Abstract:
A link server is added to a traditional telecommunication system to allow seamless integration of voice on network (“VON”) with traditional telephony. The link server accepts traditional telephony and voice on network calls. The link server can distribute the calls from a queue. The link server includes a voice card and network communication apparatus for acquiring VON data and either converting it to telephony data to forward calls to a PBX, or forwarding calls directly to a desktop.

Description:
This application is a continuation-in-part of U.S. patent application Ser. No. 08/813,970, filed on Mar. 3, 1997 (still pending), hereby incorporated by reference in its entirety, which is a continuation of U.S. patent application Ser. No. 08/758,063, filed November 27, 1996 now U.S. Pat. No. 5,724,418, herein incorporated by reference in its entirety, which is a continuation of U.S. patent application Ser. No. 08/595,861, filed Feb. 6, 1996 (now abandoned) which is a divisional of U.S. patent application Ser. No. 08/450,268, filed May 25, 1995 (now U.S. Pat. No. 5,557,668 (the “668 patent”) which is a continuation of U.S. patent application Ser. No. 07/904,196, filed Jun. 25, 1992 (now abandoned), hereby incorporated by reference in its entirety, and claims the benefit of U.S. Provisional Patent Application Ser. No. 60/059,285, filed Sep. 17, 1997, hereby incorporated by reference in its entirety. 
    
    
     FIELD OF INVENTION 
     The invention is a method for integrating voice on network with traditional telephony in a corporate network. In particular the invention relates to person-to-person calls and local and virtual call centers. 
     BACKGROUND OF THE INVENTION 
     The landscape of telephony is changing rapidly today. Traditional telephony networks no longer carry all the telephone traffic to a business. Some voice traffic is present today on public networks such as the Internet. This is termed voice on network (“VON”). Blending VON traffic with traditional telephone traffic presents difficult problems to corporations. These problems are found in both person-to-person and call center environments. Over time, a significant mix of voice traffic will shift to packet network sources. This shift will create a significant need to bring packet voice traffic into the existing telephony environments. For some years there will be a large market for adaptive rather than replacement systems. 
     The technology for packetizing voice for sending on networks is well known. Routing of packetized voice and the solution of inter-working of packetized voice in traditional telephony environments is still an area in need of significant innovation. A typical scheme for delivering VON calls to a PBX is to deliver VON calls to a gateway device which converts VON call traffic to T1 or analog. Output from this gateway looks like regular telephony traffic to the PBX. This approach enjoys the benefit of simplicity. Unfortunately much routing information and interactivity is lost in this arrangement. Calls from a packet network carry useful information relating to the caller and the caller&#39;s interests as well as history of interaction with a company&#39;s data systems such as Web servers. This call-related information is useful in forming accurate routing and meaningful dialogue with the caller—whether the dialogue is audio, video, or web interactive. 
     Calls to individuals in a company typically need less of this type of routing and interaction than calls to call centers. Due to the volume of calls handled, call centers must formalize the interaction and routing of calls. Individuals need routing and caller interaction but on a more dynamic basis. For instance, an individual needs to get calls routed to their current location—which may change. A caller also needs to be able to deliver messages and receive delivery of messages meant for their ears only. Whether calls are made to call centers or to individuals, there are significant ways to make these interactions more sophisticated and more valuable when the call is received through the network. However, this benefit is lost in conventional system because gateways strip out voice content and separate it from other call-related information. 
     As the shift to Voice on Network (VON) traffic occurs corporations need ways to bring this traffic into their existing networks. For the next several decades corporations will need a good way to handle both circuit switched voice calls and VON calls. Ultimately, the choice to replace existing infrastructure switching with all VON may occur. The same infrastructure used to facilitate the coexistence of VON with circuit switched voice needs to be capable of replacing circuit switched voice. 
     SUMMARY OF THE INVENTION 
     The present invention solves problems associated with the prior art by facilitating call routing through the PBX and the VON through a Link Server (LS). Calls are delivered to the link server as either traditional telephony (T1, analog, digital handset, or C link) or as SETUP messages in a VON call handling protocol. The LS inputs the calls and handles them appropriately: telephony calls receive voice prompts and responsive DTMF signals are collected, SETUP messages from VON protocols are sent and Web page interactions are established. Ultimately, whether the call is conventional telephony or VON, the processing required to handle it is reduced to a message to a call processing system. 
     In a preferred embodiment of the invention the call processing system is a call distributor such as that described in the &#39;668 patent. 
     In alternate embodiments of the invention the LS places or tracks a traditional telephony call in a PBX or ACD—passing status messages to a program running on a desktop PC. The desktop PC program synchronizes the display of the incoming calls with messages received from the LS. For traditional telephony calls this means display information relating to calls being processed inside the PBX or ACD switching device. For VON calls this means display information relating to calls being processed by the LS. 
     In a queuing system calls from both sources are intermixed. Agents are able to see the source of calls and handle them appropriately. VON call handling can include web interaction with the caller before or instead of a full voice connection. The Agent or LS system can also offer a callback option to the caller over traditional telephony equipment when a better grade of voice quality is desired. Whatever the interaction, the result generally leads to a completed call to an agent. Traditional telephony calls are transferred or completed to the agent through the actions of the coordinated efforts of the agent desktop software and the LS and the PBX. For example, a switchhook transfer can be used to transfer the call held in the LS to the agent. Alternately, a message passed to the PBX/ACD via a switch link can be used to force completion of the call to the agent. VON call sources are passed the network address of the agent&#39;s PC (e.g., the IP address) so that a point-to-point connection can be established between the agent&#39;s PC and the VON call source. 
     For VON calls, the packetized voice must be decoded from the network. Preferably the decoding is performed in a server, for example at the LS or at the agent&#39;s PC. Regardless, the LS can coordinate the passage of the call to the agent. It may also perform an intermediate step of performing a voice connection to the caller to play prompts or audio messages while the caller is in queue before the connection is passed finally to an agent. In this way the connection can be moved from point to point in the call center. 
     When decoding occurs at the LS, the LS must have resource cards which perform the decoding function. One such card is that supplied by the Natural Microsystems Fusion product. Fusion cards decode/encode voice to/from the network on DSP&#39;s dedicated to each voice path. National Microsystems also has a card which contains a TCP/IP protocol stack. The TCP/IP protocol stack on the card is optimized for packet passing from and to this DSP card as well as to and from a data network. This is required to make the solution independent of the microprocessor and operating system of the LS (i.e., the solution is scaleable). 
     When decoding occurs at the desktop, resources are in a voice card similar to the QX2000 board made by Natural Microsystems. This card decodes/encodes packets from/to the network. In one embodiment of the invention the TCP/IP stack is running in the operating system of the PC. This is workable since only one call path and one card are present in the PC (scaleability is not an issue). 
     One of the benefits of this invention is allowing bridge technology to be built between existing switching and data network communications features. Because the device at the desktop is fully capable of VON and standard telephony both types of communications can be processed on a per-call basis. It is necessary in this model to have conferencing capabilities on the card in the PC. Conferencing at the desktop is made considerably simpler than centralized conferencing since both conversations meet at the desktop. Centralized conferencing of VON and traditional telephony would require VON to be converted to traditional telephony and passed into a switch. This is undesirable for reasons discussed above. 
     Local conferencing makes control software much simpler since there is no resource shared by multiple users. The usual data structures, linked lists, audit programs, race conditions, and other timing conditions and software logic are unnecessary. Overall control of the conference is maintained by the single user at the PC. Resources are dedicated to this user so no sharing is necessary. Conference setup and teardown information comes from a single source—the local PC. The result is reduced complexity plus the ability to bridge VON calls with traditional telephony. 
     The VON capabilities of the card at the desktop make it ideal for decoding voicemail messages delivered through data communication methods such as attached e-mail. Putting VON, a traditional telephony interface and bridging capabilities onto a single card makes it possible to create workgroup and wide area features which transcend the feature set of the traditional telephony switch. This makes it possible to create enhanced features and cost reduced capabilities not offered or possible from the traditional switch. An example of this is long distance calling. An originated call from the desktop could be placed through the traditional telephony switch or over the network depending on cost of connection or quality of connection. Companies with high bandwidth intranets can use them for voice either routinely or as a backup. Choice of voice path over the data network could be determined by query and response time across the network or by query to and positive response from a network traffic server. 
     When this Voice on Network and Traditional Telephony (VON/TT) device is part of a client server telephony system such as that described in the &#39;668 patent still more capabilities and functions are available. An example is personalized call coverage. A caller from traditional telephony connection reaching a voice card on a PC can be given instructions programmed by the user which could include alerting, e-mail, transfer, forward, or conversion to VON for transfer or forward to other users in the system. 
     These and other objects of the present invention are described in greater detail in the detailed description of the invention, the appended drawings and the attached claims. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a schematic illustration of a telecommunication according to a preferred embodiment of the present invention. 
     FIG. 2 is a schematic illustration of an LS gateway device according to a preferred embodiment of the present invention. 
     FIG. 3 is a schematic illustration of the software architecture executing in the LS gateway device according to a preferred embodiment of the present invention. 
     FIG. 4 is a schematic illustration of an LS gateway device according to a preferred embodiment of the present invention. 
     FIG. 5 is a schematic illustration of the software architecture executing in the LS gateway device according to a preferred embodiment of the present invention. 
     FIG. 7 is a schematic illustration of the message flow a point-to-point call setup for a VON call. 
     FIG. 8 is a schematic representation of a voice card according to a preferred embodiment of the present invention. 
     FIG. 9 is a schematic representation of call distribution message flow according to a preferred embodiment of the present invention. 
     FIG. 10 is a schematic illustration of message flow for call distribution using a web server according to a preferred embodiment of the present invention. 
     FIG. 11A is a schematic illustration of WCP registration according to a preferred embodiment of the present invention. 
     FIG. 11B is a schematic illustration of call distribution using an SLPP according to a preferred embodiment of the present invention. 
     FIG. 12 is a schematic representation of a routing table according to a preferred embodiment of the present invention. 
     FIG. 13 is a flow chart for message filtering according to a preferred embodiment of the present invention. 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     FIG. 1 is a schematic illustration of a telecommunications system  10  according to a preferred embodiment of the present invention. System  10  is composed of Private Branch Exchanges (PBX)  104   a  and  104   b  connected to Central Office (CO) switches,  100  and  120 , and desktop telephones  108   a ,  108   b  and  113   a  and  113   b . PBXs  104   a  and  104   b  are preferably conventional voice telephony premise switches. Telecommunication system  10  also includes local area networks (“LAN”)  116   a  and  115   b  and Link Servers (“LS”)  105   a  and  105   b  as well as gateways  103   a  and  103   b . Connected to network  115   a  are data communications gateways  116   a  and  114   a . These allow the IAN  115  to pass packets of information to either the public internet  118  or a private intranet  122 . In addition, a tandem  125  can be connected to the Internet  118  through a link server  105   c.    
     Web server  123  is connected to the LAN  115   a  to accept web interactions from the local and wide area network connections. A web server (not shown) can also be attached to LAN  115   b  in like manner. Attached to the data networks shown are various communications gateways and VON gateways ( 103   a ,  103   b , and  119 ). Remote telephony networks are tied together using leased private facilities  124  and standard telephones attached to central offices (“CO”) such as CO  100 . PBXs are connected to the public switching network via T1 or other standard telephony communications methods  110   a  and  110   b.    
     The description of the preferred embodiment is from the perspective of the “a-side” or left side of FIG.  1 . It would be apparent to those skilled in the art that the description applies to any system having an architecture similar to that shown on the left side of FIG. 1, including the “b-side” or right side of FIG.  1 . 
     Voice telephony traffic enters the switching environment of the PBX  104   a  from many sources. Ultimately these are either from the public switched telephone network (“PSTN”) via central office (“CO”)  100 , private facilities connected  124  to other private switches (e.g., PBX  104   a  through CO  120 ), voice calls placed within the company&#39;s own data communications Intranet  122 , or voice calls placed to the company from the public data communications Internet  118 . 
     Traditional telephone voice connections reach individuals or groups within the corporation by the signaling and address information passed through telephony interfaces such as T1 or PRI, both of which are well-known in the art. Alerting and display of calls to users in the corporation is accomplished through desktop instruments such as analog phone  108   a  or multibutton digital display phone  113   a.    
     Voice calls from data communications networks can be converted to a traditional voice telephony interface and be presented to enter PBX  104   a  via gateway devices such as gateway device  103   a . These voice calls can enter the data communications network either from other gateways, such as public network gateway  119  and private network gateway  103   b , or from Workstations equipped with telephony voice cards, for example workstation  112   a , equipped with voice card  110   a . The workstations can be any computer, for example PCs, which can be configured to perform the functions described in the present specification. Such PCs are well-known to those skilled in the art and will not be described further. In addition, voice calls can enter the data communications network from the tandem switch  125  through the LS gateway  105   c.    
     Control of signaling for standard telephony is contained in either the voice band as tones passed between devices, out-of-band in associated signaling bits (e.g., in T1 or E1), or in messages contained in a separate data channel (e.g., in ISDN or SS 7 ). Regardless of the telephony technology chosen it is preferable that devices use standard protocols for communication. Such a protocol might involve change of bit value in an associated signaling bit in a T1 channel. On-hook and off-hook are examples of states communicated through T1 A-bit signaling. These states are part of a higher layer protocol of signaling, for instance E&amp;M in the T1 trunking world. In any case, software controlling call processing on switching devices interprets the signaling bit values through time as indications of state changes in telephony protocol. Where messages are available for building telephony protocols call processing state changes are driven by messages and field values within these messages. ISDN&#39;s Q.931 message protocol, hereby incorporated by reference in its entirety, is an example of such a message driven protocol. 
     In the VON telephony standard, message passing protocols exist for building telephony call processing software. The H.323 specification, hereby incorporated by reference in its entirety, is an example of such a protocol. 
     In the preferred embodiment of the present invention, LS  105   a  coordinates input from switches and VON sources. This coordination results in seamless presentation of telephony and VON calls to users. Referring to FIG. 4, the major components of LS  105   a  are described. LS  105   a  contains both standard telephony hardware such as T1, analog handset, or digital handset telephony interface hardware  200  as well as DSPs  201 , Ethernet cards  202 , and switch link control capabilities  203 . Switch link control is accomplished by sending switch link control commands over serial line  107  to a serial port  206  on PBX  104   a . Such control is well-known to those skilled in the art. Coordination of traditional telephony call traffic to users through such a server is described in the &#39;668 patent. 
     Users handle call traffic through their telephony interface software running on their desktop workstations. Calls are presented to their software via messages between the LS and Workstation. These messages can also reflect call processing status of calls presented to the server. Calls originating from VON sources are also presented to the users by the same messages to their telephony interface software. 
     In standard telephony a call is delivered to a user via a switch-hook transfer, a sequence of signaling messages or signaling bit changes over time accompanied by DTMF tones or by messages to the switching system through a switch link. In VON, call setup is accomplished via messages passed between originating VON processes (remote processes in this example) and the local VON user. In both technologies the link server helps match the remote user to the local user. For example, in a call center the local user may not be known until the call exits the queue (e.g., when the agent (local user in this example) selects a call from the queue for processing). Thus, for a time the link server may become the local user to play messages and collect in-band information from the remote user. 
     FIG. 7 illustrates schematically call setup message flow for a VON call according to a preferred embodiment of the present invention. This message flow is similar to that described in details of the H.323 message processing, which is available from the International Telecommunications Union&#39;s: “Draft Recommendation H.323: VISUAL TELEPHONE SYSTEMS AND EQUIPMENT FOR LOCAL AREA NETWORKS WHICH PROVIDE A NON_GUARANTEED QUALITY OF SERVICE,” hereby incorporated by reference in its entirety. The two endpoints communicate directly with each other. In the present example, a SETUP message is sent from PC  112  to PC  140  over network  115   a . Messages are sent via the message router  405 . A call control process in the workstation, WCP  451 , handles the incoming SETUP message. In response, PC  140  sends a CALL PROCEEDING message and an ALERTING message to PC  112 . To establish the voice connection through the network a CONNECT message is sent from PC  140  back to PC  112 . The CONNECT message carries a transport channel address to which PC  140  can connect to begin communications. In a preferred embodiment of the present invention this is a TCP/IP socket address. For example, in the H.323 standard this is an H.245 Control Channel Transport Address. 
     The socket address is the repository and source of voice packets carried to and from the Voice card  110   a  in the PC  140 . Packets are moved to and from the socket by software in the driver for the voice card  110   a . Packet movement is performed by a method known to those skilled in the art of device driver design. Voice card  110   a  converts packets to analog audio signal for presentation to a standard telephone or headset  301 . The voice card also takes input from telephone  301  (see FIG. 4) and converts it to packets for passing to the socket address. Voice card  110   b  in PC  112   a  performs similar functions to enable two way VON voice conversation. 
     In a preferred embodiment of the present invention a user (e.g., an agent) can add a conversation from another input to a conversation being handled by a voice card, for example voice card  110   a . Preferably, voice card  110   a  has interfaces shown in FIG.  8 . Other voice cards in the system preferably have a similar configuration. A local Phone interface  421  may connect to a Tip and Ring or digital telephone interface such as ISDN BRI. Outgoing and/or incoming calls can be handled from this interface. In this example an outgoing call is placed. Code running in the WCP process  451  sends commands to the DSP through a shared memory  429  (not shown) to place an outgoing call. The line interface  422  is put in an off hook condition, dial tone is detected, and touch tones are generated by the DSP  430 . The call input on line interface  422  can be conferenced into the VON conversation through conferencing software stored in a DSP memory  431  (not shown) and run on DSP  430 . It would be within the knowledge of those skilled in the art of DSP programming techniques to program DSP  430  with the required audio mixing and conferencing software so that a VON conversation can be conferenced as described above. 
     More calls can be added to the conference call. These calls can be either VON or telephony calls. For example, an additional telephone caller can be added by passing control information to the voice card, for example voice card  110   a , to perform a switch-hook transfer. The switch-hook transfer command is followed by the dialing of a dial-string, which may include a feature code such as a conference dial code, to the PBX. This action will add on a telephony caller to the conference call. 
     A VON party can be added as well. TO add a VON party a VON call is placed as described in the VON call origination message descriptions. Adding the established VON call to the conference is accomplished through conference circuit control on the card. 
     Adding parties to a conference via telephony devices, such as the PBX, is limited to the number of callers the PBX will support. Adding VON parties to a conference is limited by the processing power of the DSP on the voice card such as voice card  110   a  on PC  140 . Each VON party&#39;s voice stream must be decoded to a PCM data stream. This data can then be mixed or otherwise signal processed. Once processed this data must be encoded back to network ready form (for example G723.1 compressed format, which is hereby incorporated by reference in its entirety). 
     The present invention allows versatility in telephony call control. For example, in the conference call example discussed above, a telephony call can be added to the conference by using a Switch Link Proxy Process (SLPP  406 ) executing on an LS, such as LS  105   a . The SLPP  406  process receives messages through message router  405  (FIGS. 3 and 5) to affect call control in the telephony network by sending control messages to PBX  104   a . This interface is bidirectional. That is, SLPP  406  also receives messages from PBX  104   a  and passes them to WCP  451  processes in the workstations. These messages are used to both monitor devices in the switch and cause device control actions. 
     In a preferred embodiment of the present invention, the SLPP  406  must maintain a map table of circuit identifiers in PBX  104   a  to the WCP  451  process ID&#39;s. This map is created at initialization of the WCP  451  software by the local user. The map creation process is illustrated in FIG.  11 A. Referring to FIG. 11A, WCP  451  registers with message router  405  and the SLPP  406  when it is initialized by sending a LOGIN message to message router  405 . Message router  405  forwards the LOGIN message to the SLPP  406  process so it can make a map entry. The PBX circuit ID must be known to the WCP  451  process at initialization time. This is entered into WCP  451  by the installer. Message router  405  sends a REGISTER-OK (confirmation) message back to WCP  451  when registration is successfully completed. 
     FIG. 11 shows how a call is placed from WCP  451  by sending an ORIG message through message router  405  to the SLPP  406  to PBX  104   a . The ORIG message is an origination message. This call may be added to the conference by configuring the voice card  110  through the software driver interface discussed above. When the call is completed the PBX sends a DISCONNECT message through SLPP  406  to message router  405 , which forwards the DISCONNECT message to WCP  451 . WCP  451  then ends the call from the workstation software&#39;s point of view. Any screens, displays, etc are reset to show the call is finished. 
     Incoming calls may be answered and added to the conference by selection from a list of queued calls presented through the LS  105   a  in a manner described in the &#39;668 patent. Calls are either held at the LS  105  in ports on standard telephony cards  200  or held in PBX  104   a . Calls held in PBX  104   a  are controlled or monitored through messages passed between the LS  105   a  and the PBX  104   a  over switch link  107 . Switch link  107  is shown in FIG. 4 as a serial port interface to a serial card  206  in the PBX. It would be apparent to those skilled in the art that switch link  107  can be implemented using a number of communication methods, including Ethernet or TCP/IP socket connections. 
     To move a call from the LS  105   a  which is held on a port at a telephony interface card  200  an ACCEPT message from WCP  451  in PC  112  is sent to the CCP  450  process controlling the call. CCP  450  sends commands to the card  200  to cause a transfer of the call to the line  109  connected to the card  110  in PC  112 . The telephony interface card  200  then passes signaling information to the PBX—this could be switchhook flash and DTMF or a digital signaling information used in T1 or digital handset. 
     If a switch link such as switch link  107  is used, rather than direct control by the telephony interface card the signaling scenario is the same. Messages between WCP  451  and CCP  450  processes are the same. However, the transfer is accomplished by messages sent from the CCP  450  process to the SLPP  406  process to the PBX. 
     In addition to the point-to-point calls that are discussed above, Web server  123  or Link Server  105   a  can be used to set up calls from either the VON domain or the telephony domain. LS gateway  105   a  can be used to set up VON calls or even queue VON calls by taking SETUP messages from remote VON user processes  402  and passing connect messages back to LS  105   a . FIG. 9 shows how LS  105   a  can answer, queue, and distribute a call in a call center. A SETUP message is passed to a Network Call Control Process (NCCP)  400  in LS  105   a  from WCP  451  in PC  111 . LS  105   a  sends a CONNECT message back to PC  111  so as to answer the call. The LS  105   a  may now play messages to PC  111  via the VON voice path. At this time an ANNOUNCE message is broadcast through the message router  405  to all WCP  451  processes such as WCP  451  executing on PC  140 . Software in WCP  451  processes displays calls to agents at these workstations in a manner similar to that described in the &#39;668 patent. An agent wishing to take the call will select it through the WCP  451  user interface. When an agent selects a call for processing from the queue, an ACCEPT message is generated and sent back to NCCP  400 . The ACCEPT message a Call Signaling Channel Transport Address (CSCTA). NCCP  400  is now able to distribute the call. It passes a FACILITY message back to the WCP  451  process in PC  111  to inform this process of the intent to change the destination of the call. The WCP  451  process responds with a RELEASE COMPLETE message back to the LS  105   a  to end this call. The WCP  451  process next sends a SETUP message to PC 140 . A CONNECT message is returned to complete the transfer of the call. When the call is completed, PC  140  can send a RELEASE COMPLETE message to PC  111  to end the call. 
     In VON, Web server  401  can also play an important part in distribution of a VON call. For example, a remote user browsing a web site desires to establish a VON call to a person or to a call center in a company. In a preferred embodiment of the present invention, Mustrated in FIG. 10, information including the CSCTA of the user is passed from a User  402  to the Web Server  401  in a REQUEST message. Web server  401  passes this information and other information about the User  402  entered during interactions in the Web Server  401  to the NCCP  400  process running on LS  105  in a WEB_REQUEST message. Data regarding the call handling time, such as position in queue, is passed back to the Web Server as a stream of DATA (See HTTP 1.1 Proposed Standard RFC 2068, hereby incorporated by reference in its entirety, as an example of Web and Browser interaction messaging). This queue information is then passed to the User  402  in a RESPONSE message from the Web server. After the DATA message is sent the NCCP  400  process sends an ANNOUNCE message through the message Router  405  to all the WCP  451  processes such as WCP  451  executing on PC  111 . A user corresponding to one of the WCP  451  processes selects the call. Upon selecting the call, an ACCEPT message is sent from the WCP process requesting the call to the NCCP  400 . NCCP  400  sends an ANSWER message to all the WCP  451  processes to manage their call list information. The WCP  451  process electing to take the call takes the CSCTA value from this message and other address information of the USER  402  and sends a SETUP message directly to it. The User  402  sends a CONNECT back to complete call setup. When the call is complete, WCP  451  in PC  111  sends a RELEASE COMPLETE message to user  402  to end the call. 
     The present invention also applies to other media routing. Although the PBX and switching infrastructure this invention is designed to supplement and enhance delivery of voice media, the VON call routing discussed herein can be applied to video or data conferencing. One of the key technologies leveraged by this invention is H.323 call control messaging. This specification also allows other media extensions. Coordinating the setup and tear-down of media sessions is perhaps a better way of describing the capabilities of H.323. This invention makes it possible to blend this type of media control with existing switching systems in either a single or multi-center environment. 
     In addition the present invention applies to multi-center environments. Servers in remote sites need to pass call processing messages to effect a seamless control structure between local and remote agents. To accomplish this, message router  405  needs to contain message ports to remote message routers in remote link servers. Link servers register with each other to enable a communication path between sites. Messages are sent to a remote router executing on a remote link server to inform it of the need to pass call processing messages based on certain filter criteria. For example, the filter criteria can be agent specific, call-type and/ or maintenance. It would be apparent to those skilled in the art that other filter criteria can be used. 
     Message router  405  must keep a table containing remote router ID&#39;s (and their associated port ID) and message passing filter information. FIG. 12 shows entries in an example routing table  1201 . Referring to FIG. 12, routing table  1201  preferably contains four fields: remote route ID, remote port number, filter type and filter value. The remote router ID identifies with which remote message router message router  405  is in communication. The remote port number corresponds to the port number of the remote message router with which message router  405  is in communication. The filter type is the filter criteria to filter out message of a specific type. The filter value is the value associated with the filter type for more specific message filtering. Filtering can be preformed with respect to the sender and/or receiver. Thus, the filter can affect message prior to their sending and/or upon their receipt. The purpose of the filter is to keep network traffic between message routers to a minimum, but is not necessary to practice the present invention. 
     A ROUTER_LOGIN message sent between two Router  405  processes results in an assignment of a port and initial entry in the table. A ROUTER_SET message between two Router  405  processes places an entry in the table which sets Filter Type and Value. 
     Entries in routing table  1201  are consulted when the Router  405  processes a message. Typical Router message processing is described in the &#39;668 patent. Processing of entries in this table constitute an additional step to this processing. This additional step includes consulting the table first for the presence of any Router ID&#39;s and checking message values against filter values for a decision on whether to pass the message on to the remote Router  405  process. This filtering and table management is necessary in order to build linked multi-center communications systems. Without such a scheme the message traffic between the sites would grow exponentially—severely limiting the number and size of remote centers which could be linked together. 
     The addition of link servers and WCP processes makes it possible to build virtual dialing plans and virtualize the communications addressing and connectivity between diverse switching environments. Adding the capabilities of NCCPs  400 , voice cards such as voice cards  110   a  and  110   b  and link servers such as link servers  103   a ,  105   a ,  103   b  and  105   b  makes it possible to blend VON or media on network into the virtualized communications described. In addition, a customer can migrate all communications to the network devices LS  105  and PC  112  with voice card devices. This is illustrated in FIGS. 2 and 3 where calls from PBX  104   a  are passed to LS Gateway  103 . 
     Messages are passed to setup calls as described above. WCP  451  processes such as those running in workstation  112  receive and process these messages as before with the exception of the voice path. Voice card  110  is configured in this scenario so that the voice path is performed by packet delivery over the network between voice card  110  and a port on the Ethernet TCP/IP  202  card in the LS Gateway  103 . The combination of Standard telephony card  200 , DSP card  201 , and Ethernet TCP/IP  202  card are like those provided by Natural Microsystem&#39;s Fusion product. An example Standard telephony card  200  is Natural Microsystems (NMS) ATI  24  card, DSP  201  card NMS&#39;s AG-RT Daughter card, Ethernet TCP/IP  202  card NMS&#39;s TX2000 IP router card. These cards convert standard telephony signaling to packetized messages controllable by software such as that described in this invention. The compression and coding schemes used on these cards need to match those used at the PC  112  and Card  110 . 
     FIG. 13 is a flow chart representative of a process executed in by message router  405  prior to sending a message to a remote message router according to a preferred embodiment of the present invention. To send a message, message router begins in start step  1302 , whereupon it immediately enters step  1304 . In step  1304 , message router  405  receives a message to send. It then processes he message by router identification in steps  1306  and  1308 . If the message is for a remote message router, then it continues in step  1312 , else there is nowhere to send the message and it ends in done step  1310 . After determining where to send the message, message router  405  checks filter type in step  1314  to determine if the receiver is a receiver of the correct type. If not, the router is finished and proceeds to done step  1310 . Otherwise, the router checks the filter value to determine if the receiver value is correct for the remote router. If not message router  405  stops processing the message and proceeds to done step  1310 . If the message has the correct filter value, message router  405  sends the message to the remote router in step  1316 . 
     The foregoing disclosure of embodiments of the present invention has been presented for purposes of illustration and description. It is not intended to be exhaustive or to limit the invention to the precise forms disclosed. Many variations and modifications of the embodiments described herein will be obvious to one of ordinary skill in the art in light of the above disclosure. The scope of the invention is to be defined only by the claims appended hereto, and by their equivalents.