WO1999020032A1 - System and method for integrating voice on network with traditional telephony - Google Patents

System and method for integrating voice on network with traditional telephony Download PDF

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Publication number
WO1999020032A1
WO1999020032A1 PCT/US1998/019715 US9819715W WO9920032A1 WO 1999020032 A1 WO1999020032 A1 WO 1999020032A1 US 9819715 W US9819715 W US 9819715W WO 9920032 A1 WO9920032 A1 WO 9920032A1
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WO
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Patent type
Prior art keywords
voice
network
call
telephony
message
Prior art date
Application number
PCT/US1998/019715
Other languages
French (fr)
Inventor
Patrick K. Brady
Original Assignee
Apropos Technology
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Publication date

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Interconnection arrangements between switching centres
    • H04M7/009Interconnection arrangements between switching centres in systems involving PBX or KTS networks
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/50Centralised arrangements for answering calls; Centralised arrangements for recording messages for absent or busy subscribers ; Centralised arrangements for recording messages
    • H04M3/51Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/56Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Interconnection arrangements between switching centres
    • H04M7/12Interconnection arrangements between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step, decimal and non-decimal, circuit-switched and packet-switched, i.e. gateway arrangements
    • H04M7/1205Interconnection arrangements between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step, decimal and non-decimal, circuit-switched and packet-switched, i.e. gateway arrangements where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/50Centralised arrangements for answering calls; Centralised arrangements for recording messages for absent or busy subscribers ; Centralised arrangements for recording messages
    • H04M3/51Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing
    • H04M3/523Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing with call distribution or queueing
    • H04M3/5237Interconnection arrangements between ACD systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04QSELECTING
    • H04Q3/00Selecting arrangements
    • H04Q3/58Arrangements providing connection between main exchange and sub-exchange or satellite
    • H04Q3/62Arrangements providing connection between main exchange and sub-exchange or satellite for connecting to private branch exchanges
    • H04Q3/625Arrangements in the private branch exchange

Abstract

A link server (105a, 105b) 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 (104a, 104b), or forwarding calls directly to a desktop.

Description

SYSTEM AND METHOD FOR INTEGRATING VOICE ON NETWORK WITH TRADITIONAL TELEPHONY

This application is a continuation-in-part of U.S. Patent Application

serial number 08/813,970, filed on March 3, 1997 (still pending), hereby

incorporated by reference in its entirety, which is a continuation of U.S.

Patent Application Serial No. 08/758,063, filed November 27, 1996 (still

pending), hereby incorporated by reference in its entirety, which is a

continuation of U.S. Patent Application Serial No. 08/595,861, filed

February 6, 1996 (now abandoned) which is a divisional of U.S. Patent

Application Serial No. 08/450,268, filed May 25, 1995 (now U.S. Patent No.

5,557,668 (the "668 patent") which is a continuation of U.S. Patent

Application Serial No. 07/904,196, filed June 25, 1992 (now abandoned),

hereby incorporated by reference in its entirety, and claims the benefit of

U.S. Provisional Patent Application Serial No. 60/059,285, filed September

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 Tl 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's interests as well as history of interaction with a company'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

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 (Tl, analog, digital handset, or CTI 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 '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's PC (e.g., the IP

address) so that a point-to-point connection can be established between the

agent'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'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'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 '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. 1 IB 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) 104a and 104b connected to

Central Office (CO) switches, 100 and 120, and desktop telephones 108a,

108b and 113a and 113b. PBXs 104a and 104b are preferably conventional

voice telephony premise switches. Telecommunication system 10 also

includes local area networks ("LAN") 115a and 115b and Link Servers ("LS")

105a and 105b as well as gateways 103a and 103b. Connected to network

115a are data communications gateways 116a and 114a. These allow the

LAN 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 105c.

Web server 123 is connected to the LAN 115a to accept web

interactions from the local and wide area network connections. A web

server (not shown) can also be attached to LAN 115b in like manner.

Attached to the data networks shown are various communications gateways and VON gateways (103a, 103b, 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 Tl or other standard telephony

communications methods 101a and 101b.

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

104a 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 104a through

CO 120), voice calls placed within the company'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 Tl 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 108a or

multibutton digital display phone 113a.

Voice calls from data communications networks can be converted to a

traditional voice telephony interface and be presented to enter PBX 104a via

gateway devices such as gateway device 103a. These voice calls can enter

the data communications network either from other gateways, such as

public network gateway 119 and private network gateway 103b, or from

Workstations equipped with telephony voice cards, for example workstation

112a, equipped with voice card 110a. 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 105c.

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 Tl or El), or in messages contained in a separate data

channel (e.g., in ISDN or SS7). 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 Tl channel. On-hook and off -hook are examples

of states communicated through Tl A-bit signaling. These states are part of a higher layer protocol of signaling, for instance E&M in the Tl 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. ISDNs 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 105a

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 105a are described. LS 105a contains

both standard telephony hardware such as Tl, 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 104a. 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 '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'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 115a. 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 110a in the PC 140. Packets are moved to and from the socket by software in the driver for the voice card 110a. Packet

movement is performed by a method known to those skilled in the art of

device driver design. Voice card 110a 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 110b in PC 112a

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 110a. . Preferably, voice

card 110a 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 110a, 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 110a on PC 140. Each VON party'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 105a. 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 104a. This interface is bi-directional. That is, SLPP 406 also receives messages from

PBX 104a 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 104a to the WCP

451 process ID'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.

Figure 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 104a.

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'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 105a in a

manner described in the '668 patent. Calls are either held at the LS 105 in

ports on standard telephony cards 200 or held in PBX 104a. Calls held in

PBX 104a are controlled or monitored through messages passed between the

LS 105a and the PBX 104a 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 105a 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 Tl or digital handset.

If a switch link such as switch linkl07 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 105a can be used to set up calls from either the

VON domain or the telephony domain. LS gateway 105a 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

105a. Figure 9 shows how LS 105a 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 105a from WCP 451 in PC 111. LS 105a sends a

CONNECT message back to PC 111 so as to answer the call. The LS 105a

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 '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 105a to end this call. The WCP 451

process next sends a SETUP message to PC140. 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, illustrated 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'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

'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'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 110a and 110b

and link servers such as link servers 103a, 105a, 103b and 105b 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 104a 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's Fusion product. An

example Standard telephony card 200 is Natural Microsystems (NMS) ATI

24 card, DSP 201 card NMS's AG-RT Daughter card, Ethernet TCP/IP 202

card NMS'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.

Claims

What is Claimed is:
1. A system for blending a telephony call traffic with voice on network call traffic for distribution, comprising:
a link server to acquire the voice on network data;
a voice card located on said link server for processing said voice on
network data and the telephony data; and
a message router for sending control messages to said voice card to control said processing to distribute the voice on network data and the
telephony data.
2. The system as recited in claim 1, further comprising a public branch
exchange (PBX), wherein the telephony call is routed by said PBX to said
link server for processing with said voice on network data.
3. A method for blending telephony call traffic and voice on network call
traffic for distribution, comprising the steps of:
(a) acquiring the voice on network call traffic in a link server
(b) processing said voice on network call traffic and said telephony
data in a voice card in said link server; and
(c) sending control messages to the voice card to control said
processing to distribute the voice on network call traffic and the telephony
call traffic.
4. The method recited in claim 2, further comprising the step of routing
the telephony call traffic through a PBX to said link server for processing
with said voice on network traffic.
5. A system for distributing calls in a call center, comprising:
one or more agent computers, each agent computer operated by an
agent;
a link server to acquire the voice on network data coupled to said one
or more agent computers by a local area network (LAN);
a voice card located on said link server for processing said voice on
network data and the telephony data;
a message router for sending control messages to said voice card to
control said processing to distribute the voice on network data and the
telephony data to said one or more agents.
PCT/US1998/019715 1997-09-18 1998-09-18 System and method for integrating voice on network with traditional telephony WO1999020032A1 (en)

Priority Applications (2)

Application Number Priority Date Filing Date Title
US5928597 true 1997-09-18 1997-09-18
US60/059,285 1997-09-18

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
EP19980951923 EP1038385A1 (en) 1997-09-18 1998-09-18 System and method for integrating voice on network with traditional telephony
CA 2303840 CA2303840A1 (en) 1997-09-18 1998-09-18 System and method for integrating voice on network with traditional telephony

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Cited By (13)

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Publication number Priority date Publication date Assignee Title
EP0961508A2 (en) * 1998-05-29 1999-12-01 Siemens Aktiengesellschaft Method and telecommunication system for transmission of data from a first to a second private branch exchange
EP0961508A3 (en) * 1998-05-29 2006-01-18 Siemens Aktiengesellschaft Method and telecommunication system for transmission of data from a first to a second private branch exchange
WO2001035617A3 (en) * 1999-10-29 2002-09-12 Telera Inc Distributed call center with local points of presence
WO2001035617A2 (en) * 1999-10-29 2001-05-17 Telera, Inc. Distributed call center with local points of presence
WO2001078443A2 (en) * 2000-04-06 2001-10-18 Arialphone, Llc. Earset communication system
WO2001078443A3 (en) * 2000-04-06 2003-10-16 Arialphone Llc Earset communication system
WO2002045399A2 (en) * 2000-11-28 2002-06-06 Nortel Networks Limited A component for processing ip telephony calls
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EP1278352A3 (en) * 2001-07-18 2003-11-19 Tenovis GmbH & Co. KG IP Telecomms Equipment and Method for Operating
EP1278352A2 (en) * 2001-07-18 2003-01-22 Tenovis GmbH & Co. KG IP Telecomms Equipment and Method for Operating
DE102004061512A1 (en) * 2004-12-16 2006-06-29 Deutsche Telekom Ag Automated determination of processing instance involves determining processing instance(s), using routing algorithm capable of learning, suitable for processing ticket that has been produced within predefined period of time

Also Published As

Publication number Publication date Type
CA2303840A1 (en) 1999-04-22 application
EP1038385A1 (en) 2000-09-27 application

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