Patent Document

FIELD OF THE INVENTION 
   The present invention relates generally to enhanced call services, and in particular, to a method and apparatus that sponsors packet based voice over Internet protocol (VOIP) telephone services such as, for example, call waiting and conference calling. 
   BACKGROUND OF THE INVENTION 
   Packet-based communication systems are well known. Such systems allow user terminals (e.g., personal computers (PCs), personal digital assistants (PDAs), telephones, mobile radiotelephones, modems, network access devices, Internet peripherals, and their equivalents) to communicate with each other and with other communication networks, such as, for example, the Internet and the public telephone network. 
   Enhanced call services within packet-based communication systems are also well known. Such services include, but are not limited to offerings such as call waiting, call forwarding, and call conferencing, including 3-way calls, multi-party calls, and the like. As will be appreciated, the typical call service is implemented and controlled by the governing communication system, and not the user terminal. 
   By way of example, and with reference to  FIG. 1 , it will be appreciated that establishment of the typical conference call requires multiple steps and various network entities. First, a day and time for the conference call must be scheduled. After scheduling, resources to facilitate the call must be reserved. For example, a conference bridge for an audio or video portion of the conference is reserved. Once the necessary resources are reserved and allocated for the conference call, conference access data may need to be supplied to each participant. For example, a conference bridge number and password may need to be distributed to conference participants to permit calling into the conference bridge. Once call setup is complete the actual conference call requires additional steps. Namely, each conference participant must connect to the conference call at the appropriate time with the appropriate capability to interact with other participants, including possibly the exchange of audio data, video data, files, conference presentation materials and the like. 
     FIG. 1  is a block diagram of a communications system  100  that supports enhanced call services. The communications system  100  depicts a third generation wireless system, as defined by the 3 rd  Generation Partnership Program, also known as 3 GPP (see 3 gpp.org). In such a system, terminals  102  are typically mobile radiotelephone devices that include a user interface and an interface for coupling to communications system  100 . The user interface of user terminal  102  is typically referred to as terminal equipment and generally includes an audio interface, such as a microphone and speakers, a visual interface, such as a display, and a user input interface, such as a keyboard or touch pad. The interface for coupling to the system  100  is typically referred to as a mobile terminal and generally includes an over-the-air interface for transmitting and receiving data. In the typical environment, base stations  104  include an over-the-air interface that is complementary to the over-the-air interface of user terminal  102 , thereby permitting terminal  102  and base stations  104  to communicate over-the-air. When the interface employed by user terminal  102  and base station  104  is an over-the-air interface, the communication system  100  is generally referred to as a wireless communications system. When the interface employed by user terminal  102  is a packet-based protocol, the communication system  100  is generally referred to as a packet based communications system. 
   During operation, the communications that are directed to and received from user terminals  102  via base stations  104  are coordinated and transferred using a serving device, such as a GPRS (GSM Packet Radio System) support node (SGSN)  106 , a gateway GPRS support node (GGSN)  110 , a call session control function (CSCF)  114  and a home subscriber system  118 . SGSN  106  coordinates transmissions to and from base stations  104 . SGSN  106  is coupled to GGSN  110  via a data link  112 . GGSN  110  interfaces the communications to and from SGSN  106  to other networks. Call session control function  114  is coupled to GGSN  110  via a data link  116 . Call session control function  114  coordinates and executes a signaling protocol used to establish, maintain and control calls or sessions for communications involving user terminals  102 . A home subscriber system  118  is coupled to call session control function  114  via a data link  120 . Home subscriber system  118  includes subscriber profile information, including information traditionally associated with a home location register for a mobile subscriber. 
   To facilitate ancillary and support functions within communications system  100 , a charging gateway function (CGF)  122  and a media resource function  124  may be provided. Charging gateway function  122  is coupled to SGSN  106  and GGSN  110  to account for packets passing through these elements for accounting, billing and other purposes. Media resource function  124  is coupled to call session control function  114  and to GGSN  110 . Media resource function  124  provides resources for conference bridging, tones, announcements, text-to-speech conversion, automatic speech recognition and other service functions for communications through GGSN  110 . 
   GGSN  110  couples user terminals  102  to other networks. In particular, GGSN  110  is coupled to an Internet protocol (IP) network  146  via a data link  148 . Data link  148  preferably implements a packet-based protocol for transfers to a data network. Data link  148  and IP network  146  provide access to any elements connected to IP network  146 , such as, for example, a computer  154 . GGSN  110  is also coupled to a media gateway  130  via a data link  150 . Media gateway  130  is in turn coupled to a public switched telephone network  142  via a communications link  152 . Media gateway  130  converts packetized voice received from GGSN  110  to a circuit-switched protocol acceptable to the public switched telephone network  142 . Conversely, media gateway  130  converts circuit-switched communications received from public switched telephone network  142  to packetized communications acceptable to GGSN  110 . Media gateway  130 , data link  150 , and communications link  152  provide an interface for user terminals  102  to the public switched telephone network  142 . By virtue of this connection, user terminals  102  are coupled to elements attached to the public switched telephone network, such as telephone  144 . 
   The signaling and control necessary to interface GGSN  110  with public switched telephone network  142  is controlled and provided by call session control function  114 , a media gateway controller  126 , and a transport signaling gateway  134 . Media gateway controller  126  is coupled to call session control function  114  via a data link  128 . Media gateway controller  126  is coupled to control media gateway  130  via data ink  132 . Call session control function  114  determines, based on a signaling protocol, the media gateway resources required for a particular communication or session. These resource requirements are transmitted to media gateway controller  126 , which in turn, configures and establishes the necessary resources in media gateway  130  and provides the necessary signaling to transport signaling gateway  134 . The resources in media gateway  130  are configured to transfer the actual (bearer) data between the GGSN  110  and the public switched telephone network  142 . Transport signaling gateway  134  converts the signaling protocol from the media gateway controller  126  to a signaling protocol used by the public switched telephone network  142 . 
   Applications and enhanced call services are preferably coupled to communication system  100  for use in interaction with user terminals  102 . In particular, call session control function  114  is coupled to an applications and call services network  156  via a data link  158 . A myriad of call services and applications may reside in or be coupled to the network  156 , including database services from database  160 . For additional detail, the interested reader may refer to U.S. patent application Ser. No. 09/953,509, filed Sep. 14, 2001, entitled “TARGETED AND INTELLIGENT MULTIMEDIA CONFERENCE ESTABLISHMENT SERVICES,” invented by Henrikson et al., and assigned to the assignee of the present application. 
   The steps required to establish a successful conference call under the system of  FIG. 1  are cumbersome and time consuming, require considerable network resource, network administration, and network-based call session control. In addition, these steps may also require the scheduling of network resources and caller participation well in advance of the anticipated call. In general, these call operations are not characterized as real-time or spontaneous, and more importantly, they are not network independent. Notwithstanding these shortcomings, enhanced call services are of increasing importance and value to system subscribers. The advent and proliferation of proposed third generation wireless systems, coupled with the promise of ubiquitous availability will only make such call services more desirable. 
   In anticipation thereof, a growing need exists for on-demand call conferencing and multi-party call waiting routines that reduce the cumbersome nature and network dependence exhibited by existing call services that support call conferencing and call waiting. 
   SUMMARY OF THE INVENTION 
   In accordance with one aspect of the present invention a packet data terminal implements call waiting and call conferencing services with little or no network support. The packet data terminal in accordance with the invention employs a digital-to-analog (D/A) converter for converting a first and a second packet data stream into separate analog representations. Thereafter, a selective mixer selectively manipulates the analog representations to provide either a mixed or a non-mixed output. A mixed output is characteristic of the provision of call conferencing service. A non-mixed output is characteristic of the provision of call waiting service. The data terminal employs a digital mixer that converts analog voice into a packet data stream. A multiplexer circuit distributes the packet data stream to a plurality of call sessions during a conference call while distributing the packet data stream to a single call session for purposes of call waiting service. In accordance with another aspect of the invention, the data terminal employs a user input device permitting users to select the establishment of call waiting and/or call conferencing service. 
   In accordance with another aspect of the invention, a method is provided for establishing conference call service. The method begins with the receipt of a plurality of digital voice data streams by the data terminal, each digital voice data stream representing a separate call session. Each digital voice data stream is then converted into a separate analog representation corresponding to a call session. An election is made by the device user that selects call sessions of interest. Analog representations of the call sessions of interest are combined one with another to produce a first mixed output. The first mixed output is combined with user-generated voice signals to produce a second mixed output, which, in turn, is converted into a packet data stream. The packet data stream is then routed to the call sessions of interest to complete the conference call. 
   In another aspect of the invention, a method is provided for establishing call waiting service. The method begins with the receipt of a plurality of digital voice data streams by the data terminal, each representing a separate call session. Each digital voice data stream is then converted into a separate analog representation corresponding to a call session. An election is made by the device user that selects a call session of interest. An analog representation of the call session of interest may be combined with user-generated voice signals, to produce an analog output, which, in turn, is converted into a packet data stream. The packet data stream is then routed to a single call session of interest for purposes of supporting call waiting. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  is a block diagram of a prior art communications system that provides network-based call establishment services; 
       FIG. 2  is a block diagram of a communications system that provides on-demand, network independent, call establishment services in accordance with the present invention; 
       FIG. 3  is a first block diagram of a user terminal of  FIG. 2 ; 
       FIG. 4  is another block diagram of a user terminal of  FIG. 2 ; 
       FIG. 5  is a flow chart illustrating a method for establishing call waiting service in accordance with the present invention; 
       FIG. 6  is a flow chart illustrating a method for establishing conference call service in accordance with the present invention. 
   

   DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     FIG. 2  is a block diagram of a communications system that provides on-demand call establishment services in accordance with the present invention. The communications system  200  depicts, in part, a third generation wireless system, as defined by the 3 rd  Generation Partnership Program, also known as 3 GPP (see 3 gpp.org). In such a system, terminals  102   a  may be mobile radiotelephone devices, personal digital assistants (PDAs), modems, network access devices, Internet peripherals, and the like. Such wireless terminals  102   a  generally include a user interface and an interface for coupling to communications system  200 . The user interface of a terminal  102   a  is often referred to as terminal equipment. The user interface generally includes an audio interface, such as a microphone and speaker, a visual interface, such as a display or graphic user interface (GUI), and a user input interface, such as a keyboard, touch pad, keypad, touch screen, track-ball system, voice recognition system, hand writing recognition system, or combinations thereof. 
   The interface for coupling wireless terminals  102   a  to the system  200  is typically referred to as a mobile terminal and generally includes an over-the-air interface for transmitting and receiving data. In the typical environment, base stations  104  include an over-the-air interface that is complementary to the over-the-air interface of user terminal  102   a , thereby permitting terminals  102   a  and base stations  104  to communicate. While the suggested over-the-air interface is one defined by 3 GPP (see 3 gpp.org), it will be appreciated by those skilled in the art that several other wireless interfaces are known in the art and may be substituted therefore, without departing from the spirit of the present invention. 
   During operation, the communications that are directed to and received from user terminals  102   a  via base stations  104  are coordinated and transferred using a serving device, such as a wireless access gateway (WAG)  202 . In accordance with a preferred embodiment, when user terminals  102   a  are mobile radiotelephones, WAG  202  may consist of the GPRS (GSM Packet Radio System) equipment ( 106 – 124 ) described in association with  FIG. 1 . As will be appreciated after review hereof, WAG  202  may also couple user terminals  102   a  to other networks. In accordance, WAG  202  is also shown coupled to an Internet protocol (IP) network  146  via well-known data links (not shown). Such data links implement packet-based protocols providing access to any elements connected to IP network  146 , such as, for example, a telephone  144 , through a Public Switched Telephone Network (PSTN)  142 . 
   With further reference to  FIG. 2 , IP network  146  is shown coupled to PSTN gateway  204  via a data link (not shown). As previously discussed, such data links implement well known packet-based protocols within the knowledge of those skilled in the art, and therefore are not described herein in detail. PSTN gateway  204  is in turn coupled to PSTN  142  via communications link  152 . During operation, PSTN gateway  204  converts packetized voice received from WAG  202  to a circuit-switched protocol acceptable to PSTN  142 . Conversely, PSTN gateway  204  converts circuit-switched communications received from PSTN  142 , to packetized communications acceptable to WAG  202 . By virtue of this connection, user terminals  102   a  are coupled to devices attached to the PSTN  142 , such as telephones  144 . 
   The communications system  200  also depicts, in part, a local area network (LAN) communication system, as may be defined by the Institute of Electronic and Electrical Engineering (IEEE), American National Standards Institute (ANSI), European Transmission Standards Institute (ETSI), or other similar governmental or industry standards organization. In such a system, user terminals  102   b  may be wired or wireless devices such as, but not limited to, personal computers (PCs), personal digital assistants (PDAs), network access devices, Internet peripherals, and the like. Such terminals  102   b  generally include a user interface and a LAN interface. 
   The user interface of terminals  102   b  is typically referred to as terminal equipment. The user interface generally includes an audio interface, such as a microphone and speaker, a visual interface, such as a display or graphic user interface (GUI), and a user input interface, such as a keyboard, touch pad, keypad, touch screen, track-ball system, voice recognition system, hand writing recognition system, or combinations thereof. The LAN interface couples terminals  102   b  to the system  200  via a communications protocol for transmitting and receiving data, thereby permitting terminals  102   b  and the access point  206  to communicate. While the suggested LAN is one that may be defined by the IEEE 802.11 standard, it will be appreciated by those skilled in the art that several other wired and/or wireless LAN protocols are known in the art and may be substituted therefore, without departing from the spirit of the present invention. 
   During operation, communications that are directed to and received from a user terminal  102   b  via access point  206  are coupled to the Internet protocol (IP) network  146  via well-known data links (not shown). Such data links implement packet-based protocols providing access to any elements connected to IP network  146 , such as, for example, the other user terminals  102   a  and  102   b , or telephone  144  through PSTN  142 . 
   With further reference to  FIG. 2 , a high level structure of IP transmission packets  280  for use within the Internet protocol (IP) network  146  are shown. Since details regarding IP transmission packets  280  are well within the knowledge of those skilled in the art, no further description will be provided at this time. 
     FIG. 3  is a block diagram of a user terminal  102   a  or  102   b  of  FIG. 2 . As will be appreciated by those skilled in the art, the user terminal of  FIG. 2  is capable of receiving a plurality of packet data streams ( 320 – 324 ) comprising audio, data, video, or combinations thereof. Each packet data stream ( 320 – 324 ) is presented to respective input ports of a selective digitizer  306 . In accordance with the preferred embodiment, the selective digitizer  306  is a digital-to-analog (D/A) converter. In an alternative embodiment, the selective digitizer  306  may comprise any of the available encoder, vocoder, or transcoder techniques known in the art, used alone or in combination with a D/A converter. As shown, the input ports of selective digitizer  306  are individually controlled and enabled/disabled by call control module  308 . The selective digitizer  306  operates to convert an incoming packet data stream into separate analog representations ( 321 – 325 ). The analog representations ( 321 – 325 ) may then be communicated to respective input ports associated with selective mixer  304 . 
   Similar to selective digitizer  306 , the selective mixer  304  input ports are individually controlled and enabled/disabled by call control module  308 . In accordance with the present invention, call control module  308  is a processing device such as a central processing unit (CPU), digital signal processor (DSP), or an equivalent application specific processing unit (ASPU), with or without a separate memory storage device. The enable/disable operations controlled by call control module  308  may employ any of the well know memory device access or bus addressing techniques available in the art. Assuming a DSP-based call control module  308  implementation, it will be appreciated by those skilled in the art that several of the functions described in association with  FIG. 3  may, in fact, be performed by call control module  308 . By way of example, and not by way of limitation, the function of selective digitizer  306  may be performed by an appropriately programmed DSP, without departing from the spirit of the present invention. 
   As previously discussed, each user terminal  102   a  and  102   b  has an input device, such as a keyboard, touch pad, keypad, touch screen, track-ball system, voice recognition system, hand writing recognition system, or some combinations thereof. The input device (not shown) is coupled to call control module  308 , enabling the user to make call service elections, such as, for example call waiting service or conference call service. In response to user election and under direction of call control module  308 , selective mixer  304  sums, mixes, blends, synthesizes, combines, or otherwise manipulates the analog representations ( 321 – 325 ) to provide either a mixed or a non-mixed output  326  to speaker  302 . In accordance with the preferred embodiment, a non-mixed output  326  from selective mixer  304  is synonymous with the provision of a static call or call waiting service. A mixed, summed, blended, or otherwise composite output  326  from selective mixer  304  is synonymous with the provision of call conferencing service. That is, the composite output from selective mixer  304  represents a conference. Selective mixer  304  is preferably implemented by an analog mixer in which the set of inputs are controlled by call control module  308 . 
   During operation, the terminal user uses the audio interface consisting of speaker  302  and microphone (MIC)  312  to communicate with a party or parties of interest. With reference to  FIG. 3 , analog voice  328  from MIC  312  and feed back  326  from selective mixer  304  are provided to a mixer digitizer consisting of a mixer  314  and an analog-to-digital (A/D) converter  316 . The analog inputs  326  and  328  are then summed, mixed, blended, synthesized, or otherwise combined to produce a composite representation of the original inputs. This mixed or composite signal is delivered to A/D converter stage  316 . A/D converter  316  converts the mixed or otherwise composite signal to a packetized data stream. A/D converter  316  may include an encoder, vocoder or transcoder. The mixer  314  is preferably implemented by an analog mixer like those known in the art. 
   From A/D converter  316 , the packet data stream is provided to a multiplexer circuit (MUX)  318 . Under direction of call control module  308 , MUX  318  distributes the packet data stream  330  to call sessions of interest. The distribution operation performed by MUX  318  may employ any of the well known memory device access or bus addressing techniques available in the art. 
   MUX  318  distributes the packet data stream  330  to a single call session for purposes of static call mode and call waiting mode services. Conversely, MUX  318  communicates the packet data stream  330  to a plurality of call sessions for purposes of establishing and maintaining a conference call. Of note, the static call mode is distinguished from the multi-party call waiting mode, in that the static mode is characterized by a single in-bound call session. 
   Based upon the prior discussion and with reference to  FIG. 3 , it will be appreciated by those skilled in the art that the terminal  102  is shown engaged in the call waiting service mode. By way of example, and not by way of limitation, terminal  102  is in receipt of a plurality of in-bound call sessions. Notwithstanding, the selective digitizer  306  and selective mixer  304  inputs associated with analog representations  321  and  323  have been disabled by call control module  308 . By disabling the selective digitizer  306  and selective mixer  304  inputs associated with analog representations  321  and  323 , terminal  102  does not decode voice stream data from devices associated with call sessions  1  and  2 . With respect to call session  3 , it will be appreciated that selective mixer  304  receives the analog representation  325 , which corresponds to voice data stream  324  and call session  3 . As such, call session  3  is serviced by terminal  102  and the selective mixer output  326  comprises a non-mixed signal. Under direction from call control module  308 , MUX  318  communicates packet data stream  330  to call session  3  only. As such, call session  3  will be serviced by terminal  102 , while call sessions  1  and  2  are, in effect, on-hold. 
     FIG. 4  is another block diagram of the user terminal  102   a  and  102   b  of  FIG. 2 . The user terminal  102  of  FIG. 4  is identical to the user terminal  102  depicted in  FIG. 3 , except the user terminal  102  of  FIG. 4  is shown operating in the conference call or multi-party call service mode. By way of example, and not by way of limitation, the user terminal  102  of  FIG. 4  is in receipt of a plurality of in-bound call sessions. Notwithstanding, the selective mixer  304  inputs associated with analog representations  321 ,  323 , and  325  are enabled by call control module  308 . By enabling the selective mixer  304  inputs associated with analog representations  321 ,  323 , and  325 , terminal  102  receives voice stream data from the devices associated with call sessions  1 ,  2  and  3 . As such, call sessions  1 ,  2 , and  3  are serviced by terminal  102  and the selective mixer output  326  comprises a mixed or composite signal. Under direction from call control module  308 , MUX  318  communicates the packet data stream  330  to call sessions  1 ,  2 , and  3 , i.e., the call sessions of interest. As such, call sessions  1 ,  2 , and  3  are serviced by terminal  102  and a conference call is established and maintained. 
     FIG. 5  is a flow chart illustrating a method for establishing call waiting service in accordance with the present invention.  FIG. 5  is described herein with reference to the device shown in  FIG. 3 . It will be appreciated by those skilled in the art that the routine of  FIG. 5  is employed by user terminal  102  of  FIG. 3  when establishing call waiting service in accordance with the present invention. In accordance, the steps described in association with  FIG. 5  are those performed by a device, or under the control of a device, such as call control module  308 , which, in accordance with the preferred embodiment, is a central processing unit (CPU), digital signal processor (DSP), or equivalent application specific processing unit (ASPU) with or without accompanying memory. 
   Commencing at step  500 , flow proceeds to step  502  where a determination is made whether there is an active call session in progress. Assuming not, flow continues to monitor step  502 , until such time as an active call session is detected. Of note, it matters not whether the call is originated or terminated at user terminal  102 . 
   From step  502 , flow proceeds to step  504  where the user terminal awaits a SIP (Session Initiation Protocol) INVITE message, such as, for example, the messaging defined by the Internet Engineering Task Force—IETF—RFC 2543, indicating that an incoming call from a terminating call session is attempting to contact terminal  102 . From step  504 , flow proceeds to step  506  where a determination is made whether the user elects to accept or reject the call for call waiting purposes. Assuming the user elects to refrain from initiating call waiting service, flow branches back to step  504  to await receipt of another SIP INVITE message. Otherwise, the user utilizes an input device like those described herein above to initiate call waiting service. 
   In response to initiation of call waiting service by the user, flow proceeds from step  506  to step  508 , where the call control module  308  of  FIG. 3  disables the selective digitizer  306  and selective mixer  304  input port associated with the in progress call session detected at step  502 . From step  508 , flow proceeds to step  510  where the call control module  308  enables selective digitizer  306  and selective mixer  304  input ports associated with the call session identified by the SIP INVITE message and accepted by the user at step  506 . Collectively, steps  508  and  510  of  FIG. 5  operate to select, from amongst a number of available call sessions, the call session of interest. 
   With reference to  FIG. 3 , upon selection of a call session of interest, the user terminal  102  proceeds to mix at mixer  314  the analog representation  326  of the call session of interest, with the user generated voice from microphone  312  to produce a mixed output that is converted into a packet data stream  330 . 
   Returning to  FIG. 5 , flow proceeds from step  510  to step  512  where the packet data stream, which includes voice or data, is distributed by multiplexer  318  to the call session of interest. From step  512 , flow proceeds to step  514  where a check is made to determine whether the call session of interest has terminated. If not, flow branches back to step  504  where the terminal awaits receipt of an additional SIP INVITE message. Assuming the call session of interest terminates at step  514 , flow proceeds to step  516  where a determination is made whether a call placed on-hold at step  508  is still available. Assuming an on-hold call is available, flow branches back from step  516  to step  510 , which operates to select, from amongst a number of available call sessions, another call session of interest. Otherwise, if all call sessions have ended at step  516  the process terminates. 
     FIG. 6  is a flow chart illustrating a method for establishing conference call services in accordance with the present invention.  FIG. 6  is described below with reference to the device shown in  FIG. 4 . It will be appreciated by those skilled in the art that the routine of  FIG. 6  is employed by user terminal  102  of  FIG. 4  when establishing on-demand conference call and/or multi-party call service. In accordance, the steps described in association with  FIG. 6  are those performed by a device, or under the control of a device, such as call control module  308 , which, in accordance with the preferred embodiment is a central processing unit (CPU), digital signal processor (DSP), or equivalent application specific processing unit (ASPU) with or without accompanying memory. Commencing at step  600 , flow proceeds to step  602  where a determination is made whether there is an active call session in progress. Assuming not, flow continues to monitor step  602 , until such time as an active call session is detected. Of note, it matters not whether the call is originated or terminated at user terminal  102 . 
   From step  602 , flow proceeds to step  604  where the user terminal awaits: 1) receipt of an incoming call session, as indicated, for example, by a SIP (Session Initiation Protocol) INVITE message, of the type defined by Internet Engineering Task Force—IETF—RFC 2543, and indicating that an incoming call for a call session is attempting to contact terminal  102 ; or 2) receipt of an outbound call request, indicating that the terminal user is attempting to make a call, as indicated, for example, by a SIP (Session Initiation Protocol) INVITE message. From step  604 , flow proceeds to step  606  where a determination is made whether the user elects to accept or reject the call for conference call purposes. Assuming the user elects to refrain from initiating conference call services, flow branches back to step  604  to await receipt or initiation of another call. Otherwise, the user utilizes an input device like the ones described herein above to initiate a conference or multi-party call. 
   In response to user election, flow proceeds from step  606  to step  608 , where the call control module  308  of  FIG. 3 , in response to the user selected input, enables the selective digitizer  306  and selective mixer  304  input ports associated with the call session detected at step  602 . From step  608 , flow proceeds to step  610  where the call control module  308  of  FIG. 4  enables selective mixer  304  input ports associated with the call sessions identified by the SIP INVITE message and accepted by the user at step  606 . Collectively, the steps  604 – 608  of  FIG. 6  operate to select, from amongst a plurality of available call sessions, those call sessions of interest to the user. 
   With reference to  FIG. 4 , upon selection of said call sessions of interest, the user terminal  102  proceeds to mix at selective mixer  304  the analog representations  321 ,  323 , and  325  of the call sessions of interest. Thereafter, the mixed output  326  is combined with user-generated voice from microphone  312  to produce another mixed output that is converted into a packet data stream  330  by mixer  314  and A/D converter  316 . 
   Returning to  FIG. 6 , flow proceeds from step  608  to step  610  where the packet data stream, which includes at least one of voice and data packets, is distributed by multiplexer  318  of  FIG. 4  to the call sessions of interest. From step  610 , flow proceeds to step  612  where a check is made to determine whether a conference call session has terminated. If not, flow branches back to step  604  where the terminal awaits receipt or initiation of additional calls. Assuming a conference call session of interest terminates at step  612 , flow proceeds to step  614  where associated selective digitizer  306 , selective mixer  304  and MUX  318  inputs/outputs are disabled to halt distribution of packet data  330  to terminated call sessions at step  612 . Call termination ends the flow at step  616 . 
   Advantageously, the invention described herein allows a party to elect call-waiting and/or conference call services in a very timely and cost efficient manner. Based upon this arrangement, the user elects on-demand call waiting and call conferencing services. Unlike the prior art, the user is permitted to establish call conference services for originating and terminating calls, alike. Moreover, the call conference service described herein is established without the coordination of, and use of, substantial network resources. Of additional importance, call waiting as described herein is a multi-party call service, permitting multiple calls to be placed on-hold. 
   Whereas the present invention has been described with respect to specific embodiments thereof, it will be understood that various changes and modifications will be suggested to one skilled in the art and it is intended that the invention encompass such changes and modifications as fall within the scope of the appended claims.

Technology Category: h