an indication of the location of the caller may be provided to the emergency services provider. This redundant, battery-backed, always-available emergency call recognition and forwarding feature  322  is in contrast to some internet telephones, which depend upon local power sources that may be unavailable during an emergency.  
         [0048]    In addition, database  202  has a capability for storing Communications Assistance for Law Enforcement Act (CALEA)  323  information corresponding to each user, unlike some competing voice over IP technologies. CALEA essentially mandates that new forms of technology allow law enforcement officials to conduct authorized electronic surveillance. Upon receiving the required authorization from a court to conduct a wiretap of a user&#39;s telephone, PC, PDA and/or other communication device, the CALEA part of the database can store that authorization, in order to provide a copy of a communication to the authorized law enforcement agency, for example by connecting a wiretap at the same time that the communication is connected. In the example illustrated in FIG. 6, the CALEA  323  authorization corresponding to both Sue and Bob are inactive, whereas Terry&#39;s has an active authorization, indicating the ability to perform some form of wiretapping.  
         [0049]    Although not shown, database  202  may include other features such as voice recognition software that can be employed by the user to access various functions including those shown in the database. The voice recognition software can maintain a record of a user&#39;s speech patterns for biometric identification and increased accuracy in recognizing the user&#39;s commands.  
         [0050]    [0050]FIG. 7 shows a communication device such as PDA  232  with a display  400  that reflects the user&#39;s connection to web server  218 , as interpreted by a web browser program of the PDA. The PDA has a speaker  404 , a microphone  408 , a keyboard  410  (which may be touch-screen) and a camera  414 . The display offers links to several communication options for the user: voice  420 , chat  422 , video  424  and email  426 . Using PDA  232 , a user can connect to another user via packet manager  102 , for example, by selecting a link to one of the communication options, which may be displayed as icons, and then entering the number of the person or selecting from a link to a list of addresses  428 .  
         [0051]    Once connected to a person, it is possible to select another communication option simply by clicking or touching the link to that option. Thus, for example, a first user can send an email to a second user, by first selecting the email link  426 , then selecting the address link  428 , and then selecting the second user&#39;s address. The second user happens to be online and, after reading the email and seeing that it was just sent, selects a chat function to respond. The chat response is received by the first user, who decides that he would like to communicate by voice, and selects the voice  420  icon. After speaking for a few minutes, the first and second users decide that they would rather communicate by video, and select the video icon  424 .  
         [0052]    Alternatively, more than one communication format can be used simultaneously, for example voice and text communication, so that the two users can speak to each other while working together on a text document that is displayed on both communication devices. As another example, more than two users can communicate simultaneously using one or more communication formats. The packet manager  102 , in coordination with the web server  218 , is able to seamlessly connect, disconnect and transfer between these various communication formats.  
         [0053]    [0053]FIG. 8 shows various protocol layers of the gatekeeper  202 . A physical layer  500  provides the transmission of bits between the gatekeeper and various devices, such as PCs and other digital communication devices, the SS7 database for telephone call control, internet gateways and routers. A data link layer  502 , which may employ Ethernet protocols, controls reception and transmission of bits to and from higher protocol layers. The physical transmission and reception of bits as well as media access control of layers  500  and  502  may be performed by hardware chips.  
         [0054]    A network layer  505  of gatekeeper  202  runs internet protocol (IP), which may in this embodiment be IP versions 4 or 6 (IP4 or IP6). For a transport layer  508 , transport control protocol (TCP) or uniform datagram protocol (UDP) are employed, with TCP more frequently used for connection setup, teardown and monitoring, and UDP more frequently used for data transfer.  
         [0055]    A session layer  510  of gatekeeper  202  runs realtime transport protocol (RTP) which, as mentioned above, is a protocol that runs over IP and supports “realtime” transmission of voice and video. An RTP packet rides on top of UDP and includes timestamping and synchronization information in its header for proper reassembly at the receiving end. Realtime transport control protocol (RTCP) is a companion protocol that is used to maintain QOS. RTP nodes analyze network conditions and periodically send each other RTCP packets that report on network congestion. Session layer  510  also runs reservation protocol (RSVP), which is a communications protocol that signals a router to reserve bandwidth for realtime transmission. RSVP is designed to clear a path for audio and video traffic and thereby reduce latency and jitter.  
         [0056]    A codec layer  512  is included in the gatekeeper for handling various voice transmissions bit rates and compressions, such as G.711 for uncompressed voice at 64 kilobits per second (Kbps), G.723 at 32 Kbps and G.729 at 8 Kbps. Also supported are fax protocols such as T.37 and T.38 for use of fax within H.323. Although not shown in FIG. 8, the codec layer  512  can also contain video codecs, including H.261 and H.263. The codec layer  512  provides the gatekeeper with the ability to control a codec protocol of gateways such as gateway  130 ,  133 ,  205  and/or  225 , so that a pair of gateways that are communicating have matching codec protocols.  
         [0057]    A governing layer  515  of the gatekeeper  502  contains various high level protocols that may govern communication between various end user devices and over various networks including the PSTN and IP networks. The governing layer  515  includes H.323, an International Telecommunications Union (ITU) standard for realtime, interactive voice and videoconferencing over IP networks such as LAN  170  and the Internet  30 . H.323 defines protocols by which any combination of voice, video and data can be transported over an IP network. H.450, which is also included in governing layer  515 , specifies various messaging options in concert with H.323.  
         [0058]    Governing layer  515  also includes Media Gateway Control Protocol (MGCP), which is a protocol for IP telephony from the Internet Engineering Task Force (IETF). Working in conjunction with the Gateway Location Protocol (GLP), MGCP enables a caller with a PSTN phone number to locate the destination device and establish a session. It provides the gateway-to-gateway interface for session initialization protocol (SIP), another protocol included in governing layer  515 . SIP is a protocol that provides IP telephony services similar to H.323, but is less complex and uses less resources, making it suitable for very small portable devices. Thus governing layer  515  provides for a plurality of otherwise distinct protocols governing voice communications, and more than one of the distinct protocols may be used for a particular conversation.  
         [0059]    The provision of all these various alternative and competing protocols in the governing layer  515  allows gatekeeper  202  to control and integrate a wide variety of communication devices and networks, so that end users can communicate via otherwise incompatible devices, networks and protocols that their communication may encounter.  
         [0060]    Although we have focused on teaching the preferred embodiments of a system for handling diverse communication formats and communicating messages between various communication devices, other embodiments and modifications of this invention will be apparent to persons of ordinary skill in the art in view of these teachings. Therefore, this invention is limited only by the following claims, which include all such embodiments, modifications and equivalents when viewed in conjunction with the above specification and accompanying drawings.