System and method of expediting call establishment in mobile communications

System and method for accelerating call establishment in mobile communication

Cross-reference of related applications This application requires the priority rights of the U.S. Provisional Application Serial No. 60/386,883 filed on June 7, 2002, entitled "System and Method of Optimizing Latency Time in Calling Systems", the entire content of which is incorporated here. Enlisted as a reference.

This application is a continuation of the U.S. Application Serial No. 09/845,934 entitled "System and Method of Group Calling in Mobile Communication" filed on April 30, 2001, the entire content of which is incorporated herein by reference.

Technical field

The present invention relates to mobile communication, and more specifically, to speeding up call establishment in mobile communication.

Background technique

As described in the co-pending US application Serial No.09/845,934, all modern mobile communication networks have a hierarchical configuration, where the geographic "coverage area is divided into multiple smaller ones called "cells Geographical area. Referring to FIG. 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. Sometimes BTS and BSC are collectively referred to as Base Station Subsystem (BS) 107. Several BSCs 106b-n may be aggregated into a mobile switching center (MSC) 110 via fixed links 108a-n.

The MSC 110 functions as a local exchange (with additional features that handle mobility management requirements) and communicates with the telephone network (PSTN) 120 through a trunk group. In the mobile network in the United States, there are concepts of home MSC and serving MSC. The home MSC is the MSC corresponding to the switch associated with the mobile station (MS, which is also referred to as "mobile phone", "mobile phone handset" or "cell phone"); the association is based on the MS's phone number, such as the area code. (The home MSC is responsible for the HLR discussed below.) On the other hand, the serving MSC is a switch used to connect MS calls to the PSTN (when users roam in the area covered by the service provider, different MSCs perform the functions of the serving MSC ). Therefore, sometimes the home MSC and the serving MSC are the same entity, but at other times they are not the same entity (for example, when the MS is roaming). Typically, the visitor location register (VLR) 116 is co-located with the MSC 110, and a logically single HLR is used in the mobile network. The HLR and VLR are used to store many types of user information and profiles.

Simply, one or more wireless channels 112 are associated with the entire coverage area. The wireless channel is divided into channel groups allocated to independent cells. This channel is used to transmit signaling messages to establish a call connection, etc., and to transmit voice or data information when establishing a call connection.

In a relatively high-level abstract concept, mobile network signaling involves at least two main aspects. One aspect involves signaling between the MS and the rest of the network. For 2G ("2G" is an industry term, that is, "second generation") and later technologies, the signaling relates to the access method used by the MS (for example, time division multiple access, or TDMA; code division multiple access, or CDMA), wireless channel allocation, authentication, etc. The second aspect relates to the signaling between different entities in the mobile network, such as the signaling between 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 (SS7).

Transmit and receive different forms of signaling (as well as data and voice information) according to different standards. For example, the Electronic Industries Association (EIA) and the Telecommunications Industry Association (TIA) have assisted in the development of many American standards, such as IS-41, which is the MAP standard. Similarly, CCITT and ITU assisted in defining international standards, such as GSM-MAP, which is an international MAP standard. Information about these standards is well known and can be found in related organizations and documents, see, for example, Bosse, Signaling in Telecommunications Networks (Wiley 1998).

To transfer the call from MS 114, the user dials on the cell phone or other MS and presses "send". MS114 sends the called number indicating the service request to MSC110 via BS107. The MSC 110 cooperates with the associated VLR 116 (see below) to determine whether to allow the MS 114 to obtain the requested service. The serving MSC routes the call to the local exchange of the called user on PSTN 120. The local exchange reminds the called user terminal, and the response signal is routed back to the MS 114 through the serving MSC 110, and the serving MSC 110 then completes the call to the MS. Once the setup is complete, you can make a call.

In order to transfer the call to MS114, (assuming the call originates from PSTN120) the PSTN user dials the phone number associated with the MS. According to at least US standards, the PSTN 120 routes the call to the MS's home MSC (which may or may not be the MSC that provides services to the MS). The MSC then queries the HLR 118 to determine which MSC is currently serving the MS. This can also be used to notify the serving MSC that a call has occurred. The home MSC then routes the call to the serving MSC. The serving MSC pages the MS via the appropriate BS. The MS responds and sets up the appropriate signaling link.

During the call, BS107 and MS114 can work together to change the channel or BTS102 when needed, for example due to signal status.

Newer services have been added to mobile communication networks, such as "data calls" for the Internet. For the Internet, multicast communication refers to the transmission of the same data packet to multiple selected targets on the Internet Protocol network. (In contrast, broadcast communication refers to indiscriminately transmitting data packets to all targets, while unicast communication points to a single target to transmit data packets.) Each participant in the multicast receives the data from any other participant in the multicast Information transmitted. Users who are connected to the network and are not participants of a specific multicast do not receive the information transmitted by the participants of the multicast. In this way, multicast communication uses only network components (for example, switches and relay stations) actually required for multicast transmission.

In multicast processing, when a potential participant (host) is guided to join a specific IP multicast group, the host sends a "request to join" message to the nearest multicast capable router to request to join the multicast group Group and receive messages 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 appropriate, router R propagates the request upwards to the multicast source.

When an IP packet is received from the group X, for example, the router R maps the IP multicast group address to an Ethernet multicast address, and sends the resultant Ethernet packet to an appropriate switch.

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

For the interaction between MSs, two versions of Nextel service (known as Nextel Direct Connect®) have been proposed, which use dedicated mobile wireless services, in http://www.nextel.com/phone_services/directconnect.shtml Description), used for dedicated connection calls between MSs. Both versions of this dedicated connection call require all members to be in the same switching area controlled by the BSC/DAP (dispatch application processor) combination. In the first version, two mobile phone users, such as one-to-one calls between A and B, are allowed. When A wants to have a dedicated connection communication with B, A enters B's personal identification number, presses the "push to talk (PTT)" button, waits for a voice prompt indicating that B is ready to receive, and starts speaking. To listen, A releases the PTI button. If B wants to speak, B presses the PTT button and waits for the voice confirmation that A is ready to receive. The service allows users to select personal identification numbers from a scrolling list displayed on mobile phones, or allows users to search a list of pre-stored user names.

In the second version, calls between members of a pre-defined user group (called a "Talkgroup") are allowed, which are identified by numbers. The mobile phone allows searching the call group number through the control interface of the mobile phone. To set up a group call, initiate a user, such as A, look up the group call number in the mobile phone, press and hold the PTT button, and, when receiving a voice confirmation such as a chirp, can start speaking. When A presses down the PTT button, all other members of the call group in the group call can only listen. If A releases the PTT button, another member of the group call can hold down the PTT button, get the control of the voice confirmation reminder, and start speaking.

One of the earliest examples of group call systems is the TwoWay Talk Radio (TWTR) system, which is an analog half-duplex wireless system, which predates NextelDirect Connect®, and in which, during the transmission process, the transmission (Broadcast) transceivers turn on their transmitters and turn off their receivers, while receiving transceivers turn their transmitters off and turn their receivers on. The delay in the TWTR system is almost zero, which is governed by the speed of radio waves and the number of propagation times of electronic components. Another feature of this system is that the broadcast caller has no prior knowledge about the existence of the listener. Only when at least one listener responds, the caller can determine the existence of some listeners. Therefore, a typical model of a group call includes a "human agreement", where when a group call is established, the caller, for example, using a phrase such as "are you there?", first determines the existence of one or more listeners. If no meaningful communication can occur in the group call before the presence of the listener is confirmed, a delay period called Human Round Trip Response Time (HRTRT) indicates the perceived delay of TWTR. In at least some cases, when the mobile phone can easily access the called party, the range of HRTRT is 1.5 to 4 seconds relative to the delay caused by the speed of radio waves (approximately 0.03 milliseconds at a distance of 5 miles).

In some PTT systems, digital wireless services are used for coded and framed half-duplex voice communication. Unlike the TWTR system, the PTT system based on digital wireless services uses explicit signaling to establish group calls. Due to the active behavior of the explicit signaling and group call establishment, the encoding and digital framing of the original analog voice signal, and the transmission delay, the system has a large delay, which can be in the range of at least in some cases 750 milliseconds to 1.5 seconds. Moreover, the difference between the PTT system based on digital wireless services and the TWTR system is that the caller understands the presence of the listener. Typically, a PTT system based on digital wireless services plays a sound called "chirp" to indicate the presence of one or more listeners, after which the caller can make a call. In this way, since the caller needs to know whether the listener is available and paying attention, the HRTRT delay remains relevant in a PTT system based on digital wireless services. This "chirp" only indicates that the phone is available; it does not give an indication of the status of the listener. The caller does not know whether the listener is busy with other things or whether the phone is at a certain distance from the listener, for example, at a kitchen table a few feet away from the listener. In at least some cases, in the current digital wireless service-based PTT system, when the mobile phone is easily available to the listener, the range of HRTRT can be 2 to 5 seconds.

In some implementations of the PTT system based on digital wireless services, it uses a standard air transmission interface (RF modulation), such as the CDMA 1xRTT interface, and the HRTRT can be as large as 12-15 seconds. These interfaces are not optimized for PTT-type group calls, and different delays are introduced when used to transfer PTT calls. A typical PTT call in a 1xRTT network can have an HRTRT delay of 15 seconds, which creates a serious obstacle to the successful development of a new PTT system.

The total delay includes at least the following factors. As mentioned above, the presence delay is the delay caused by the time it takes for the caller to determine that the called party is present and able to start the call. The existence delay occurs once when the caller initiates the group call. The call setup delay is the delay caused by the time it takes for the called party to determine the caller's intention. The call setup delay occurs once at the beginning of the group call. Media delay is the delay caused by the time consumed by the audible pulse sent by one party in a group call before being heard by other parties in the call, and the media delay includes buffering time, encoding time, and transmission delay of voice media . As mentioned above, HRTRT is the delay caused by the time it takes before the caller hears the called party, that is, after the caller speaks and releases control, and the called party hears the request for control and speaks.

The traditional 1xRTT PTT service uses Packet Switched Data (PSD) as the transmission mechanism, uses RTP/UDP/IP, uses EVRC (Enhanced Variable Rate Codec) for voice encoding, and uses SIP (Session Initiation Protocol) as the explicit Signaling protocol. In the 1xRTT network, if there is no active activity of packet data in a time period called the sleep interval (which is a parameter that can be set by the network), the mobile phone enters the sleep state. When the data activity for the dormant phone starts, the phone performs the transition from the dormant state to the active state. In this way, if a participant of a group call has a dormant mobile phone, the time it takes for the mobile phone to enter the active state from the dormant state also contributes to the overall delay in the group call. In at least some cases, for participants with active mobile phones, the average call setup delay (including existing delays) can range from 1.5 to 3 seconds, while for participants with dormant phones, the average call setup delay (including There can be a delay in the range of 5 to 10 seconds. In at least some cases, the average media delay can range from 400 milliseconds to 600 milliseconds, and for participants with active mobile phones, the range of HRTRT can be from 5 to 7 seconds, while for participants with dormant mobile phones, The range of HRTRT can be 7 to 14 seconds.

Another aspect of the implementation of a typical 1xRTT network is the implementation feature of the "RP environment". According to this feature, if there is a lack of active behavior in a certain period of time, the PPP session associated with the mobile phone is controlled by the network, that is, the RP node termination.

For the lack of active behavior in a certain period of time, according to the sleep characteristics of the 1xRTT network, the PPP session is maintained, but the air transmission resources are released for other purposes. When the data is available for transmission, it takes time to restore the over-the-air transmission resources (ie, wake up the mobile phone), which also contributes to the delay.

Summary of the invention

Generally, the present invention provides a system and method for mobile communication, and in particular, provides a system and method for accelerating call establishment in a communication system, especially in "push-to-talk" calls and group calls. When the mobile station (MS) is in the dormant state, the mobile station is prepared for a half-duplex mobile communication telephone call. In response to the user initiating a half-duplex mobile communication phone call, a half-duplex mobile communication phone call is established based on the preparation of the mobile station.

By accelerating call establishment, the mobile communication system can provide users with a PTT system or group call system with almost no delay. Providers can efficiently allocate network resources based on economic incentives to effectively reduce delays. Users can communicate quickly, accurately, and cost-effectively, while having prior knowledge about the availability of other users.

Description of the drawings

In the drawings, Figure 1 is a system diagram of a prior art mobile network; Figure 2 illustrates a system block diagram including group call or "push to talk" logic; Figures 3 to 4 illustrate the proxy switch in the mobile network And some configurations; Figures 5-6 and 8 illustrate the structure of a group communication system or "push-to-talk" communication system; Figures 7, 9-20 are calls that use a group communication system or a "push-to-talk" communication system Flow chart; and Figures 21 to 28 are graphs showing the test results of the delay reduction technique.

detailed description

The co-pending US application Serial No. 09/845,934 describes a system and method for configuring calls between members of a pre-defined mobile phone user group. For Figure 2, as described in the co-pending US application Serial No. 09/845,934, the proxy switch or other device 1010 that implements group call logic detects the group call initiated by the member 1012A of the group 1014, and automatically Try to connect all members of the group 1012A, 1012B, and 1012C in the group call. In a specific implementation scheme, the communication in the group is half-duplex (that is, only one member can speak at a time), and the voice service of the group is in the form of a multicast session on the Internet Protocol (IP) network get on.

For the case where the group call logic is implemented by a proxy switch, the proxy switch can be submitted on November 22, 2000, entitled "System and Method of Servicing Mobile Communication with a Proxy Switch" co-pending US application Serial No Operate as described in .09/721,329, which is incorporated here as a reference. As described in the co-pending US application Serial No. 09/721,329 and illustrated in FIG. 3, the exchange 1034 is performed between at least one mobile switching center (MSC) 1030 and at least one base station subsystem (BS) 1032 operating. This exchange allows communication traffic to be transferred in or out of an alternative network 1036 such as an IP network. The exchange is transparent, so that the MSC and BS can work together with the switch of the present invention without any modification.

The proxy switch described in the co-pending US application Serial No. 09/721,329 includes signaling message processing logic 1038 for receiving 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 processing logic, and sends a response message to the MSC or BS that transmits the signaling message. The message interception logic also prevents the signaling message from being delivered to the other of the BS and MSC respectively. The message conversion logic 1042 works in cooperation with the signaling message processing logic, and converts the 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 transmission logic 1044 works in cooperation with the signaling message processing logic, and transmits the signaling message from one of the MSC and the BS to the other of the BS and the MSC, respectively.

The bearer circuit group 1046 from the BS is allocated to the proxy switch. Receive and analyze the signaling messages between the MSC and the BS to determine whether they correspond to the assigned bearer circuit group. If it corresponds, the control information in the signaling message is transferred to the alternative communication network; and the information carried on the bearer circuit group is transferred to the alternative network.

FIG. 4 shows a preferred configuration of the proxy switch 300, where the proxy switch 300 is placed between the BS107 and the MSC110. In the proxy switch, only the subset 306 of the trunk lines performing user services needs to be terminated; the other trunk lines 308 can directly connect the MSC110 and the BS107. All control links 312 from the BS 107 terminate at the agent 300. The proxy switch includes a control plane 302 and a data plane 304 (also referred to as "bearer plane"). The control plane 302 handles all signaling services, and the data plane 304 handles all user services of the trunk line connected to the proxy switch.

In some embodiments, there is one-to-one communication between the MSC and the proxy switch. Several BSs can work with a single proxy switch.

The proxy switch 300 includes software that accepts all signaling messages and, depending on the message and the state of the system, performs at least one of the following operations: 1. Deliver the unchanged message to the one addressed in the message MSC or BS; 2. Intercept the message between MSC and BS; 3. For some intercepted messages, convert the intercepted message into a different message, and use the converted message to replace the original intercepted message and send it to The MSC or BS addressed in the intercepted message; 4. Transfer the message from the mobile network and PSTN network to an alternative network, such as an IP network.

The following describes the types of operations performed in each case and the triggering events.

In many cases, the proxy switch 300 can be used as the MSC 110, especially when a message from the MS 114 is transferred and traffic is directed to an alternative network. In this role, the proxy switch assumes the responsibilities and tasks performed by the traditional MSC. Some of these functions and tasks involve mobility management. Consider the case of a roaming MS; when it roams from one cell to another, it may roam into a cell served by a different MSC, so handover between the source MSC and the target MSC is required. If the proxy switch 300 calls in a message, and the call/session is directed to the alternative network, the proxy switch manages the handover similar to the way the traditional MSC manages the handover.

Another function of the proxy switch involves the allocation of resources. In particular, when the MS initiates a message requesting a new call/session, it needs to allocate an appropriate line (channel) for the session. Depending on the configuration and system status of the system, similar to the way the traditional MSC allocates lines, the proxy switch performs this allocation.

Fig. 5 shows an exemplary configuration in which the proxy switch 300 is connected to several alternative networks, such as an IP backbone network 412, or a circuit-switched-based network 414, for example, different telecommunications companies. These alternative networks can be used to deliver voice and/and data services to the desired destination while avoiding the entire PSTN120 or part of the PSTN120 from being connected to expensive MSC resources. Alternatively, these configurations can be used so that circuit-switched services can be transmitted back to different networks; for example, circuit-switched services from Nashua NH can be transmitted back to the MSC in Waltham MA. Or they can be used to connect to other networks. For example, the IP backbone network 412 may communicate with the IP voice network 418 or the Internet 416. As explained in the co-pending application, when the service is transferred to the alternative network, control information from the bearer circuit on link 306 (e.g., from signaling messages) as well as voice or data can be sent via the alternative network.

In the specific implementation of the group communication system described in the co-pending US application Serial No.09/845,934, it provides fast and easy access to mobile communication users (users) belonging to a closed user group (group or CUG) The ability to communicate with each other, and thus start talking with each other. Each group includes two or more users ("members"), and users can belong to multiple CUGs. The call can occur between two members of the group (private mode), or between all available members of the CUG (public mode). Group communication systems use traditional mobile communication devices such as cellular phones and mobile PDAs.

In a specific implementation scheme, the group communication system is logically placed in the proxy switch between the MSC and the BSC (as described above) to implement the group call logic to intercept the group call initiation message and bypass the MSC and PSTN, and the group call is implemented as an IP multicast session that performs Voice over IP (VoIP). Multiple MSCs across the aggregation network can provide services to users in groups in different geographic locations, which rely on one or more wireless technologies, such as CDMA, TDMA (including 1S-136 and GSM), GPRS, and Third-generation technology. For example, among group members who join any group call, one or more users can roam in the GSM network at the same time, while one or more users roam in the CDMA network. The control information related to the group call can be made available to one or more users, such as displaying the participants of the group call while the group call is in progress. By using standard coding schemes, such as MIN, IMSI and ESN, group callers can dynamically create and modify group call lists.

The example of FIG. 6 shows the general structure of an exemplary embodiment of a group communication system. FIG. 6 shows four users in a group call. They use wireless devices 1060A-1060D to connect to different BTS systems 1062A-1062D. For the following description, it is assumed that the wireless device has sound playback and text display capabilities. The BTS is connected to base station controllers (BSC) 1064A to 1064D, and the BSC is connected to proxy switches (group call switches) 1066A to 1066C that implement group call logic. Each group call switch is connected to an MSC such as MSC 1068A, 1068B, or 1068C. Provide at least one group call switch for each MSC in the group call service enabling network. For signaling information, each group call switch is logically located between the corresponding BSC and the corresponding MSC. The group call switch receives the signaling and data from the MSC, and receives the signaling and data from the wireless device via the BTS and BSC in the opposite direction. Each group call switch operates such that neither the BSC nor the MSC is aware that the group call switch is located between the BSC and the MSC. The signaling and control information from the MSC and BSC is intercepted by the group call switch, and is seamlessly transferred to related elements when needed without any perceptible 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, which provides access to the CUG Active Directory 1074 and the Enhanced Home Location Register (HLR) 1076 access.

As described above about the proxy switch of the co-pending application, the group call switch includes a control plane and a data plane. The function at the control plane is to terminate signaling messages from BSC or MSC or both. 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 for forward transmission to the MSC or other elements. The control plane also supports the multicast functions described below.

In a specific embodiment, the data plane of the group call switch receives TDM services from BSC or MSC or both, and uses TDM cross-connect (DACS) (Figure 4) to connect the incoming services to the outgoing target. In other embodiments, the data plane may also receive incoming IP services from the base station complex (also called "radio access network", or "RAN"), and exchange the incoming IP services to the outgoing IP business. The program control in the control plane determines the cross-connection between the incoming TDM service and the outgoing target (especially the target on the traditional MSC and/or IP network).

In the case where the MSC is used as a target outgoing from the DACS, the group call switch is basically transparent to the network; business and control flow seamlessly from the BSC to the MSC, and seamlessly from the MSC to the BSC. When the outgoing target is on the IP network as a substitute, the media gateway in the data plane (described in the co-pending application) transfers the selected part of the incoming TDM service from the MSC, and converts the incoming TDM service It is an RTP/UD/IP service and inserts the RTP/UD/IP service into the backbone IP network.

CUG Active Directory (CUG AD) 1074, also known as Group Call Registration Record (GCR), is a database system containing CUG data. In the specific implementation, the CUG AD in Figure 7 is implemented as a distributed database system with scalability. 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, the query requires the definition of the specified CUG, and the result is a list of group user IDs of all members of the specified CUG. For example, a query specifying CUG ID 2347 can cause CUG AD to produce the result that it identifies the mobile identification numbers (MIN) xxx, yyy, zzz, and www of the four users in the CUG. In a specific implementation, the MIN number is assigned to the user of the GIR service by the service provider.

For the system, each CUG is identified by a unique identifier ID derived from the CUG namespace. The CUG namespace is divided in such a way that different parts are allocated to different distributed parts of the CUG AD. Make the partition index of the partition plan available to all group call exchanges. When the group call switch needs to restore the definition of CUG, the group call switch can use the index to determine the CUG AD part to be inquired.

In the specific implementation described in the co-pending US application Serial No. 09/845,934, the group call service uses IP multicast to operate in an IP network. IP multicast allows a source to send a single copy of a VoIP packet stream, which is received by multiple receivers who have been explicitly registered to receive the stream. Multicast is based on the concept of the receiver, that is, the receiver joins a specific multicast conversation group, and the stream is delivered to all members of the group through the network infrastructure. Only one copy of the multicast stream is delivered on any link in the IP network, and the copy is only obtained at the IP multicast-enabled media gateway when needed.

The call establishment including connection and communication can be accelerated by using delay reduction techniques as described below. In particular, this technology improves the delay characteristics of group calls in the 1xRTT network, and allows telecommunications companies to provide different levels of PTT services, which are differentiated by changing the delay. For example, the following three levels of services can be provided: gold: the users mobile phone does not enter the dormant state, that is, it is a "always-on" device; silver: the user mobile phone can enter the dormant state, but the user's PPP session is never terminated, that is, the "always-on" device "PPP; and copper: regular service without delay reduction.

In a specific implementation scheme, the system can be implemented by including an appropriate method and system in the mobile phone and an appropriate method and system in the proxy switch. The methods and systems implemented in mobile phones may include methods and systems for user interface enhancement and signal interpretation. FIG. 8 illustrates the components of an exemplary implementation scheme 2010, in which the first and second mobile phones 2012, 2014 pass through the Internet 2020 via the first radio access network (RAN) 2016 and the first packet data service node (PDSN) 2018 and The second PDSN 2022 and the second RAN 2024 communicate with the third and fourth mobile handsets 2026 and 2028. Each RAN has at least one base station (BS), such as BS2030, and at least one base station controller (BSC) 2032. At least one proxy switch 2034 uses SIP explicit signaling to communicate with the PDSN 2018 and 2022 via the Internet. The BSC 2032 communicates with a traditional mobile switching center (MSC) via a proxy switch 2034, such as MSC 2035, which is connected to the PSTN. RAN2016 and 2024 use bearer signals (RP) to communicate with corresponding PDSN2018 and 2022.

As described in the co-pending US applications Serial No. 09/721,329 and 09/845,934, the proxy switch monitors the services passed between the MSC and the BSC, and can rely on the content and conditions of the services to intercept and/or take action.

Each PDSN acts as a router to route packets to and out of the corresponding RAN, and maintain the RP environment, thereby maintaining the session when the mobile phone is roaming. Each PDSN can also perform authentication of data users.

The MSC receives explicit signaling from the mobile phone and uses logic to perform tasks such as processing group call setup requests and managing call control. The MSC also performs mobility management on mobile phones.

The sample system can use one or more of the following delay reduction techniques. The Periodic Presence Information Push (PPIP) technology utilizes the Group Call Register (GCR), which is the database described above and the co-pending US application Serial No. 09/845,934. The GCR contains information about users and their group call lists. In PPIP technology, GCR is also used to maintain the presence information about the user, which is "pushed" to the user's mobile phone. In this way, due to the "presence push", the caller always or almost always knows the presence status of at least some caller group list members (for example, 32 users per group). Therefore, the delay is effectively eliminated, and the caller can make a meaningful conversation while the caller presses the PTT button.

When the MS is turned on, the MS becomes "existing" and completes its registration procedure. When the periodic positioning is updated to the HLR and the response to the paging request is performed in a timely manner, the MS remains present. Otherwise, such as when the MS shuts down or leaves the coverage area of the signal, the MS's registration is eliminated and it is considered "non-existent".

The rate of presence push can be set to generate a manageable network overhead level, and the refresh rate of presence push can be linked with the user's service level. For example, for gold level users, the network can be refreshed every few seconds, while for other users it is refreshed rarely or not at all.

In a specific example, the caller may wish to initiate a group call with members of a football club. In systems lacking PPIP technology, the caller does not know whether the intended recipient exists. In an exemplary implementation of the PPIP technology, the indication of the presence or absence of group members is always displayed in a column at the top of the mobile phone screen. As a result, if at least one group member exists, the caller can press the button and immediately ask "Shall we go to play football?".

By updating the form of information about the existence of group members, PPIP technology can add a large number of services to the network, which can support 5 million or 10 million users. In this way, the different service levels described above can correspond to different update rates and different loads on the network.

In another delay reduction technology called "early stream processing" technology here, the registration phase of the PTT service that appears when the phone is first turned on is also used to initiate media gateway port negotiation (in the co-pending United States to apply for Serial This negotiation is described in No.09/845,934). In this way, pre-negotiation of the ports used by users and user groups for group calls, as part of the registration process, saves time in the process that contributes to the call setup delay. Another aspect of the early stream processing technology is that since the port is pre-identified, any packet (meaning non-silent) can be detected in-band on the port. If service is detected from any group call member, as described in the co-pending US application Serial No.09/845,934, the dialog control (call control) process can be initiated to provide the call of the call User control, which reduces the delay by reducing or eliminating the call control setup time in PTT or group calls.

In the case of delay, the registration procedure is executed when the mobile phone is turned on, and no voice packets are sent until the signaling connection is established based on the registration procedure. However, in the early stream processing delay reduction technology, when the signaling is established, the proxy switch can accept and buffer the voice packets, so there is no need for the caller to wait for the start of the call before the signaling connection is established. When a signaling connection is established, the buffered voice packet can be played back on the receiver's mobile phone immediately.

In group calls and PTT calls, the media gateway port is usually selected and used in the passive data service mode. In the case of delay, port assignment is not performed before the call is established, and port assignment is performed dynamically when the call is established; the port assignment is valid for the duration of the call and the holding time of 2 to 3 minutes, and is for the next call Assign a new port set.

In particular, in the early stream processing technology, the media gateway port was pre-allocated and monitored to assist call control in group calls. In the example of a football club, at any particular moment, one person is the caller and everyone else is the receiver. The person who presses the appropriate phone button has call control; when the button is used to release the call control, another member in the call can take the call control by pressing the corresponding button on the member's mobile phone. If no one presses the button during a certain period of time, the call will go to sleep.

The transfer of control described in the co-pending US application Serial No. 09/845,934 consumed time that contributed to the delay. When the system recognizes that the call control has been abandoned, and when the system transfers the call control to another member. The pre-allocation of ports allows monitoring of ports, whereby call control can be allocated based on the detection of active behavior on specific ports. For example, if the initial voice packet is detected when it is directed to the port corresponding to person A, it can be assumed that the voice packet represents a message equivalent to "Are you there?" and can be the person before the person A presses the button. A assign call control. If the active behavior of the packet is detected on more than one port, a random selection process is performed to assign call control.

Here is another delay reduction technique called "optimized transmission" by compressing the Session Initiation Protocol (SIP) header information for explicit signaling messages, compressing registration information and using short message service (SMS) to deliver registrations Information, at least partially reduces media delay. Therefore, because SMS uses a signaling channel that is not governed by the RP environment, it is not necessary to use a PSD session for delivering SMS services, thereby reducing the delay caused by sleep.

In the specific embodiment of compression, the MS strips unnecessary information from the SIP header information. Other data compression or data reduction methods can also be used.

In particular, SIP can be used for PTT services, and the technology includes reducing the amount of information transmitted by SIP. In addition, the technology can rely on SMS to deliver the information as SMS messages, which reduces the delay, because the SMS that relies on the signaling link never sleeps; the information can be transmitted and received without the need for the mobile phone to perform sleep mode conversion . In particular, to send SIP signaling, SMS is used instead of using the channel associated with the RP environment. The proxy switch receives and interprets the SMS message and acts for SIP signaling accordingly.

Another delay reduction technique referred to herein as a user interface optimization technique reduces the delay by responding to the user's interface conditions. In at least some cases, the user uses the user interface on the mobile phone to find the group before initiating the PTT call. The technology detects that the user's attention is directed to the group on the user interface layer, and as a result, sends a "hint" message to the phone of the potential recipient to initiate the transition from the dormant state to the active state . In at least some cases, SMS can be used to send the hint message.

In a specific embodiment, the user may have multiple group call groups listed on the user interface of the user's mobile phone, for example, a football club group and a solitaire entertainment group. In order to select a group, the user scrolls through the list of groups. If it is determined that the user intends to select a specific group (for example, because the user keeps the cursor on the entry of the group for a period of time), a hint message is sent to the mobile phone of the recipient belonging to the group. In this way, before the user completes the initiation of the group call, the recipient's mobile phone can begin to prepare for the group call.

Another delay reduction technique, referred to herein as the prompt optimization technique, allows the caller to "notify" or "prompt" the expected reception by using the caller's user interface on the caller's mobile phone to send a prompt message to the intended recipient's mobile phone By. In this way, the caller can use the cell phone to help determine whether the intended recipient is available and whether it is willing to receive PTT calls. As a result of the reminder message (which can be sent via SMS), the recipient's mobile phone can also perform the transition from the dormant state to the active state.

In a specific example, a group can be selected from the user's phone book, and the user can press a button to send a prompt message to the phone of the desired recipient to notify the recipient and let him know about the group The call is being initiated. A sound signal can be generated on the mobile phone of each desired recipient to prompt the desired recipient to pick up the mobile phone or make other call preparations.

Figures 9-20 illustrate a sample flow chart of the delay situation and its corresponding process, which can be used in one or more techniques with less delay to speed up call establishment.

Figure 9 illustrates the delay in a registration request (for example, requesting a group call), where mobile phone A (phone 2012 in Figure 8) is turned on and sends a "SIP registration" registration initiation message to the proxy switch, and the proxy switch processes the The registration request, and in response, the proxy switch sends an "ACK" response message.

For the delay situation of Figure 9, Figure 10 shows a delay reduction technology, in which the proxy switch determines the group members of the mobile phone user group call and negotiates port parameters for potential calls with other users mobile phones (such as The co-pending US application Serial No.09/845,934 described in) acts on SIP registration messages.

Figure 11 illustrates such a delay situation, where the user of mobile phone A operates the user interface of the mobile phone to locate and select the group for the group call, mobile phone A sends the first SIP invitation to the proxy switch, and the proxy switch processes the first SIP invitation message And send a second SIP invitation message to mobile phone B, and mobile phone B processes the second invitation message. Mobile phone B sends a first response message to the proxy switch, and the proxy switch sends a second response message to mobile phone A. The mobile phone A sends the first DTP/UDP message to the proxy switch, and the proxy switch sends the second DTP/UDP message to the mobile phone B. If the mobile phone A sends a call control (speaking control) abandon message to the proxy switch, the proxy switch sends a call control available message to the mobile phone B. If mobile phone B sends a call control request to the proxy switch, the proxy switch processes the call control request and other call control requests that may come from other mobile phones, and as a result, sends a call control transfer message to mobile phone B. Mobile phone B then sends a third RTP/UDP message to the proxy switch, and the proxy switch sends a fourth DTP/UDP message to mobile phone A. If one or both of the mobile phones A and B are dormant at the beginning, an additional delay is added due to the transition or transition from the dormant state to the active state.

Figure 12 illustrates such a delay, in which mobile phone A sends a SIP invite message to the proxy switch, and the proxy switch processes the invite message, assigns call control, and sends a response message to mobile phone A.

For the delay situations shown in Figures 11-12, in the delay reduction technology for responding to registration requests as illustrated in Figure 13, mobile phone A sends a registration request to the proxy switch, and the proxy switch processes the registration request and shares the same one or more other registration requests. The proxy switch and PDSN perform port negotiation and send a response message to mobile phone A.

In addition, for the delay situations of Figures 11-12, Figure 14 illustrates such a delay reduction technique, in which mobile phone A sends a SIP invite message to the proxy switch, and the proxy switch processes the SIP message and tries to use the previously negotiated port (as shown in the figure). 13) on the inspection service. If a service is detected, the call control is assigned to the corresponding user, and a response message is sent to mobile phone A (or any mobile phone in the group), which indicates that the call control has been assigned.

Figure 15 illustrates the delay in a "push-to-talk" call (for example, a group call), where A sends a first SIP invitation message to the proxy switch, and the proxy switch sends a second invitation message to mobile phone B and mobile phone C The third invitation message. After receiving the first and second responses from mobile phone A and mobile phone B, the proxy switch receives the first RTP/UDP message from mobile phone A, and sends a second RTP/UDP message to mobile phone B and a third RTP/UDP message to mobile phone C. UDP message. Before the user's call can start, the identification and presence information is established, and the call control exchange is performed.

For the delay situation of FIG. 15, FIG. 16 illustrates such a delay reduction technique, in which mobile phone A has been notified that mobile phone B exists and mobile phone C does not exist. Mobile phone A sends a first SIP invitation message to the proxy switch, and the proxy switch sends a second invitation message to mobile phone B. Mobile phone B sends a first response to the proxy switch, and the proxy switch sends a second response to mobile phone A. The mobile phone A sends the first RTP/UDP message to the proxy switch, and the proxy switch sends the second RTP/UDP message to the mobile phone B. The user's call can start. Since it is pointed out that the mobile phone C does not exist, it is not necessary to send the third SIP message or the third RTP/UDP message to the mobile phone C, and it is not necessary to receive the response from the mobile phone C, which saves time.

Figure 17 illustrates such a delay situation, which is executed as follows: mobile phone A is in a dormant state, mobile phone A performs a transition to an active state, mobile phone A activates the RP environment, and mobile phone A sends a registration message.

Regarding the delay situation of FIG. 17, FIG. 18 illustrates such a delay reduction technique, which is performed as follows: mobile phone A is in a sleep state, and mobile phone A uses SMS to send a registration message while mobile phone A is performing a transition to an active state. (Activating the RP environment is optional and can be performed after the mobile phone A performs the transition to the active state.) Since the mobile phone A can send the registration message before completing the transition to the active state, time is saved.

Figure 19 illustrates such a delay situation, in which the following is performed. The user of mobile phone A scrolls through the entries in the user interface to find the group called by the group, and selects the group called by the group in the user interface. Mobile phone A sends the invitation message to the mobile phone corresponding to the group member. The mobile phone performs the transition from the dormant state to the active state, and responds to the invitation message.

For the delay situation of Fig. 19, Fig. 20 illustrates such a delay reduction technique, in which when the user of mobile phone A scrolls through the list in the user interface to find the group of the group call, the users attention to an item is detected , And for the mobile phone corresponding to the user in the group identified by the entry, determine its presence status information. The mobile phone A sends the invitation message and the prompt message (promoting the transition from the sleep state to the active state) to the mobile phone that is determined to exist, and the mobile phone responds.

Figures 21-28 illustrate the graphs, which show the results compared to systems without delay reduction techniques (non-optimized systems), which rely on one or more of the delay reduction techniques described above. System (optimized system) test results. Figure 21 illustrates that at least for SIP registration transmission time, SIP invitation transmission time, SIP 200 OK, SIP ACK, and SIP INFO for speech control (call control), it is found that the optimized system has a reduced delay and the call originator With 2 seconds of sleep-active conversion. Figures 22-23 illustrate that when the mobile phones of both parties are active at the beginning, at least for call establishment and floor control signaling, it is found that the optimized system has a reduced delay. Figures 24 to 25 illustrate that when the mobile phones of both parties are dormant at the beginning, at least for call establishment and speech control signaling, it is found that the optimized system has a reduced delay. Figure 26 illustrates that at least for SIP registration transmission time, SIP invitation transmission time, SIP 200 OK, SIP ACK, and SIP INFO for speech control (call control), it is found that the optimized system has a reduced delay and the call originator With 4 seconds of sleep-active conversion. Figures 27-28 illustrate that when the mobile phones of both parties are dormant at the beginning, at least for call establishment and speech control signaling, it is found that the optimized system has a reduced delay.

Variations The above embodiments all contribute to the realization of the present invention and speed up call establishment in mobile communications. However, this subset of functions also provides advantages over existing technologies. For example, other call establishment parameters or other call establishment information can be sent on the SMS to avoid the delay caused by the transition from the dormant state to the active state. In another example, one or more delay reduction techniques may be used for full-duplex calls, two-party calls, non-PTT calls, non-group calls, or non-voice calls. In another example, the user interface can be set up so that whenever the user enters the group selection area (for example, menu) of the group call of the user interface, the wake-up information will be sent to the group call link in the group call To the mobile phones of many other users or all other users of the user, that is, to users who are many of the potential recipients of the group call initiated via the group selection area of the users group call People or all of them. The wake-up message can make the mobile phone perform the transition from the sleep state to the active state to reduce the delay. In another embodiment, the availability of one or more delay reduction techniques may depend on the service level of more than one participant in the call to provide the participant with an incentive to request a higher level of service.

In addition, although the embodiment has been described in the context of specific wireless technologies such as TDMA or CDMA protocols, the embodiment can be modified to work with wireless technologies including one or more of the following protocols: TDMA, CDMA , GSM, IS-136 and other 2G and 3G protocols.