System and method of expediting call establishment in mobile communication

BACKGROUND ART System and method of expediting call establishment in mobile communication

The present invention relates to U.S. Provisional Patent Application No. 60/, filed June 7, 2002, for the title "System and Method of Optimizing Latency Time in Group Calling Systems." 386,883, and the proceeds thereof, incorporated herein by reference.

This invention is a continuation-in-part of U.S. Patent Application Serial No. 09/845,934, filed April 30, 2001, entitled "Systems and Method of Group Calling in Mobile Communications," It is incorporated herein by reference.

1. <u>field of invention</u>

FIELD OF THE INVENTION The present invention relates to mobile communications, and more particularly, to facilitating call establishment in mobile communications.

2. <u>A review of related technologies</u>

As disclosed in co-pending U.S. Patent Application Serial No. 09/845,934, all modern mobile communication systems have a hierarchical arrangement in which geographic "coverage areas" are distributed into many smaller geographic areas called "cells." . Referring to Figure 1, each cell is preferably operated 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. BTSs and BSCs are sometimes collectively referred to as a base station subsystem ("BS") 107 . Several BTSs 106b-n may be aggregated into a mobile switching center ("MSC") 110 via fixed links 108a-n.

MSC 110 operates as a local switching exchange (with additional features for handle mobility management requirements) and communicates with a telephone network ("PSTN") 120 via trunk groups. Under United States mobile networks, the concept of a home MSC and a working MSC exists. A home MSC is an MSC corresponding to an exchange associated with a mobile station ("MS", also referred to as "mobile handset", "mobile telephone handset" or "handset"); This association is based on a phone number, eg the area code of the MS (the home MSC is responsible for the HLR discussed below). A working MSC, on the other hand, is an exchange used to connect MS calls to the PSTN (as subscribers roam in an area covered by a service provider, different MSCs perform the functions of an operating MSC). Consequently, a home MSC and a working MSC are sometimes the same entity, but other times (eg, when the MS roams) they are different. Typically, a visited location register ("VLR") 116 is co-located with the MSC 110 , and logically a single HLR is used in the mobile network. HLR and VLR are used to store many types of subscriber information and profiles.

Briefly, one or more radio channels 112 are associated with an entire coverage area. The radio channels are distributed into groups of channels assigned to individual cells. Channels carry signaling information to establish a paging connection, etc., and are used to carry voice or data information once a paging connection is established.

At a relatively high level of abstraction, mobile network signaling involves at least two main phases. One aspect involves signaling between the MS and the rest of the network. With 2G ("2G" being the industry term used for "next generation") and the latter technology, this signaling is based on the access methods used by the MS (e.g., time division multiple access or TDMA; code-division multiple access or CDMA), allocation of radio channels, authentication, and the like. A second aspect is signaling among various entities in a mobile network; Examples include signaling among MSCs, VLRs, HLRs, and the like. This second part is sometimes referred to in particular as the Mobile Application Part ("MAP") when used in the context of Signaling System No. 7 ("SS7").

Various forms of signaling (as well as data and voice communications) are transmitted and received according to various standards. For example, the Electronics Industries Association ("EIA") and the Telecommunications Industries Association ("TIA") help define many United States standards, such as the MAP standard IS-41. Similarly, CCITT and ITU help define international standards, such as GSM-MAP, an international MAP standard. Information on these standards is well known and can be found in the literature as well as in relevant organized bodies, see, for example, Bosse, Signaling in Telecommunication Networks (Wiley 1998).

To transfer the call from MS 114, the user dials the number onto a cell phone or other MS and presses "Send". MS 114 transmits to MSC 110 via BS 107 the dialed number indicating the requested service. The MSC 110 looks into the associated VLR 116 (below) to determine whether the MS 114 permits the requested service. The MSC in operation routes the call to the dialed-in user's local exchange on the PSTN 120 . The local exchange alerts the called user terminal, and the answer recall signal is routed back to the MS 114 via the working MSC 110, which completes the voice path to the MS. Once the setup is complete, the call can proceed.

To forward the call to MS 114 (assuming the call originates from PSTN 120), the PSTN user dials the MS's associated phone number. At least in accordance with US standards, PSTN 120 routes the call to the MS's home MSC (which may or may not be running the MS). The MSC then queries the HLR 118 to determine which MSC is operating the MS. It also serves to notify the operating MSC that the call is approaching. The home MSC then routes the call to the working MSC. A working MSC pages the MS through the appropriate BS. The MS responds and the appropriate signaling links are set up.

During a call, BS 107 and MS 114 may cooperate to change channels or BTSs 102 if necessary, eg, due to signal conditions.

Mobile communication networks are adding new services to the Internet, for example "data calls". With respect to the Internet, multicast communication relates to the transmission of identical data packets to multiple selected destinations on an Internet Protocol network (by contrast, broadcast communication refers to the differential transmission of data packets of all destination stations, and unicast Communication refers to the transmission of data packets to a single destination).

Each participant in the multicast receives information sent by any other participant in the multicast. Users connected to the network that are not participants in a specific multicast do not receive information transmitted by participants in the multicast. In this way, multicast communication uses only those network components (eg, switches and trunks) that are actually necessary for multicast transmission.

In multicast processing, a potential participant ("host") is directed to join a particular IP multicast group, and the host receives a request to join the multicast group and the nearest multicast- Send a "join request" message to possible routers. For example, host A sends a message to join multicast group, and host B sends a message to join multicast group X. Router R propagates the request to the multicast source if the data fetch is not already in place.

For example, upon receiving an IP packet for group X, router R maps the IP multicast group address into an Ethernet multicast address and forwards the resulting Ethernet packet to the appropriate switch or switches.

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

Regarding the interaction between MSs, the Nextel service with two versions (using specialized mobile radio technology, Nextel Direct Connect® disclosed at http.//www/nextel.com/phone_services/directconnect.shtrol) service is It is proposed for special connection calls between All versions of special connection calls require all members to be located in the same switching area controlled by the BSC/DAP (Dispatch Apply Processor) combination. In a first version, a one-to-one conversation is allowed between two mobile phone subscribers, for example A and B. When A wishes to have special access communication with B, participant B's private identification number presses a push-to-talk ("PTT") button, followed by an audible beep confirming that B is ready to receive and begin to speak. wait A releases the PTT button. If B wishes to issue, B presses the PTT button and waits for an audible confirmation that A is ready to receive. The service allows a subscriber to select private identification numbers from scrollable lists displayed on mobile telephone handsets, or to browse a list of pre-stored names of subscribers.

In the second version, conversation is allowed between members of a given group of subscribers, identified by some number and known as a talkgroup. The mobile telephone handset allows talkgroup numbers to be retrieved through the handset's control surface. To make a group call, an initiating subscriber can initiate a conversation by placing a talkgroup member in the handset, pressing a PTT button, and receiving an audible confirmation such as a chirp. All other talkgroup members on the group call can only listen while A presses the PTT button. When A releases the PTT button, other members on the group call can press the PTT button, acquire control signaled by audible confirmation, and start a conversation.

Among the early embodiments of the group call system, the transmitting (broadcasting) transceiver turns on its transmitter, and the receiving transceivers turn off their transmitters and those turned on during Nextel Diret Connect®. There exists a two-way talk radio (TWTR) system, i.e. a similar half-duplex radio system, in which the receivers are turned off while having them. The delay in TWTR systems is substantially zero, which is dominated by the speed of radio waveforms and the propagation times of electronic components. Another characteristic of such systems is that the broadcast caller has no prior knowledge of the presence of listeners. This is only when at least one of the listeners responds that the caller can confirm the presence of any listeners. Thus, the typical mode of a group call is when establishing a group call, for example, "Are you there?" Includes "human protocols" in which the caller first confirms the presence of one or more listeners using phrases such as . If no meaningful communication can occur in the group call before the presence of listeners is ascertained, the delay, called the Human Round Trip Response Time (HRTRT), indicates the perceived delay of the TWTR. In at least some cases, when the handset initially accesses the called party, the HRTRT ranges from 1.5 to 4 seconds, as opposed to the delay due to the speed of wireless waveforms, which can be 0.03 milliseconds over a five-mile distance.

In some PTT systems, digital radios are used for coded and framed half duplex communication. Unlike TWTR systems, digital radio base PTT systems use explicit signaling to establish group calls. Due to explicit signaling and group call setup activity, coding and digital framing of the raw analog voice signal, and transmission delays, such systems have significant delays and, in at least some cases, may range from 750 milliseconds to 1.5 seconds. Also, digital radio-based PTT systems differ from TWTR systems in which the caller is aware of the presence of a listener. Typically, digital radio-based PTT systems play a sound such as a "chirp" to indicate one or more listeners after the caller can proceed by calling. Thus, HRTRT delay is left relevant in digital radio-based PTT systems because the caller needs to know that the listener is valid and engaged. The chirping only indicates that the handset is available, which provides no indication of the listener's status. The caller does not know whether the listener is busy with something or whether the handset is at some distance from the listener, for example on a kitchen counter a few feet away from the listener. In at least some cases, HRTRT in current digital base PTT systems may range from 2 to 5 seconds when the handset is easily accessed by the listener.

In some implementations of digital radio base PTT systems using standard air interfaces (RF modulation), such as the CDMA 1xRTT interface, the HRTRT may be a logic of 12-15 seconds. These interfaces are not optimized for PTT style group calls and introduce various delays when used to deliver PTT calls. A typical PTT call in 1xRTT networks can have an HTRT delay of 15 seconds, which can create a serious impediment to the successful deployment of new PTT systems.

The overall delay includes at least the following factors: As noted above, the delay shown is the delay due to time spent as the caller decides that the called party exists and can thus initiate a conversation. The current delay occurs once, when the caller initiates a group call. Call setup delays occur once out of the group call. Medium delay is the delay due to the time spent before the parties in the call hear the conversation spoken by one party in a group call, and includes buffering time, coding time, and transmission delays of the voice medium. As mentioned above, HRTRT is the delay due to the time the caller spends before the caller hears the called party's voice, i.e. after the caller speaks, releases the control button and the called party listens, gains control, and speaks. .

The conventional 1xRTT PTT service is a transport mechanism with RTP/UDP/IP by voice coded as EVRC (Enhanced Variable Rate Codec) and SIP (Session Initiation Protocol) as an explicit signaling protocol, etc. Packet switched data (PSD) use. In 1xRTT networks, the handset enters a dormant state when there is no packet data activity for a period known as a dormant period, which is a network configurable parameter. When data activation for the dormant handset begins, the handset performs a transition from the dormant state to the active state. Thus, if a participant in a group call has a handset that is dormant, the time spent as the handset progresses from the dormant state to the active state also contributes to the overall delay in the group call. In at least some cases, the average call setup delay (including the current delay) may range from 1.5 seconds to 3 seconds for participants with active handsets, and range from 5 seconds to 10 seconds for participants with dormant handsets can be In at least some cases, the average media delay may range from 400 milliseconds to 600 milliseconds, the HRTRT may range from 5 seconds to 7 seconds for participants with active handsets, and for participants with dormant handsets It may range from 7 seconds to 14 seconds.

Another aspect of a typical implementation of 1xRTT networks is the RP node when the "RP context" implementation feature lacks the "RP context" implementation feature, ie, activity for a period as the PPP session associated with the handset is terminated by the network. The lack of RP context also contributes to the delay of group calls in typical 1xRTT networks.

In case of lack of activity for a period, according to the dormant characteristics of the 1xRTT network, the PPP session is maintained, but air resources are released for other uses. When data has to be transmitted, air resources (ie, "handset wake up") consume time contributing to the delay.

SUMMARY OF THE INVENTION The present invention generally provides mobile communication systems and methods, and more particularly, provides a system and method for facilitating call establishment in mobile communications in facilitating conversation calls and group calls. While the mobile station MS is dormant, the mobile station is prepared for half duplex mobile communication telephone calls. In response to user initiation of the half-duplex mobile communication telephone call, the half-duplex mobile communication telephone call is established based on the manufacture of the mobile station.

By facilitating call establishment, the mobile communication system can provide users with a PTT system or group call system with virtually no delay. The provider can allocate network resources efficiently according to economic stimuli to effectively reduce the delay. Users can communicate quickly, accurately, and cost-effectively by knowing in advance the effectiveness of other users.

1 is a system diagram of prior art mobile networks;

2 illustrates a block diagram of a system including a group call or push to dialog logic;

3-4 illustrate proxy switches and specific deployments in a mobile network;

5-6, 8 illustrate the architecture of a group or push for a conversational communication system;

7, 9-20 are call flow diagrams of the use of group or push for a conversational communication system;

21-28 are charts showing test results of delay reduction techniques.

Co-pending US patent application Ser. No. 09/845,934 discloses a system and method for arranging calls between members of a predetermined group of mobile phone users. 2 , as disclosed in co-pending U.S. Patent Application Serial No. 09/845,934, a proxy switch or other device implementing group call logic 1010 initiates a group call by member 1012A of group 1014. , and automatically attempts to contact all members 1012A, 1012B, and 1012C of the group in the group call. In a specific implementation, communication within a group call is half duplex (ie, only one member can speak at a time), and voice traffic for the group is carried over an Internet Protocol ("IP") network in a multicast session.

Regarding the case where the group call logic is implemented by a proxy switch, the proxy switch was introduced in 2000 under the heading "System and Method of Serving Mobile Communications with a Proxy Switch". may operate as disclosed in co-pending U.S. Patent Application Serial No. 09/721,329, filed November 22, which is incorporated herein by reference. As disclosed in co-pending U.S. Patent Application Serial No. 09/721,329, and illustrated in FIG. 3, switching 1034 operations are performed between at least one mobile switching center ("MSC") 1030 and at least one base station subsystem. ("BS") 1032 . Switching allows communication traffic to be absorbed to or from an alternate network 1036 , such as an IP network. The switching is transparent so that neither the MSC nor the BS need to trigger any changes by progressive switching.

The proxy switch disclosed in co-pending US Patent Application Serial No. 09/721,329 includes signaling message handling logic 1038 to receive signaling messages from the MSC and the BS in accordance with a mobile signaling protocol. The message blocking logic 1040 cooperates with the signaling message handling logic, and sends an acknowledgment message to the MSC or BS that sent the signaling message. The message blocking logic also prevents signaling messages from proceeding towards each other of the BS and MSC. The message transformation logic 1042 cooperates with the signaling message handling logic and converts a signaling message from one of the MSC and BS to a signaling message that is transformed for transmission to each of the other of the BS and MSC. Message transfer logic 1044 cooperates with signaling message handling logic and sends signaling messages from one of the MSC and BS to each of the other of the BS and MSC.

A set of carrier circuits 1046 from the BS is assigned to the proxy switch. The signaling messages between the MSC and the BS are received and used to determine whether they correspond to an assigned set of carrier circuits. If so, the control information in the signaling messages is conveyed to the alternate communication network; Information conveyed on the set of carrier circuits is absorbed into the alternate network.

4 shows one preferred deployment of proxy switch 300 in which proxy switch 300 is disposed between BS 107 and MSC 110 . Only one of the trunks 306 carrying user traffic needs to be terminated on the proxy switch; The remaining trunks 308 may directly connect MSC 110 and BS 107 . All control links 312 from BS 107 terminate at proxy switch 300 . The proxy switch includes a control plane 302 and a data plane 304 (also known as "carrier plane"). Control plane 302 handles all signaling traffic, and data plane 304 handles all user traffic for trunks connected to the proxy switch.

Under certain embodiments, there is a one-to-one correspondence trick between the MSC and the proxy switch. Multiple BSs can work with a single proxy switch.

The proxy switch 300 comprises software that accepts all signaling messages, depending on the message and the state of the system, as follows:

One. pass messages unrelated to the MSC or BS addressed in the message;

2. blocking messages between MSC and BS;

3. For some blocked messages, convert the blocked messages into different messages, and send the converted message to the MSC or BS addressed to the blocked message instead of the original blocked message;

4. at least one of absorbing messages from the mobile- and PSTN-based networks to an alternate network, such as an IP network.

The type of actions performed in each case according to the triggering events is described below.

In many cases, proxy switch 300 may act as MSC 110 , particularly when messages are absorbed from MS 114 and traffic is directed to an alternate network. In such a role, the proxy switch fulfills its responsibilities and serves as a traditional MSC. Some of those functions and roles relate to mobility management. Considering the case of a roaming MS; As it roams from one cell to another, it may roam to a cell operated by a different MSC, thus necessitating a handoff between the source and target MSCs. When the proxy switch 300 absorbs the message and the call/session is directed to an alternate network, the handoff is managed by the proxy switch similar to how this handoff could be managed by a conventional MSC. The proxy switch allows the appropriate databases to be updated with the new locations of the MS.

Another function of the proxy switch concerns the allocation of resources. In particular, when the MS initiates a message requesting a new call/session, the appropriate circuits (channels) need to be allocated for this session. Depending on the configuration of the system and the state of the system, the proxy switch makes such assignments similar to how a conventional MSC assigns circuits.

Figure 5 shows a typical deployment in which the proxy switch 300 is connected to various alternative networks, such as an IP framework 412 or an alternative circuit-based network 414, for example a different carrier. These alternative networks may be used to deliver voice and/or data traffic to desired destinations while avoiding the PSTN 120 in whole or in part along with the costly resources of the MSC 110 . Alternatively, these arrangements can be used so that circuit traffic can be carried back to circuit traffic from a different network, for example Nashua. NH can be delivered otically to MSCs in Waltham MA. Alternatively, they may be used to connect to other networks. For example, IP framework 412 may communicate with IP voice network 418 or Internet 416 . As described in the co-pending application, when absorbing traffic to an alternate network, both control information (eg, from signaling messages) and voice or data from the carrier circuits on link 306 can be transmitted over the alternate network. have.

In a particular implementation of the group communication system disclosed in co-pending U.S. Patent Application Serial No. 09/845,934, mobile communication users ("users") belonging to a closed user group ("group" or "CUG") communicate quickly with each other and They have the ability to make contact easily, and thus initiate conversations with each other. Each group contains two or more users ("members"), and a user may belong to multiple CUGs. A conversation may take place between two members of a group ("private mode") or between all valid members of a CUG ("public mode"). Group communication systems use conventional mobile communication equipment such as cellular telephones and mobile PDAs.

In a specific implementation, the group communication system blocks group call initiation, bypasses MSCs and PSTN, and implements group call, such as IP multicast sessions, which conduct voice over IP ("VoIP"). Implement group call logic in proxy switches logically placed between MSCs and BSCs as shown. Users in the group are not at all affected by multiple MSCs via an aggregate network that rely on one or more of the following radio technologies: CDMA, TDMA (including IS-136 and GSM), GPRS, and 3rd generation technologies. Service may be provided in separate geographic locations. For example, among group members joining any one group call, one or more users may roam in a GSM network concurrently with one or more users roaming in a CDMA network. Control information pertaining to a group call may be valid for one or more users during the group call, such as display participants in the group call. Group call lists can be dynamically created and modified by the group call user using standard numbering schemes such as MIN, IMSI and ESN.

A general architecture for one embodiment of a group communication system is shown by way of example in FIG. 6 . 6 shows four users in a group call using wireless devices 1060A-1060D connected to different BTS systems 1062A-1062D. For the purposes of the following description, it is assumed that wireless devices have both audio and text display capabilities. BTSs are connected to base station controllers ("BSCs") 1064A-1064D, which are connected to proxy switches implementing group call logic ("group call switches") 1066A-1066C. Each group call switch is connected to an MSC, such as MSC 1068A, 1068B, or 1068C. At least one group call switch is provided for every MSC in a group call service enabled network. Regarding the signaling information, each group call switch is logically placed between the corresponding BSC and the corresponding MSC. The group call switch receives signaling and data via the BTS and BSC from the MSC and from wireless devices in the reverse direction. Each group call switch operates so that neither the BSC nor the MSC is aware of the group call switch placed 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 passed seamlessly over the relevant elements as needed without any appreciable change.

MSCs are connected to a common mobile network ("PLMN") 1070 , group call switches are connected to a skeleton multicast-enabled IP network ("skeleton network") 1072 , CUG Active Directory 1074 and promotion A home location register ("HLR") 1076 is provided.

As described above with respect to the proxy switch of the co-pending application, the group call switch includes a control plane and a data plane. The function in the control plane is the termination of signaling messages from the BSC or MSC or both. For example, in CDMA networks, signaling messages are defined by the IS-634 protocol specification. The control plane terminates incoming signals and generates a new signaling message to proceed with transmission to the MSC or other devices. The control plane also supports the multicast function described below.

In one particular aspect, the data plane of the group call switch receives TDM traffic from the BSC or MSC or both, and establishes a TDM cross connection ("DACS") (FIG. 4) to interface the incoming traffic to the outgoing destination. use. In another embodiment, the data plane may also receive incoming IP traffic from a base station complex (also known as a radio access network or "RAN"), and switches the incoming IP traffic to outgoing IP traffic. Programmatic control in the control plane determines the cross connections between the incoming TDM traffic and the outgoing destination, in particular the traditional MSC and/or destinations on the IP network.

In the case of an MSC acting as a destination originating from a DACS, the group call switch is inherently transparent to the network; Traffic and control flow seamlessly from BSC to MSC and from MSC to BSC. When the outgoing destination is taken over on the IP network, a media gateway in the data plane (disclosed in a co-pending patent application) diverts selected portions of the incoming TDM traffic from the MSC and redirects the incoming TDM traffic to RTP/UD/ Converts to IP and inserts RTP/UD/IP traffic into the backbone IP network.

The CUG Active Directory ("CUG AD") 1074, also known as the Group Call Register ("GCR"), is a database system that contains CUG data. In a particular implementation, the CUG AD of FIG. 7 is implemented as a distributed database system for scalability. CUG AD contains the definitions of all CUGs in the group call network. A query for a CUG AD specifies the identifier of the CUG, ie, a query is made for the definition of the specified CUG, and the result is a list of group user IDs for all members of the specified CUG. For example, a query specifying CUG ID 2347 may result in identifying mobile identification numbers ("MINs") xxx, yyy, zzz and www for four users in the CUG. In a particular implementation, MIN numbers are assigned to users of the GIR service by the service provider.

Each CUG is identified to the system by a unique identifier ID derived from the distributed CUG signature space such that different partitions are assigned to different, distributed portions of the CUG AD. The distribution index of the distribution scheme is valid for all group call switches. When the group call switch needs to retrieve the definition of the CUG, the group call switch may use that index to determine which component of the CUG AD to be queried.

In a particular implementation disclosed in co-pending U.S. Patent Application Serial No. 09/845,934, the group call service operates over IP networks using IP multicast. IP multicast allows sending a single copy of a stream of VoIP packets received by multiple recipients explicitly registered to retrieve the stream to the source. Multicast is a receiver-based concept that allows receivers to join a specific multicast session group and a stream is delivered by the network infrastructure to all members of that group. When only one copy of the multicast stream passes over any link in the IP network, copies are made only at IP multicast enabled media gateways as needed.

Call establishment, including connections and communications, may be facilitated by using delay reduction techniques as described below. In particular, these techniques improve the delay characteristics of group calls (including PTT calls) in xRTT networks and allow a carrier to provide distinct different classes of PTT services by varying the degree of delay. For example, the following three classes of services:

Gold: the user's handset does not enter a dormant state, ie, the device state is "always on";

Silver: The user's handset may enter a dormant state, but the user's PPP session is not terminated, ie, the "always on" PPP state;

Dong: A regular service with no reduced delay can be provided.

In a particular implementation, the system may be implemented by incorporating the appropriate methods and systems into handsets and by incorporating the appropriate methods and systems into a proxy switch. Methods and systems implemented among the handsets may include user interface augmentation and signal interpretation methods and systems. FIG. 8 shows that first and second mobile handsets 2012 and 2014 are connected to a first data operations node (PDSN) 2018 and a first data operation node (PDSN) 2018 via a first radio access network (RAN) 2016 and via the Internet 2020. 2 illustrates the components of an implementation 2010 that communicates the PDSN 2022 and the second RAN 2024 with the third and fourth mobile handsets 2026 , 2028 . At least one proxy switch 2034 communicates with the PDSNs 2018, 2022 over the Internet using SIP explicit signaling. The ESC 2032 communicates with a legacy mobile switching center (MSC), such as an MSC 2035 , that connects to the PSTN via a proxy switch 2032 . RNAs (2016, 2024) communicate with corresponding PDSNs (2018, 2022) using carrier signals (RP).

As disclosed in co-pending US patent applications 09/721,329 and 09/845,934, a proxy switch monitors the traffic passing between the MSC and the BSC and, depending on the traffic content or condition, blocks the traffic and (or) for action.

Each PDSN acts as a router to route packets to and from the corresponding RAN and maintains RP contexts such that the session is maintained as the handset roams. Each PDSN may perform authentication of the data subscriber.

MSC receivers receive explicit signaling from the mobile handset, process group call setup requests, and use logic performing tasks such as managing conversation control. The MSC also performs mobility management for the handset.

A simple system may use one or more of the following delay reduction techniques.

Periodic Presence Information Push (PPIP) technology enables the use of the Group Call Register (GCR), a database as disclosed in co-pending U.S. Patent Application Serial No. 09/845,934, as described above. The GCR contains information about subscribers and their group call lists. In PPIP technology, GCR may be used to maintain presence information about subscribers, where presence information is "pushed" to the subscribers' handsets. Thus, with "presence push", the caller is constantly or nearly constantly aware of the presence of at least some of the caller's group list members (eg, 32 users per group). Thus, the current delay is effectively eliminated, and the caller can meaningfully transition as soon as the caller presses the PTT button.

As the MS comes to "exist" when turned on, it completes its registration process. The MS remains current as long as the periodic location updates to the HLR and responses to paging requests are executed in time. Otherwise, when the MS is turned off or is outside the signal coverage area, the MS is deregistered and considered "not present".

The current rate of pushes may be configured to produce a manageable level of network overhead, and the current rate of refreshes of pushes may be daily for the subscriber's class of service. For example, the network can replay per second for Gold class subscribers, and often little or no for all other subscribers.

In certain embodiments, the caller may wish to group calls to members of a soccer club. In a system lacking PPIP technology, the caller does not know whether the intended returns exist. In one embodiment implementation of the PPIP technology, an indication of whether group members are present is constantly displayed directly at the top of the handset's screen. Consequently, if at least one group member is present, the caller presses a button and immediately asks "Can we play soccer?"

PPIP technology can add significant traffic on a network that can support 5 million or 10 million users, in the form of update information regarding the presence of group members. Thus, different classes of service as described above may correspond to different update rates and different burdens on the network.

In another latency reduction technique, referred to as the "initial streaming" technique, the registration period of the PTT service, which occurs when the handset is first powered up, is also the media gateway port negotiation (this negotiation is a co-pending US patent). Application No. 09/845,934). Thus, the ports used by a subscriber and a group of subscribers for group calls are pre-negotiated as part of the registration process, saving time spent in this process contributing to call setup delays. Another aspect of early streaming technology is that because pods are identified in advance, any packets (confirming that they are not silent) can be detected in-band on the ports. When traffic is thereto detected from any members of the group call, a speech control ("conversation control") process is initiated as described in co-pending U.S. Patent Application Serial No. 09/845,934 to provide caller control of the call; This reduces latency by reducing or eliminating dialog control setup time in PTT or group calls.

In the delay situation, the registration procedure is executed when the handset is turned on, and voice packets are not transmitted until a signaling connection is not established based on the negotiation procedure. In the initial streaming delay reduction technique, voice packets can be accepted and buffered by the proxy switch while signaling is set up, so that the caller is not asked to wait until a signaling connection is made to initiate a conversation. The buffered voice packets can then be played back onto the recipient's handset as soon as the signaling connection is established.

Media gateway ports are typically selected and used in passive data service mode in group calls and PTT calls. In a delay situation, no port assignment is performed until a call is made, at which point port assignment is performed dynamically; The port assignment is valid for the duration of the call, waits 2-3 minutes, and the next call is assigned to a new set of ports.

In particular, in early streaming technology, media gateway ports are pre-assigned and monitored to aid in call control within a group call. In the example of a football club, at any particular point in time, one person is the caller and all others are the recipients. a person pressing the appropriate handset button controls the conversation; When conversation control is relaxed using a button, other members on the call can assume conversation control by pressing a corresponding button on the member's handset. If no one has pressed the button within the period, the call goes dormant.

The transition of control described in co-pending U.S. Patent Application Serial No. 09/845,934 consumes time that contributes to delay. Time is wasted as the system recognizes that control of the conversation has been relinquished and as the system grants control of the conversation to another member. By allowing preliminary assignment of ports to be monitored, call control can be assigned based on detection of activity for a particular port. For example, if initial voice packets are detected as being forwarded to the port corresponding to person A, those voice packets display a message corresponding to "Are you there?" can be assigned to A. If packet activity is detected for more than one port, a random selection process may be performed to assign dialogue control.

Another delay reduction technique, referred to herein as a "best delivery" technique, is to compress Session Initiation Protocol (SIP) headers used in explicit signaling messages, compress registration information, and shorten the messageization service (SMS) to convey registration information. ), at least in part, reducing the media delay. Thus, delay due to dormancy is reduced because PSD sessions do not need to be validated to carry SMS traffic, and thereafter SMS uses signaling channels that do not apply to RP contexts.

In a particular embodiment of compression, the MS strips unnecessary information from the SIP headers. Other methods of data compression or data reduction may be used instead or likewise.

In particular, SIP may be used for PTT service, and the technique includes reducing the amount of information transferred by SIP. In addition, the technique can rely on SMS to convey information such as SMS messages, which reduces delay because SMS, which relies on signaling links, is put to sleep, and the information triggers a transition from dormant mode. can be transmitted and received without requiring handsets for In particular, to transmit SIP signaling, SMS is used instead of using channels associated with the RP context. The proxy switch receives and interprets SMS messages and thus acts on SIP signaling.

Another latency period reduction technique presented herein as a user interface optimization technique reduces latency by responding to user interface conditions. In at least some cases, the subscriber uses a user interface on the handset to place the group before initiating a PTT call. This technique detects that the user's attention has been directed to the group at the user interface level, and consequently an "initialization" message is sent to the potential recipient's handsets to initiate the transition from the dormant state to the active states. In at least some cases, SMS may be used to send an initiation message.

In certain embodiments, a user may have multiple group calling groups listed on the user's handset user interface, such as a soccer clock group and a card game group. To select a group, the user scrolls down through the list of groups. If it is determined that the user intends to select a particular group (eg, because the user causes the cursor to waste on the group's listing for a period of time), an initiation message is sent to the recipient handsets belonging to that group. Thus, the recipient handsets can start preparing for the group call before the user completes the initiation of the group call.

Another delay technique reduction technique, referred to herein as an alert tone optimization technique, is that the caller "pings" or " Allow "beep". Thus, the caller can use the handset to assist in determining whether the intended recipient is valid and willing to receive a PTT call. As a result of an alert message that may be sent via SMS, the recipient's handset may initiate a transition from a dormant state to an active state.

In certain embodiments, a group may be selected from the user's phone book, and an alert message may be sent to the intended recipient's handsets to alert the intended recipients that the user is aware that a group call is being initiated. button can be pressed. Each of the intended recipients' handsets may generate an audible signal to inspire the intended recipients to pick up the handsets or otherwise prepare the call.

9-20 illustrate simple flow diagrams of delay conditions and corresponding procedures that may be used in one or more delay reduction techniques to facilitate call establishment.

Fig. 9 shows that mobile handset A (handset 2012 in Fig. 8) is turned on, issues a "SIP Register" registration initiation message to the proxy switch, the registration request is processed by the proxy switch, and the proxy switch acknowledges "ACK" Illustrated a delay situation (eg for group calls) in a registration request that is answered by a message.

With respect to the delay situation of FIG. 9 , FIG. 10 determines members of a handset user's group call group and directs potential calls by other user's handsets (eg, handsets 2014 , 2026 , 2028 in FIG. 9 ). It illustrates a technique for reducing delay in responding to a SIP register message by negotiating port parameters (disclosed in co-pending U.S. Patent Application Serial No. 09/845,934) to be used for

11 shows a user of handset A operates a user interface of the handset to place and select a group call group, handset A sends a first SIP invitation message to a proxy switch, and the proxy switch processes the first SIP invitation message; and sends a second SIP invite message to handset B, which illustrates a situation of delay in processing the second SIP invite message. Handset B sends a first response message to the proxy switch which sends a second response message to handset A. Handset A sends a first RTP/UDP message to a proxy switch that sends a second RTP/UDP message to handset B. When handset A issues a conversation control ("floor control") requirement message to the proxy switch, the proxy switch sends a conversation control valid message to handset B. When handset B sends a conversation control request to the proxy switch, the proxy switch processes the conversation control request along with any other conversation control requests that may come from other handsets and, consequently, sends a conversation control acknowledgment message to handset B. can be sent to Then, handset B sends a third RTP/UDP message to the proxy switch which sends a fourth RTP/UDP message to handset A. When one or both of handsets A and B are initially dormant, an additional delay is added due to the transition or transitions from the dormant state to the active state.

12 illustrates a delay situation in which handset A sends a SIP invitation message to a proxy switch, where it processes the invitation message, assigns conversation control, and sends a recognition message to handset A;

With respect to the delay situations of FIGS. 11-12 , in the technique for reducing the delay period responding to a registration request as illustrated in FIG. 13 , handset A sends a registration request to a proxy switch, where it processes the registration request and one or more It performs port negotiation with other proxy switches and PDSNs and sends an acknowledgment message to handset A.

Furthermore, with respect to the delay situation of Figs. 11-12, Fig. 14 shows that handset A sends a SIP invite message to the proxy switch, and the proxy switch processes the SIP invite message, and negotiated previously as illustrated in Fig. 13 . It illustrates a delay reduction technique that attempts to detect traffic on ports that have been used. When such traffic is detected, conversation control is assigned to the corresponding user, and a recognition message is sent to handset A (or any handsets in the group) indicating that conversation control has been assigned.

15 shows a conversation call (eg, group call) in which handset A sends a first SIP invite message to a proxy switch, where it sends a second SIP invite message to handset B and a third SIP invite message to handset C; A delay situation is illustrated by a push to . After receiving the first and second responses from handset A and handset B, the proxy switch receives a first RTP/UDP message from handset A, sends a second RTP/UDP message to handset B, and sends a third RTP/UDP message to handset B. Send a UDP message to handset C. Identification and presence information is established, and a conversation control exchange is executed before a user conversation is initiated.

With respect to the delay situation of Figure 15, Figure 16 illustrates a delay reduction technique in which handset A is provided with information that handset B is present and handset C is not. Handset A sends a first SIP invite message to the proxy switch, which sends a second SIP invite message to handset B. Handset B sends a first response to the proxy switch, which sends a second response to handset A. Handset A sends a first RTP/UDP message to the proxy switch, which sends a second RTP/UDP message to handset B. A user's conversation may be initiated. Since handset C can be indicated as not present, there is no need to send a third RTP/UDP message to handset C and there is no need to receive a response from handset C, which saves time.

Figure 17 illustrates a delay situation in which the sequence is executed as follows: handset A is dormant, handset A executes a transition to active state, handset A has an active RP context, handset A has a registration message to send

With respect to the delay situation of Fig. 17, Fig. 18 illustrates a delay reduction technique in which the sequence is executed as follows: handset A is dormant, handset A is in parallel with handset A sending a registration message using SMS; Execute transition to active state (activating the RP context is optional and may be selected and made after handset A executes transition to active state). This saves time because handset A can send a registration message before completing the transition to the active state.

Figure 19 illustrates a delay situation in which the sequence is executed as follows: the user of handset A scrolls the listing in the user interface to find the group call group, and selects the group call group in the user interface. Handset A causes invitation messages to be sent to handsets corresponding to members of the group. Handsets make transitions from a dormant state to an active state and respond to invitation messages.

With respect to the delay situation of Fig. 19, Fig. 20 shows that the user of handset A scrolls a list in the user interface to find a group calling group, the user's focus on the listing is detected, and the presence status information of the group identified by the listing. determined for the handsets corresponding to the users in Handset A causes invitation messages and warning messages (inciting transitions from dormant to active state) to be sent to the handsets determined to be present, and the handsets respond by sending a response.

21-28 are charts showing test results comparing the results of a system relying on one or more of the delay reduction techniques ("optimal system") and a system lacking delay reduction techniques ("non-optimal system"). exemplify Figure 21 shows that the optimal system delays at least with respect to SIP Register Transmit Time, SIP Invite Transmit Time, SIP 200 OK, SIP ACK and SIP INFO and a 2 second sleep for activating a transitional call initiator for floor control (conversation control). What has been found to reduce time is exemplified. 22-23 illustrate that the optimal system has been found to have at least reduced delays with respect to call setup and floor control signaling when both parties' handsets are initially activated. Figures 24-25 illustrate that the optimal system has been found to have at least reduced delays with respect to call setup and floor control signaling when both parties' handsets are dormant. 26 shows that the optimal system delays at least with respect to SIP Register Transmit Time, SIP Invite Transmit Time, SIP 200 OK, SIP ACK and SIP INFO and 4 second sleep for activating the transition call initiator for floor control (conversation control). What has been found to reduce time is exemplified. 27-28 illustrate that the optimal system has been found to have at least reduced delays with respect to call setup and floor control signaling when both parties' handsets are initially activated.

<u>changes</u>

All of the above embodiments encourage the realization of progressive facilitation of call establishment in mobile communications. However, subsets of functionality still provide advantages over the prior art. For example, other call establishment parameters or other call setup information may be sent over the SMS to avoid delays that result in the transition from dormant state to active state. In another embodiment, one or more delay reduction techniques may be used for a full duplex call, a two-way call, a non-PTT call, a non-group call, or a non-voice call. In another embodiment, the user interface is only when the user enters a group invocation group selection area (eg, menu) of the user interface. Wake up messages are sent to the handsets of many or all other users connected to the user of the group call groups, ie to the handsets of many or all users that are potential recipients of group calls originating through the user's group call group selection area. Wake-up messages can cause handsets to make a transition from a dormant state to an active state to reduce delay. In another embodiment, the effectiveness of one or more delay reduction techniques may depend on the classes of services of one or more participants in a call, which provide a stimulus to the participants to obtain a more advanced class of service.

Further, to the extent of the embodiments described in the context of specific radio technologies, such as the TDMA or CDMA protocol, these embodiments may include: one of the following: TDMA, CDMA, GSM, IS-136 and other 2G and 3G protocols. It may be modified to work with wireless technologies including more than one.