System and method of expediting call establishment in mobile communications

System and method for group call in mobile communication

Technical field

The present invention relates to mobile communication, in particular to a system and method for group calling in mobile communication.

Background technique

All modern mobile communication systems have a telephone hierarchy (hierarchical arrangement) in which the geographical "coverage area" is divided into a large number of smaller geographical areas called "cells". With reference to Figure 1, each cell is preferably served by a base transceiver station ("BTS") 102a. Several BTSs 102b-n are aggregated into a base station controller ("BSC") 106a via fixed links 104a-n. The BTS and BSC are sometimes collectively referred to as the base station subsystem ("BS") 107. Several BSCs 106a-n may be aggregated to a mobile switching center ("MSC") 110 via fixed links 108a-n.

The MSC 110 acts as a local switching switch (with additional functions for controlling mobility management and requirements, discussed later) and communicates with the telephone network ("PSTN") 120 through a trunk group. There are concepts of home MSC and serving MSC in American mobile networks. The local MSC is the MSC corresponding to the exchange associated with the mobile station ("MS"); the association is based on the phone number of the MS, such as the area code. (The local MSC is responsible for the HLR, which will be discussed below.) On the other hand, the serving MSC is used to connect MS calls to the switch on the PSTN (when the user is roaming in the area covered by the service provider, different MSCs will complete the service MSC function). Therefore, sometimes the local MSC and the serving MSC are the same entity, but other times they are no longer the same entity (for example, when the MS is roaming). Usually the visitor location register ("VLR") 116 is co-located with the MSC 110, and a logically separate HLR is used in the mobile network. As will be explained below, both HLR and VLR are used to store many types of user information and profiles.

Simply put, one or more wireless channels 112 are associated with the entire coverage area. These wireless channels are divided into channel groups allocated to individual cells. These channels are used to transmit signaling information to establish a call connection, etc., and to transmit voice or data after the call connection is established.

At a relatively high level of abstraction, mobile network signaling includes at least two main aspects. One aspect includes the signaling between the MS and the network area part. In 2G ("2G" is an industry term for "second generation") and later technologies, this signaling relates to the access method used by the MS (for example: Time Division Multiple Access or TDMA; Code Division Multiple Access or CDMA) , Allocation and identification of wireless channels, etc. The second aspect includes signaling in different entities in the mobile network, such as signaling in MSC, VLR, and HLR. This second part is sometimes referred to as the mobile application part ("MAP"), especially when used in the environment of the No. 7 signaling system.

Different forms of signaling (as well as data and voice communications) are sent and received according to different standards. For example, the Electronics Industry Association ("EIA") and the Telecommunications Industry Association ("TIA") helped define many American standards, such as IS-41 of the MAP standard. Similarly, CCITT and ITU helped define many international standards, such as the international MAP standard GSM-MAP. Information about these standards is well known and can be found in related organizations and works (see, for example, Bosse, Signaling in Telecommunications Networks (Wiley 1998)).

To provide a call from MS 114, the user dials the number and presses "send" on the cell phone or other MS. The MS 114 sends the dialing number indicating the service request to the MSC 110 through the BS 107. The MSC 110 checks the relevant VLR 116 (detailed below) to determine whether the MS 114 is allowed to perform the requested service. The serving MSC sends the call to the local exchange of the dialed user on PSTN 120. The local exchange prompts the called user terminal, and sends a reply signal back to the MS 114 through the serving MSC 110, and the MSC 110 then completes the voice path to the MS. Once you have completed this setting, you can make a call.

In order to provide a call to MS 114, (assuming the call is initiated from PSTN 120) the PSTN user dials the relevant phone number of the MS. According to at least US standards, the PSTN 120 sends the call to the MS's local MSC (which may or may not be an MSC serving the MS). The MSC then queries the HLR 118 to determine which MSC is currently serving the MS. This will also operate to inform the serving MSC that the call is coming. The local MSC then sends the call to the serving MSC. The serving MSC calls the MS through the appropriate BS. After the MS responds, a proper signaling link is established.

During the call, the BS 107 and the MS 114 can coordinate to change the channel or the BTS 102 if necessary, for example because of signal conditions. These changes are called "handoffs" and they include known messages and signaling of their respective types.

One aspect of MAP includes "mobility management." Simply put, when the MS 114 roams to different locations, different BSs and MSCs may be needed and used to serve the MS. Mobility management ensures that the serving MSC has a brief profile of the user and other information that the MSC needs to properly service and pay for calls. To this end, the MSC uses a visitor location register ("VLR") 116 and a home location register ("HLR") 118. Use HLR to store and retrieve the mobile identification number ("MIN"), electronic serial number ("ESN"), MS status and brief information of MS services. The VLR stores similar information in addition to the MSC identification that identifies the (local) MSC. In addition, under a suitable MAP protocol, a location update process (or registration notification) will be performed to make the mobile user's local MSC know the location of the user. These steps are used when the MS roams from one location to another or when the MS powers up and registers itself to access the network. For example, the MS 114 may send a location update request to the VLR 116 through the BS 107 and the MSC 110 to perform the location update process. The VLR 116 sends a location update message to the HLR 118 serving the MS 114, and the user profile will be downloaded from the HLR 118 to the VLR 116. MS 114 sends a confirmation that the location update was successfully performed. HLR 118 requests that the VLR (if any) with profile data is reserved in advance to delete the data of the MS 114 involved in the relocation.

Figure 2 shows in detail the signaling and user communication interface between the BS 107 and the MSC 110 in the CDMA mobile network. The BS 107 uses the A1 interface to transmit signaling information. The A2 interface supports the user's communication between the switching component 204 of the MSC and the BS 107. The A5 interface is used to provide a path for user communication to conduct a circuit-switched data call between the source BS and the MSC (different from a voice call).

In addition, users demand newer services, such as "data calls" to the Internet. For some of these services, MSC will not save costs, because they are mainly designated for voice calls. Due to the proprietary and closed design used by many MSC software structures, it is very complicated or impossible to integrate new services into MSC. In other words, the software logic that needs to provide services cannot be simply added to the MSC 110. Often use switching accessories to provide such services. For example, the interworking function ("IWF") is an attachment that sends data calls to the Internet. The method of integrating functions into the MSC or adding relay-end accessories will involve the MSC providing the service. Since new services are expected to stimulate requirements, the integration of new services through MSC design changes or through relay terminal attachments is likely to aggravate network congestion on the MSC and require expensive MSC resources.

For the Internet, multicast communication refers to the sending of the same data packet to a selected multi-destination address in the Internet Protocol network. (On the contrary, broadcast communication refers to the indiscriminate sending of data packets to all destination addresses, and unicast communication refers to the sending of data packets to a single destination address).

Each participant in the multicast can receive the information sent by any other participant in the multicast. Users who are connected to the network and are not participants in a particular multicast will not receive messages sent by the multicast participants. In this way, multicast communication uses only the network components (such as switches and trunks) that are actually required for multicast.

In multicast processing, when a potential participant ("host") joins a specific IP multicast group, the host will send a "request to join" message to the nearest multicast router to request to join the multicast group And receive the information sent to the group. For example, host A sends a message to join multicast group Y, and host B sends a message to join multicast group X. If the data path is not yet in place, router R will propagate the request up to the multicast source.

After receiving the IP packet of group X, for example, router R will map the IP multicast group address to an Ethernet multicast address, and send the generated Ethernet packet to the appropriate one or more switches.

According to the current Internet Group Management Protocol (Internet Group Management Protocol, "IGMP"), the membership of a host in a multicast group will be terminated when the router does not receive a periodic membership report from the host.

For the interaction in MS, two versions of Nextel services have been proposed (the so-called Nextel Direct Connect®, which uses special mobile wireless technology, at http://www.nextel.com/phone services/directconnect. shtml has description) is used for special connection calls in MS. Both versions of the special connection call require all members to be in the same area served by a BSC. In the first version, one-to-one conversations between two mobile phone users (for example, A and B) are allowed. When A wants to have a special connection communication with B, A enters B's private identification number, presses the push-to-talk ("PTT") button, waits for an audible reminder that B is ready to receive, and starts speaking. You can release the PTT button while listening. If B wants to speak, B will press the PTT button and wait for an audible confirmation that A is ready to receive. The service will allow users to select a personal identification number from a scrollable list displayed on a mobile phone or search a list of pre-stored usernames.

In the second version, a conversation is allowed among members of a predetermined user group (called a talk group), and the members of the user group can be identified by a number. Mobile phone handsets allow searching for group numbers through the control surface of the phone. To arrange a group call, the user (for example, A) locates the group number in the phone, presses the PTT button, and can start speaking after receiving an audible confirmation (such as a chirp). All other group members on the group call can listen only when A presses the PTT button. If A releases the PTT button, another member on the group call may press the PTT button, gain control notified by the audible confirmation, and start speaking.

Summary of the invention

The present invention generally provides a system and method for mobile communication, and more specifically, provides a system and method for group calling. Retrieve information from the member list of the group call group. Based on the retrieved information, a group call is established between the first and second mobile station (MS). The first base station controller (BSC) serves the first BS, and the second BSC serves the second BS. Send the voice data of the group call in the multicast session. Based on the history of the group call between two points in the mobile communication network, a decision is made whether to establish a multicast session between the two points in the expected future group call, for example.

By initiating a single call, the members of the group can establish a group call among all the group members that can be contacted. Group calls can be established among members located in different areas served by different BSCs and perhaps different access methods (such as TDMA or CDMA). The interactive BSC voice communication among members in the group call may be transmitted by another communication network, such as an Internet Protocol network.

Description of the drawings

Fig. 1 is a system diagram of a prior art mobile network; Fig. 2 illustrates the prior art interface between a BS and a mobile switching center in a prior art mobile network; Fig. 3 illustrates a group call logic including The block diagram of the system; Figure 4-5 illustrates the proxy switch in the mobile network and some configurations; Figure 6 illustrates an example data plane of the proxy switch according to the preferred embodiment of the present invention; Figures 7, 9, 16- 17 illustrates the structure of the group communication system; FIGS. 8A-8C and 11-15 are the call flow diagrams using the group communication system; and FIG. 10 illustrates the flow chart of the group call logic.

detailed description

Referring to Figure 3, a system and method for establishing a call among members of a predetermined mobile phone user group is provided. As detailed below, the proxy switch or other device that implements the group call logic 1010 detects the group call initiated by the member 1012A of the group 1014, and automatically tries to connect all the members 1012A, 1012B, and 1012A of the group in the group call. 1012C. In a specific implementation, the communication in the group call is half-duplex (for example, only one member can speak at a time), and the voice communication used for the group is based on the Internet Protocol in the multicast session (" IP) network.

For the case where the group call logic is implemented by the proxy switch, the proxy switch can, as submitted on November 22, 2000, named System and Method of Servicing Mobile Communications with a Proxy Switch. Operate as described in pending U.S. Patent Application 09/721329, which is hereby incorporated by reference. As described in this co-pending application and shown in FIG. 4, the handover 1034 operation is performed between at least one mobile switching center ("MSC") 1030 and at least one base station subsystem ("BS") 1032. The handover causes the traffic to be transferred out or transferred into another network 1036 (such as an IP network). The handover is transparent, so neither the MSC nor the BS can work with the handover of the present invention without any changes.

The proxy switch described in the co-pending application includes signaling message control logic 1038 to receive signaling messages from the MSC and BS according to the mobile signaling protocol. The message interception logic 1040 works in conjunction with the signaling message control logic, and sends a confirmation message to the MSC or BS that sends the signaling message. The message interception logic also prevents signaling messages from being forwarded to the other of the BS and MSC respectively. The message conversion logic 1042 works in cooperation with the signaling message control logic, and converts a signaling message from one of the MSC and the BS into a converted signaling message for transmission to the other of the BS and the MSC, respectively. The message sending logic 1044 works in conjunction with the signaling message control logic, and sends the signaling message from one of the MSC and BS to the other of the BS and MSC, respectively.

Assign a set of bearer circuits 1046 from the BS to the proxy switch. Receive and analyze the signaling messages between the MSC and the BS to determine whether they correspond to the assigned set of bearer circuits. If corresponding, the control information in the signaling message will be transmitted to another communication network; the information carried by this group of bearer circuits will be transferred to another network.

FIG. 5 shows a preferred configuration of the proxy switch 300, where the proxy switch 300 is located between the BS 107 and the MSC 110. Only a subset 306 of the trunk lines that carry user communications need to be terminated on the proxy switch; other trunk lines 308 can directly connect the MSC 110 and the BS 107. All control links 312 from the BS 107 are terminated on the proxy switch 300. The proxy switch includes a control plane 302 and a data plane 304 (also referred to as a "load plane"). The control plane 302 controls all signaling communications, and the data plane 304 controls all user communications on the trunk lines connected to the proxy switch.

In some embodiments, there is one-to-one communication between the MSC and the proxy switch. Several BSs can operate together with separate proxy switches.

The proxy switch 300 includes software that accepts all signaling messages, and performs at least one of the following operations according to the message and system status: 1. The message is delivered unchanged to the MSC or BS addressed in the message; 2. Truncate the message between the MSC and the BS; 3. For some truncated messages, convert the truncated message into a different message and send the transformed message instead of the original truncated message to the MSC or BS addressed in the truncated message ; 4. Transfer messages from mobile and PSTN-based networks to other networks such as IP networks.

The following will describe these types of operations in each case that triggers the event.

In many instances, the proxy switch 300 can act as the MSC 110, especially when the message from the MS 114 is tuned and the communication is sent to another network. In such a role, the proxy switch fulfills the responsibilities and tasks that the traditional MSC will complete. Some of these functions and tasks belong to mobility management. Consider the case of a roaming MS. When it roams from one cell to another, it may roam to a cell served by a different MSC, so it needs to be transferred between the source and target MSC. If the proxy switch 300 transfers the message and the call/session is transferred to another network, then the proxy switch will manage the transfer, just like a conventional MSC manages the transfer. The agent switch prompts the appropriate database to update the new location of the MS.

Another function of the proxy switch belongs to resource allocation. Especially when the MS initiates a message to request a new call/session, it needs to allocate a suitable circuit (channel) for the session. According to the configuration and system status of the system, the proxy switch performs the distribution in the same way as the conventional MSC distributes the circuit.

Figure 6 shows an example configuration in which the proxy switch 300 is connected to multiple additional networks, such as an IP backbone 412 or another circuit-based network 414 (e.g., a different carrier). These additional networks can be used to deliver voice and/or data communications to the desired destination address, while completely or partially avoiding the use of the PSTN 120 that accompanies the expensive resources of the MSC 110. In addition, such a structure can also be used so that circuit communications can be sent back to different networks; for example, circuit communications from Nashua NH in New Hampshire can be sent back to the MSC in Waltham, Massachusetts. Or, you can use them to connect to other networks. For example, the IP backbone 412 may communicate with the IP voice network 418 or the Internet 416. As explained in the joint pending patent, when the communication is tuned to another network, the control information on the link 306 (for example, from signaling messages) and the voice or data from the bearer circuit can be sent through the other network .

In the specific implementation of the group communication system described above, mobile communication users ("users") belonging to a closed user group are provided with the ability to quickly and conveniently communicate with each other, thereby starting mutual conversations. Each group includes two or more users ("members"), and users can belong to multiple CUGs. The conversation can take place between two members of the group ("private mode") or between all existing members of the CUG ("public mode"). The group communication system uses conventional mobile communication devices such as cellular phones and mobile PDAs.

In a specific implementation, as described above, the group communication system is logically distributed in the proxy switch between the MSC and the BSC to implement the group call logic, cut off the group call start, bypass the MSC and PSTN, and implement the group call as a complete Voice over IP ("VoIP") IP multicast sessions. Users in multiple MSC service groups can be served in completely different geographic locations, where the multiple MSCs span the aggregated network, thereby eliminating the need for wireless technologies such as CDMA, TDMA (including IS-136 and GSM), GPRS, and third-party services. Generation technology) one or more networks. For example, among the group members joined on any group call, one or more users may be roaming in the GSM network, and at the same time, one or more users are roaming in the CDMA network. During the group call, the control information belonging to the group call can be made effective for one or more users (such as display participants in the group call). Group call users can use standard numbering schemes (such as MIN, IMSI, and ESN) to dynamically create and modify group call lists.

The general structure of an exemplary embodiment of the group communication system is shown as an example in FIG. 7. Figure 7 shows four users in a group call. The users use wireless devices 1060A-1060D connected to different BTS systems 1062A-1062D. For the convenience of the following description, it is assumed that the wireless device has audio and text display functions. The BTS is connected to a base station controller ("BSC") 1064A-1064D, and the BSC is connected to a proxy switch ("group call switch") 1066A-1066C that implements group call logic. Each group call switch is connected to an MSC such as MSC 1068A, MSC 1068B, or MSC 1068C. Each MSC in the group call service enabling network is provided with at least one group call switch. As for the signaling information, each group call exchange is logically located between the corresponding BSC and the corresponding MSC. The group call switch receives signaling and data from the wireless device through the BTS and BSC from the MSC and in the opposite direction. The operation of each group call switch makes neither the BSC nor the MSC aware of the group call located between the BSC and the MSC. The group call switch intercepts the signaling and control information from the MSC and BSC, and seamlessly transfers the signaling and control information to related components without any discernible changes.

The MSC is connected to the public land mobile network ("PLMN") 1070, and the group call switch is connected to the backbone multicast-enabled IP network ("backbone network") 1072. The backbone network provides support for the CUG Active Directory 1074 and enhanced local Access to Location Register ("HLR") 1076.

As mentioned above, for the proxy switch of the joint pending application, the group call switch includes a control plane and a data plane. The function on the control plane is to terminate signaling messages from the BSC and/or MSC. For example, in a CDMA network, signaling messages are defined by IS-634 protocol specifications. The control plane terminates the incoming signal and generates a new signaling message to be forwarded to the MSC or other components. The control plane also supports multicast functions as described below.

In a specific embodiment, the data plane of the group call switch receives TDM communications from the BSC and/or MSC, and uses TDM cross-connect ("DACS") to connect incoming communications to outgoing destination addresses. In other embodiments, the data plane may also receive incoming IP communications from a base station complex (also referred to as a radio access network or "RAN"), and switch the incoming IP communications to outgoing IP communications. The programming control in the control plane determines the cross-connection between the incoming TDM communication and the outgoing destination address (especially the conventional MSC and/or destination address on the IP network).

In the example where the MSC serves as the outgoing destination address from the DACS, the group call switch is basically transparent to the network; communication and control flow seamlessly from the BSC to the MSC and from the MSC to the BSC. When the outgoing destination address is replaced with the IP network, the media gateway in the data plane (described in the joint pending application) transfers the selected part of the incoming TDM communication from the MSC, and converts the incoming TDM communication to RTP/UD /IP communication, insert RTP/UD/IP communication into the backbone IP network.

CUG Active Directory ("CUG AD") 1074, also known as Group Call Registration ("GCR"), is a database system containing CUG data. In a specific implementation, CUGAD in Figure 15 is implemented as a measurable distributed database system. CUG AD contains the definition of all CUGs in the group call network. The query to the CUG AD specifies the identifier of the CUG, that is, if the query specifies the definition of the CUG, the returned result will be a list of group user IDs of the specified CUG members. For example, a query specifying CUG ID 2347 may cause CUG AD to produce results that identify the mobile identification numbers ("MIN") xxx, yyy, zzz, and www of the four users in the CUG. In a specific implementation, the service provider assigns the MIN number to the user of the GIR service.

Each CUG is identified to the system by a unique identifier ID. The unique ID comes from the CUG namespace. Dividing the namespace allows different parts to be allocated to different distributed parts of the CUG AD. Make the partition index of the partition plan effective for all group call exchanges. When the group call switch needs to retrieve the definition of the CUG, the group call switch can use the index to determine the component of the CUG AD that it wants to query.

Enhanced HLR 1076 is an enhanced version of the standard HLR database for cellular phones. The standard HLR can perform location updates from roaming mobile users. The conventional path that these updates traverse is from the mobile phone to the BTS, from the BTS to the BSC, and then to the MSC, where the update message is sent to the HLR. In the specific implementation of the group call network, the group call switch is located between the BSC and the MSC, and the MSC makes the group call switch aware of all location updates. For users who subscribe to the group call service, the group call exchange intercepts location update messages and copies them to the HLR'. In addition to storing the cell location of the group call user, the HLR' also stores a list of all CUGs to which each group call user belongs. The query to HLR' specifies the MIN that identifies the group call user, and generates a response: including a CUG list of which the group call user is a member.

In a specific implementation, HLR' is a distributed database, where the distribution of data is based on the MIN level. In addition, HLR' can also be based on International Mobile Subscriber ID ("IMSI") or Equipment Serial Number ("ESN"). An index similar to the above-mentioned partition index for the CUG name space allows the group call exchange to determine the HLR' partition to be queried when the incoming request is about to be processed.

In the following examples of this series, CUG can be defined as having a user's father, mother, and juvenile, each of whom carries a cellular phone. By pressing a special key sequence on his cell phone (or if a special key is provided on the phone, press a special key), the father can perform the initial operation of the group call, which will locate the mother and the teenager, and invite the mother and the teenager Join the group call. In the following half-duplex implementation, when the group communication system confirms that at least one CUG member has joined the group call, the father, as the member who initiated the join, has the speech control right, and when one or more of the other joined members answer the call Start talking. When the father gives up speech control, any other joined CUG member (for example, if the juvenile joins the call, it can be the juvenile) can obtain speech control and speak. Therefore, in the half-duplex implementation, only the joining member assigned the speech control right can speak; the other joining member cannot speak until the speech control right is withdrawn and reassigned. As detailed below, in order to request speech control rights, the user on the group call can press the standard number key on the phone, if the phone manufacturer provides a special key, or press the special key, or if the phone has a text message function If you want, you can send a text message to the current speaker to request the next talk. If at a certain point in the call process, the speech control remains unallocated for a period of time (that is, the call ends), the call will be terminated.

When all members of the CUG are in the same switching area (that is, in a geographical area controlled by a separate MSC or proxy switch) or in different switching areas, group calls can be arranged. For example, if the CUG has users A, B, C, and D, then A and B may currently be roaming in exchange area S1, C in exchange area S2, and D in exchange area S3. If the S1, S2, S3, and S4 exchange areas are all operated by the same service provider, or managed by operators who have agreed to cooperate with each other to provide group call services, connections A, B, C, and D can be established Group call. If this is the case, individuals can be scattered across a wide range of physical locations, for example, A and B can be in Boston, C can be in Texas, and D can be in California.

The group communication system provides interoperability of calls in different network technologies, that is, whether or not CUG members roam on networks based on different technologies, group calls are allowed to be established. For example, user A can roam in a code division multiple access ("CDMA")-based network in Boston, while user B can roam in a global digital mobile phone system ("GSM")-based network (GSM) in the United Kingdom ("UK"). Use time division multiple access ("TDMA") technology). If A and B belong to the same CUG and CDMA and GSM operators have agreed to cooperate in providing group call services, a group call can be established between A and B.

The system can include one or more enhancements that will be detailed below. It is obvious from the above description that a group call can be established regardless of whether all CUG members have joined the call. If some members of the CUG cannot join the group call, the system will generate and record an exception list that lists the missing members. If one or more CUG members have a phone with a display, the exception list can be displayed on the display. Further actions can be taken based on the exception list. For example, during or after the call, a voice mail message can be sent to the members listed in the exception list; for example, the voice mail message can be recorded once by the call originator and can be provided to the previously designated voice mail through the group communication system The voice mailbox of the member in the exception list of the phone number. In addition, CUG members who have joined a specific group call can be listed on the available display screens of all joined members' phones during the call, so that each joined member knows which other members are on the call. Visual indications are provided on the available display screens to identify members who currently have the right to speak.

One user of the group communication system and another user of the group communication system can establish private calls, so that users can discuss confidential or personal information outside of the group call. Therefore, the two participants in the ongoing group call can temporarily make a private call and then return to the ongoing group call.

A user of a group communication system may request a list of CUGs in which the group call is in progress ("active group call"). The user is a member of the CUG and joins such a call. Users of the group communication system can initiate or join a public group call, that is, a group call that includes the CUG of each user of the group communication system. The operator can define any number of public user groups ("PUG"). Make each user of the group communication system automatically become a member of all PUGs. To join the active public group call, the user requests a list of active public group calls and selects one of the calls to participate.

If the user of the group communication system has a call waiting function, the user can interrupt the group call briefly and answer the incoming call signal. Users who do not want to receive any incoming call signals (including group calls) can initiate call forwarding or call blocking (blocking). The user can choose to block only incoming call signals for group calls and/or private calls.

The group communication system can combine speech to text conversion in group calls, which is beneficial to users in noisy environments, users in public places where the use of voice calls is restricted, and users with hearing loss, and can be combined with language conversion (for example, from English to French) .

Group communication system users can contact by using the users mobile phone number or a special group communication system identification number ("group user ID"). The group user ID can be based on the service providers storage in the users phone during the group service contract process. The contact information in the directory is assigned. In addition, the group user ID can also be self-established ("self-provided") using a web-based provisioning system, as described below.

As mentioned above, the group communication system provides three modes of operation: closed user group ("CUG") mode, private mode, and public user group ("PUG") mode. Except in the case of user-controlled calls as described below, the user presses the number key (or presses a special key if a special phone key is provided), and waits for a voice indicating that the speech control has been transferred to the user. When the user finishes speaking, he can press a button, which will make other joining users hear the voice indicating that the speaking control right is available. Another user who joins can then press a button, hear the sound, and start speaking. After at least one other user joins the call as indicated by the voice, the initiator starts speaking control. When all joined users have been suspended or no one has requested speech control for a period of time, the call is ended, as mentioned above. The system resolves any conflicts between simultaneous requests for speech control from users. For example, if full-duplex mode is available for the conversation, the "human protocol" can be used. In this case, the control right can be transferred to multiple users, and finally all users except one user are silent, and the speech control right is transferred to the non-silent user.

In the CUG mode, group call users create a unique group ID list and assign members to the list using the member group call ID and their mobile phone numbers to form a closed user group. Since the radio access network ("RAN") signals the mobile phone under the guidance of the proxy switch, and because the RAN uses the mobile phone number to complete such signaling, the CUG includes the mobile phone number. Each CUG arbitrarily chooses to include two or more users according to the maximum number implemented by the service provider.

In a specific implementation, if the group call user wants to contact the user group of the user's group (ie CUG mode), the user can type ("input") the call initiation sequence, such as *4 followed by the CUG ID, and then press the send button . (The call initiation sequence can also be stored in the speed dial directory of the user's phone and dialed from it.) CUG members who cannot join the call at the first notification can (if the call is still active) by entering the call initiation sequence and pressing the send button And later joined the call.

In CUG mode, the call initiator can choose to request user-controlled calls, which is a call with "barge-in" capability. According to this barge-in capability, the listening user can be configured by pressing the service DTMF key sequence to send prompt messages to the speaking user to indicate to the speaking user that the listening user wants to control the speech. At this time, the speaking user can press the button to give up the speaking control right, or can continue speaking. Therefore, the interrupt capability provides the speaking user with an audible notification that the listening user wants to speak. If the phone has a text display screen, it can also provide a text message showing the name of the listening user who sent the inserted message. Therefore, the speaking user does not have to guess whether the listening user wants to speak. When the speaking user does not want to be disturbed (for example, in the process of preaching to a large group of people), this insert function can be disabled.

As mentioned above, the CUG mode can also provide attendance reports through the speaker identification function, and provide attendance exceptions through the broadcast function.

The call transcription function can be provided in the CUG mode, so that the voice-to-text conversion technology can be used to transcribe group calls in real time. Group members who have a text display phone and notify the group call can request to mute the call and receive text transcription instead. In a specific embodiment, non-text display phones only send and receive audio, while text-only devices (such as text pagers) are used to receive text. If the call transcription function is not turned on, text-only devices cannot receive group calls. Notice. At the end of the call, a full transcription of the call can be available and can be sent to all members of the CUG, CUG members who have not joined the call (according to the attendance exception list), or the call originator. This function can be extended to a translation service, using preferred language indicators available according to IS-41-C. The preferred language indicator is an information unit contained in the user profile and stored in the HLR database, used to indicate the user's language preference, in which announcements or other reports should be presented. The switch uses the preferred language indicator when pre-stored announcements. Service providers can provide additional resources such as human or automatic translators.

A web-based provisioning application from a personal computer or from a WAP-enabled device can be used to build a group call ID and related member list. The user can also build a list from the user's mobile phone. The service provider specifies limits on the number of members on the list, one or more networks to which the members belong, and the number of lists that the group call user can maintain.

In the dedicated mode, by entering the group call initiation sequence (for example, *4 followed by the group call ID of the member to be received) and pressing the send button, the group call user can quickly call the group call to the group call enabled network Calls from any member of the user group. The member who intends to receive is notified of the call from the user. In a specific embodiment, when the member who intends to receive has answered the call, the user hears the chirp.

In PUG mode, group call users can view the list of current chat groups and decide to join the chat group. By specifying a unique group call ID and providing a brief text description of the group subject, the group call user can create a new chat group that is effective for any group call user. As long as the text name is unique to the currently active chat group, the user can freely associate the text name with the group call ID.

The group call function does not take precedence over the existing mobile phone functions (such as call forwarding, do not disturb and call blocking). If call waiting is activated, the caller on the group call can switch the incoming call. During the group call, for security reasons, three methods are disabled: calling, conference call, and call forwarding. These three methods can be re-enabled at the peak of the group call.

If the corresponding base station supports voice privacy ("VP"), as requested by the mobile phone, the group call service supports session encryption using IS-41-C's VP function.

In the specific implementation described in more detail below, the group call service operates using IP multicast on an IP network. As mentioned above, IP multicast allows a source to send a duplicate stream of individual VoIP packets, to be received by multiple receivers that have explicitly registered to receive the stream. Multicast is a receiver-based concept, so the receiver joins a special multicast session group and provides the stream to all members of the group through the network infrastructure. Only one replicated multicast stream is delivered on any link in the IP network, and if required, the replication is only made at the IP multicast-enabled media gateway.

In a wireless network, the service has certain characteristics of any conventional wireless call. As mentioned above, the group call originator sends the DTMF feature escape sequence and follows the ID of the group call list of the user that the originator wants to contact (for example, "*4" is followed by the user group ID), Thus calling the group call. The characteristic escape sequence is used by the proxy switch to detect whether the call is a group call, and to query the global call registration ("GCR" or "CUG AD") to retrieve the list of mobile phones to be contacted and their current location. The current location information determines which media gateway and BSC will be involved in the group call. Through the data plane of the group call switch, a bearer channel is established between each such BSC and the corresponding media gateway. The group call is presented to each BSC because it has conventional point-to-point call establishment and teardown features, and the group call is not presented to each MSC.

Figures 8A-8C show example call flow diagrams of basic services in a wireless network, where WC-1 and WC-2 represent group call switches. Also in Figure 9-10, the main entities in the flowchart are group call switches WCS-1 and WCS-2, which have separate media gateways MG1 and MG2; GCR reachable by WCS-2; BSC-1, BSC- 2; and mobile stations (such as phones) MS-A, MS-B. In the example, MS-A and MS-B happen to be controlled by the same group call switch WCS-1, and if MS-A and MS-B are controlled by different group call switches, the steps will be the same.

Fig. 10 illustrates a group call logic flow chart that summarizes the call flow chart for establishing a group call of Figs. 10A-10C. The logic part controlled by WCS-1 is shown on the left, and the logic part controlled by WCS-2 is shown on the right. WCS-1 detects that MS-A has requested a group call (step 3010) and therefore notifies WCS-2 (group call coordinator) (step 3020). WCS-2 queries the GCR to determine the other members of the CUG of MS-A and their latest known positions (step 3030). (In this simple example, MS-B represents the only other member.) WCS-2 creates a media gateway connection to WCS-1 (group call switch for MS-A) (step 3040). WCS-1 queries the wireless channel of MS-A (step 3050). WCS-2 tells WCS-1 (group call switch for MS-B) to call MS-B (step 3060). WCS-1 calls MS-B (step 3070), and then informs WCS-2 that MS-B is found (step 3080). WCS-2 creates a media gateway connection to WCS-1 (group call switch for MS-B) (step 3090). WCS-1 queries the wireless channel of MS-B (step 4000) and notifies WCS-2 that the query has been completed (step 4010). WCS-2 tells WCS-1 to go to MS-A to indicate that it has reached MS-B (step 4020), and then play a sound to MS-A through the media gateway (step 4030).

The following will further explain call flow examples in conjunction with Figures 12-15.

As mentioned above, in a group call, only one joining user is allowed to speak at a certain time; joining users with speech control can send a preset DTMF number (such as "1") to give up this control, and then another A joining user can request speech control by sending DTMF digits (such as "8"). Joining users with speech control rights will hear the sound played when the sending path is established, as shown in Figure 11.

Before the group call, the group call user can choose to enable or disable one or more of the following functions as described above: attendance exception report, attendance report, call transcription and barge functions.

In a network-based system that enables end users to establish their GCR lists, the network server is connected to the GCR over the IP link, so that the user group call list can be updated in real time. The establishment system supports WAP protocol and industry standard browsers.

Collect the call records of all participants in the group call for account and network engineering purposes.

Fig. 12 illustrates an example of a group call service application (described now). According to this example, CUG1 is a CUG with four users A, B, C, and D. Users A and B are currently served by the group call switch G1, C is served by the group call switch G2, and D is served by the group call switch G3. These users have been assigned a unique MIN by the group call service provider. For the sake of brevity, the only MIN is called A, B, C, and D here. CUG1 has a unique identifier, which is assigned by the service provider and is recorded here as CUG1. The definition of CUG1, that is, the member list of CUG1, is kept in the distributed component of CUG AD and is called AD1.

In the first example use, A initiates a group call to group CUG1 (that is, to B, C, and D). Figure 12 is a call flow chart of the group call.

A initiates a group call request to the closed user group CUG1. In the IS634 interface, this request is recorded as CM_service_request of CUG1. In the case of the radio access network ("RAN"), the request is very similar to any other call request. The IS634 command and the included information unit include the calling number and the called number in addition to the information entry (for example, assume that it is CUG1 in the current example). In at least some cases, the RAN does not have the logic to distinguish between valid numbering plans and invalid numbering plans, and this logic can be implemented in the MSC. In this case, the RAN forwards the number information to the MSC as an IS634 message set. Since the proxy switch intercepts such messages, the number information can be used by the proxy switch. Since the proxy switch can function as an MSC in at least some ways, the proxy switch can determine that the incoming call request is not a regular call request, but a group call request for a closed user group. In such an example, the proxy switch assumes the role of the group call switch and initiates the group call process.

The group call switch G1 sends back a channel allocation request to A. G1 also initiates directory processing to retrieve the definition of group CUG1 from component AD1 of CUG AD. The required response from AD1 includes a list of MINs corresponding to the CUG1 group members (B, C, and D). When receiving the response, G1 obtains the following information: The members of CUG1 (except A) include B, C and DMIN B is responsible for MIN by switch G1 (ie itself) C is responsible for MIN by switch G2 and MIN D by switch G3 is responsible for G1 Initiate a call establishment request to B (G1 is responsible for itself), and send the call establishment request to G2 and G3 of C and D, respectively. Therefore, G2 and G3 act as proxy switches for G1 for this special group call. G1 also ensures the creation of a new environment in which TDM communications are directed from the RAN to the media gateway, and the media gateway converts the TDM communications into RTP/UDP/IP packets and sends the RTP/UDP/IP packets to the multicast router. Notify the multicast router to receive these packets, add A to the multicast group, and multicast these packets to the specified multicast group. In the call flow chart of FIG. 18, all instructions are shown as "join a multicast group (TDM A)".

Then G1 waits for connection messages from B, G2, and G3. You can receive any of these three messages in any order, or you can receive only a subset of them. In this example, G1 first receives the connection message from B, and then the connection message from G2 and G3. After receiving the connection message from B, G1 will send a "Join Multicast Group (TDMB)" message, which will make the media gateway receive the RTP/UDP/IP communication from the multicast router and convert the RTP/UDP/IP communication TDM, and then send TDM to G1, G1 will make TDM sent to B through BSC and BTS. Notify the multicast router to add B to the current multicast group. In this way, for the current group call, the switch G1 serves as the source of TDM communication, and the switch G2 serves as the receiver of TDM communication.

In this example, C sends the connection message to G2 (that is, the switch is responsible for C), and G2 then sends the connection message to G1. Then G1 sends a "join multicast group (TDM C)" message to make C join the group call in receiving mode. Similarly, the connection message from D to G3 is also forwarded to G1, making D join the multicast group.

The control plane informs the media gateway to only receive or send packets on a certain RTP port in a special environment. The router multicasts the packet to the members of the multicast group.

Since G1 has received the confirmation that at least one user has joined the group call at this time, G1 sends a success tone to A, indicating that the group call is now ready. A is in sending mode, B, C, and D are in receiving mode, and the multicast router can perform multicast to B, C, and D.

In another example use, the right to speak is to be converted, as shown in the call flow chart in FIG. 13. Specifically, A gave up speech control, and C gained speech control.

In order to give up speech control, A signals G1, which causes G1 to put the current group call in inactive mode. (As explained above, if the user does not obtain the speech control right within the specified time measured by the system timer, the group call will be terminated.) C sends a speech acquisition command to its responsible switch G2. For the current group call, the switches G2 and G3 are the agents that control the switch G1, so the speech command is forwarded to G1. G1 sends a modify environment command to the media gateway, so that the media gateway associated with switch G2 receives the incoming TDM communication from C, converts the TDM communication into RTP/UDP/IP, and then sends the RTP/UDP/IP to the multicast router. Notify the media gateway to stop receiving TDM communications from A. Notify the multicast router to change the mode of A to receive and change the mode of C to transmit. G1 sends a speech permission message to G2, and G2 sends a success tone to C, indicating that C can now speak. As shown, the multicast session can now continue, where C is the speaker and A, B, and D are the listeners.

For example, if B sends an acquire speech message, and C now has speech control, the message will be sent to Bs responsible switch G1, and G1 will reject the request (because C did not give up speech control) and send a failure tone message to B.

In a further example, the situation is simplified so that the group call has two participants A and B, and A has speech control. The network includes a separate group call switch G1, which includes a control plane CS and controls two media gateways MG1 and MG2. The communication to and from A is transmitted through MG1, and the communication to and from B is transmitted through MG2, as shown in the call flow chart of FIG. 14.

Since A has the speech control right, the system responds to the abandon control command from A only. The control plane CS of G1 receives the command, and G1 sends a modify environment command to MG1 to notify MG1 to modify the environment of the call by rejecting the TDM from A. The system enters an inactive state, waiting to obtain a control command. If such a command is not received within the specified time measured by the inactivity timer, a call release request will be sent from CS to A and MS B. In addition, a command to invalidate the environment is sent to MG1 and MG2. When the release completion message is received from A and B, and the environment void complete message is received from MG1 and MG2, the group call release sequence is completed.

Figure 15 illustrates an example of a call flow regarding database record retention in a roaming situation, such as location update.

Three mobile stations A, B and C are involved. A and B are in charge of the control plane CS1 of the group call switch, and C is in charge of the control plane CS2 of another group call switch. A roams and sends out location updates, and CS1 receives location updates through BTS and BSC. CS1 consults the index based on IMSI/MIN/ESN to determine the appropriate HLR' of A, such as HLR'1. HLR'1 determines all CUGs to which A belongs through the database. Based on this information, HLR'1 sends a location update request to CUGAD. Therefore, in the CUG to which A belongs, CUG AD includes the updated position of A.

When roaming, B will also send a location update, which reaches CS1. Based on IMSI/MIN/ESN, CS1 determines the appropriate HLR' of B, which is HLR'2. CS1 sends a location update request to HLR'2, and finds all CUGs to which B belongs. HLR'2 sends a location update request to CUGAD, allowing CUG AD to update all CUGs of B with the new location of B.

Receive location update from C at CS2. CS2 determines the appropriate HLR', such as HLR'2, and sends the location update to HLR'2, and HLR'2 makes CUG AD update the corresponding CUG of C.

The changes to the above embodiments all contribute to the realization of the group call of the present invention. But certain subsets of functionality still provide advanced advantages in the field. For example, group calls using technologies other than multicast on IP networks can be inserted to replace IP multicast connections.

In another example, a group call switch can be used at the trunk ("the back") end of the MSC. In such an embodiment, the group call function may operate as described below.

Figure 16 illustrates a group call switch used on the back end of the MSC with a fixed link using standard ISDN User Equipment ("ISUP") landline signaling. The MSC is connected to a home location register ("HLR"), and the HLR uses the IS-41 (also called MAP) protocol. The group call exchange and the MSC are also interactively connected to a bearer trunk line that transmits voice communications between the two exchanges. The group call switch has from its data plane (also called a media gateway), a TDM connection to the PSTN, and an IP connection to the IP network. The group call switch can also use IS-41 to query the HLR. (Figure 16 shows two MSC switches independently connected to the PSTN, but both switches can be connected to the same PSTN.) Both group call switches can use Active Directory (CUG-AD) through the IP network.

The configuration shown in Figure 16 can be used for group calls. For example, mobile station (MS) A can be connected to MSC-1 through RAN-1, and two mobile stations B and C can be connected to MSC-2 through RAN-2. User A may have a CUG that includes B and C as members. As explained above, A can use a special group call initiation sequence to signal to the MSC, hoping that the MSC will establish a group call. The logic of MSC-1 determines that the incoming call request is a group call and uses the ISUP protocol to transfer the call request to the group call switch GCS-1. The group call switch GCS-1 uses its internal logic to access the active directory CUG-AD to determine the members of the called CUG.

In this example, the query produces the MIN numbers of members B and C. In the example configured on the back end of the MSC, the group call switch cannot use location update; therefore, the HLR' does not contain the current location of the called mobile station. However, the HLR contains this information. Therefore, the group call exchange GCS-1 can perform IS-41 query ("location request") to make the HLR query the locations of mobile stations B and C. For this example, mobile stations B and C may currently be located in the handover area controlled by MSC-2. According to standard mobile phone conventions, this information is contained in the HLR database, and the HLR is now in contact with MSC-2 (via "route request"). Since MSC-2 is using GCS-2 on the trunk side, the routing request from HLR will be received by GCS-2. GCS-2 returns the temporary local directory number ("TLDN") to the HLR, and the HLR forwards this information to the sender of the original location request (GCS-1). GCS-1 determines that the TLDN belongs to GCS-2 and informs GCS-2 of the group call information. GCS-2 notifies MSC-2 to establish group calls to mobile stations B and C. Further interactive processing continues. As described in the above non-backend situation, MSC-1 and MSC-2 are actually transparent to group calls.

It is obvious from Figure 16 that the back-end configuration of the switch used for group calls has side benefits. In such a configuration, conventional landline telephones (such as telephone D shown in FIG. 16) can also be included in the group call. Therefore, CUG members can register the route phone number as an "arrival" number in the CUG Active Directory. If a user such as D needs to be included in the group call, the corresponding group call switch can make the serving MSC use the stored user arrival number to complete the PSTN call to D.

As mentioned above, IP multicast technology can be used as the basic transmission technology for group calls. However, in at least some cases, it can be proved that the standard implementation of IP multicast technology cannot effectively meet the needs of widely distributed CUG members. Establishing multicast channels dynamically, call after call, to transmit communications between multicast-enabled routers, may take a very long time. A user who has experienced a long group call establishment time may hang up or try to call again, which may cause user dissatisfaction. Long delays in call setup time can also lead to ineffective signaling network utilization.

For example, as shown in FIG. 17, the numbers of multicast-enabled routers (such as MCR-1, MCR-2, MCR-3) can be presented in the IP network. In this example, the locations of these routers are fixed and will not change. The multicast router is connected to the group call switch (before the MSC or at the back end of the MSC as described above), and therefore connected to the mobile phone through the corresponding radio access network (RAN). (Although Figure 17 shows each multicast router connected to a single group call switch, in fact, a multi-group call switch can also be connected to a single router.) If the user initiates a group call based on the members of the relevant CUG, one Or multiple multicast routers can be included in the call. In particular, as mentioned above, a channel is established between the corresponding multipoint routers. If the establishment of these channels is excessively delayed, the quality of group calls may be compromised.

As described now, IP channels can be established early in this way, and the number of group call activation requests that can arrive later can be served by these channels. Due to the early establishment of the channel, in anticipation of future group call requests, the establishment delay after activation will be reduced or eliminated. The work involved includes: predicting the requirements of group calls that are expected to arrive in the future, and determining the topology of the IP channels that need to be established to meet the predicted requirements.

The forecast is required to rely on historical record information, which is the group call record in form. The history record of group calls is divided into a series of windows, each window is defined on a period of time (called "window length"), which can range from a few minutes to tens of minutes, and is related to the average occupancy of the group call Time is proportional. The actual window length used for the required prediction may vary based on the accuracy of the required prediction and the computing resources that will be used to provide the prediction. Generally speaking, a shorter window length can produce a more accurate prediction than a longer window length, but it will also cause more waste and consumption of computing resources. In addition, a shorter window length is more sensitive to burst channels, and it is less likely to eliminate deviations. Each window includes a large number of group call requests and parameters describing these calls, that is, the number of GIR calls between any two multicast routers. As used below, X(I, J, N) represents the number of group calls between routers I and J in window N.

The following example illustrates how to predict the future requirements of group calling. In this example, the history record of the group call is divided into 4 windows, window 1 is the earliest in time, and window 4 is the latest in time (that is, the current window). In order to calculate the requirements in the next window (ie window 5), the following filtering formula should be used: X(I,J,5)=(1-α)*X(I,J,4)+α*(1-α) )*X(I,J,3)+α*α*(1-α)*X(I,J,2)+α*α*α*(1-α)*X(I,J,1) In the formula, α is a weighting factor determined empirically, and its value is between 0 and 1. As indicated in the formula, the prediction of the communication in the next time window is based on: a window closer to the current window has a greater weight than an earlier window. If the formula is rewritten into a more general circular form, this choice is obvious: X(I,J,N)=(1-α)*X(I,J,N-1)+α*X '(I, J, N-1) where X'(I, J, N-1) is the filtered estimate, which compresses the past history to N-1.

In this formula, "current" refers to the last window in the time series. Based on this formula and given the required history records (windows 1 to N), as the parameter value X (I, J, N) exemplified, the table (requirement matrix) of the value T (I, J) can be calculated for the following A window, so that the value in row I and column J represents the number of predicted group calls in the upcoming window between multicast routers I and J.

For determining the channel topology used to meet the requirements, use the following information as input: the point-to-point requirement matrix for group calls between any two multicast routers; the cost structure of the service providers channel, that is, between any two routers The cost of establishing a channel with a special capacity; the delay guarantee of the service provider, that is, the maximum delay in the IP transmission network between any two special multicast routers; and the quality of service ("QoS") constraint, which needs to pass group calls Come to be satisfied.

Taking into account such input, the channel topology between multicast routers (that is, which channel with which capacity is connected to which multicast router) is determined in this way, and the topology should take into account the following constraints: The channel starting from a multicast router cannot exceed the routers total output capacity (bits/sec), and the number of channels starting from each multicast router cannot exceed the internal limit of the routers number of channels.

As described below, the mathematical optimization technique of integer linear programming ("ILP") can be used in determining the least cost topology, taking into account the following points explained as follows: the identification of "NP-hard" situations; and The working formula is expressed as a degree constrained multi-product flow, which can be solved by using ILP technology.

According to the theory of NP-hardness, examples that can be shown as NP-hardness are not expected to have effective algorithmic solutions. In a hypothetical situation, such as X, it can be shown as NP-hard by observing that it is a generalization of the multi-product flow problem that is not easily separated (see J. Kleinberg, "Single source unsplittable flow", Proc. Of the 37th IEEE Symposium on Foundations of Computer Science, 1996).

Therefore, the channel topology can be calculated separately on the table shown below, which is suitable for the application of ILP approximation technology.

Input (INPUT)N represents the number of multicast routers in the network.

D(max, I) represents the maximum number of channels that router I can establish.

P(t, 1) represents the unit cost of channel 1 of type t. Here "1" represents a pair of nodes 1=(i, j) of multicast enabled routers (MCR) i and j.

τ represents the set of all possible types of trunks (DS0, DS1, OC3, etc.).

T(I, J) represents the demand matrix, that is, the predicted group call traffic between routers I and J in the future time period.

C(I) represents the capacity of router I (bits/sec).

R(I,J) is the set of all feasible routes used to send communication between routers I and J. (The preprocessing step is to generate all the service path quality between MCR I and J.) Output result variable Y(t, 1): the number of units of type t trunks allocated on link 1.

X(p): The number of communication streams on path p.

z1: Binary value variable. If a non-zero capacity is allocated to link "1", the value is 1; otherwise, the value is 0.

Minimize the ILP formula ΣlΣtτpltylt]]> so that-the requirements are met (the channel topology meets the requirement matrix): ΣpRijxpTij]]> i, j-sufficient channel capacity (all paths All flows on the network are controlled by the capacity of the selected trunk): ΣijΣpRijlpxpΣtτylt]]>l-port constraints (start and stop on a router The number of channels cannot exceed the maximum internal set of the router): Σl:(i,j)=^lzlDimax]]>iΣl:(i,j)=^lzlDjmax]]>j Formalize the above analysis.

Input N: the number of routers Dimax: the maximum number of channels that router i can establish p1t: the unit cost of channel 1 of the trunk of type t τ: the set of all possible trunk types

T(I,J): required matrix T(I,J)Ci: capacity of router i (bits/second) Rij: the aggregate output result of all feasible routes for sending communication between routers i and j. Variable y1t: assigned to The number of units of t-type trunks of link 1 xp: the traffic on path p (communication traffic) z1: if the link is allocated non-zero capacity, it is equal to 1, otherwise it is equal to 0ILP. Minimize the formula ΣlΣt&tau ;pltylt]]>make-requirements meetΣpRijxpTij]]>i,j-sufficient channel capacityΣijΣpRijlpxpΣtτylt]]>l -Port constraint Σl:(i,j)=^lzlDimax]]>iΣl:(i,j)=^lzlDjmax]]>j-Channel existence constraint ZlΣt&Element ;τyltlM]]>where M=(ΣiΣjTij)+ϵ]]> where the damping system ε is a parameter greater than 0 given by the user.

In addition, with regard to the embodiments already described in the environment of special wireless technologies (such as TDMA or CDMA protocols), these embodiments can also be modified to work through one or more of the following wireless technologies: TDMA , CDMA, GSM, IS-136 and other 2G and 3G protocols.

Because the foregoing is only exemplary embodiments, it should be apparent to those of ordinary skill in the art that, without departing from the spirit and scope of the present invention, the