摘要:
An apparatus and method for communicating system information in a wireless communication network. A first step 200 includes defining unicast threshold parameter(s). A next step 201 includes receiving a request for system information. A next step 202, 204 includes determining if the system information exceeds the threshold parameter(s). A next step 206-216 includes scheduling an ad-hoc broadcast of the system information if the system information exceeds the threshold parameter(s). A next step 218 includes sending a pointer to the scheduled ad-hoc broadcast. A next step 220 includes broadcasting the network service provider information per the schedule.
摘要:
A method, information processing system, and wireless device are disclosed for managing network scanning intervals. The method includes detecting a new wireless network coverage area (130). At least one local dynamic scanning profile (110) is analyzed in response to the determining. The at least one local dynamic scanning profile (110) is determined to include identification information (306) associated with the new wireless network coverage area (130). A network scanning interval for identifying wireless sub-networks (112) within the new wireless network coverage area is dynamically adjusted in response to the determining that the at least one local dynamic scanning profile 110 includes the identification information (306). The adjustment is based on a scanning interval (312) indicated by the at least one local dynamic scanning profile 110 for the new wireless network coverage area (130).
摘要:
A method (600,700) and apparatus (800) are arranged and operate for facilitating call routing between sites in an enterprise connected by a private IP network (130). A range of identifiers (220-223) served by IP PBXs (120-123) in the enterprise may be registered with a SIP registrar (212) of an inter-site router (210). Call requests to a communication unit (350) outside the home site are routed to the SIP registrar. A site identifier associated with the destination IP PBX allows the call to be completed by the SIP Proxy or the IP PBX where the call was received. When no private IP network is available, a roaming CU registers with a cellular proxy in the visited network. Calls are routed by when an HLR (340) associated with the CU obtains the roaming number from the cellular proxy (330) in the visited site. The MSC completes the call to the visited IP PBX using the roaming number and a local SIP registrar 440 forwards the call to the CU.
摘要:
During (101) a communication session for a plurality of user platforms wherein at least one of the user platforms is on hold and wherein the communication session is presently occurring in a first network and is terminable by a Session Initiation Protocol server as comprises a part of that first network, one establishes (102) in the first network a Session Initiation Protocol instance as corresponds to the communication session wherein the Session Initiation Protocol instance comprises, at least in part, session context information for the user platform that is on hold. Then, following a handoff of bearer support of the communication session from the first network to a second network, one uses (104) the Session Initiation Protocol instance to maintain the hold status of the user platform that is on hold with the Session Initiation Protocol server subsequent to the handoff such that the Session Initiation Protocol server does not terminate the communication session.
摘要:
The present application describes various embodiments that address the need in wireless environments and within the SIP framework to efficiently maintain SIP contact addresses. It introduces the concept of a SIP proxy user agent (UA) (e.g., 123 or 124) to serve as a gateway between a SIP core network and a SIP-unaware mobile (101). A new parameter, called “created,” is described for the contact header in SIP 200 OK messages, which is used to trigger deregistrations by SIP proxy UAs that no longer serve a particular mobile. It is this process for deregistering SIP proxy UAs that allows SIP contact addresses to be more efficiently maintained.
摘要:
During a communication session (101) for a multi-network user platform (which communication session is presently occurring in a first network and is terminable by a Session Initiation Protocol server as comprises a part of that first network), one establishes (102) in the first network a Session Initiation Protocol instance as corresponds to the communication session. Thereafter, and particularly following a handoff of the communication session from the first network to a second network, one uses (104) the Session Initiation Protocol instance to maintain communications with the Session Initiation Protocol server such that the Session Initiation Protocol server does not terminate the communication session.
摘要:
Disclosed is a method that includes initiating a handoff (403) by transmitting a handoff (403) start message to a mobility manager by a first mobile station. The first mobile station is in a call with a second mobile station and moving out from a first communication network to a second communication network. The handoff start message comprises a first call context. The first communication network comprises various network entities such as a SIP server, mobility manager, and the like. The method further includes creating a SIP instance (404) in the mobility manager to hold the first call context. Further, a handoff extension call is placed (405) by the first mobile station to the SIP server via the second communication network and once the handoff extension call is placed, an INVITE message is sent (406) from the SIP server to a preconfigured extension of the mobility manager. The INVITE message comprises a second call context. Thereafter, the handoff extension call is redirected (409) from the second communication network to the first communication network via the mobility manager, in response to the INVITE message received by the preconfigured extension of the mobility manager.
摘要:
A Session Initiation Protocol (SIP)-compliant proxy (106) receives a first INVITE message destined for a mobile station (102). The first INVITE message is associated with an initial Internet Protocol (IP) contact address for the mobile station (102). Subsequently, it is determined whether an IP address update for the mobile station (102) may have occurred and a new IP contact address is obtained for the mobile station (102). A second INVITE message is sent to the mobile station (102) at the new contact address so as to continue a session initiation.
摘要:
Fast call set-up for a call to a multi-mode communication unit (102) is facilitated by a method and apparatus for registering and re-registering priorities with a SIP registrar (111). A list of contacts with first priorities associated with a mode of operation in a first enterprise network (106) is established and used. A switch to second priorities is made when a probability of a switch from the first mode of operation to a second mode of operation in a second cellular network (108) satisfies a threshold. The list of contacts includes a contact having an expiration time for the first mode of operation. A quality factor is determined and the list of contacts is re-registered and the expiration time changed if the quality factor does not satisfy a threshold.
摘要:
A system and method switches between an active call and a call on hold when handing off a mobile station (204) from a Wireless Local Area Network (WLAN) (202). A Handover Starting message is received from a mobile station. The Handover Starting message includes at least one first identifier. A Notification message is also received. The Notification message includes a second identifier. The at least one first identifier is compared to the second identifier and a handover of the mobile station (204) proceeds when a match is determined. Subsequently, the mobile station (204) switches from an active call to a call on hold.