摘要:
Novel tools and techniques are provided for implementing originating number or address-based route determination and routing. In various embodiments, a computing system may receive, from a first router among a plurality of routers in a first network operated by a first service provider, first SIP data, the first SIP data indicating a request to initiate a SIP-based media communication session between a calling party at an originating address in an originating network and a called party at a terminating address in a terminating network. The computing system may determine a communication route among a plurality of routes through the plurality of routers in the first network for establishing the SIP-based media communication session, based at least in part on the originating address, and may establish the SIP-based media communication session between the calling party and the called party via the determined communication route.
摘要:
A system and method to facilitate communication between telecommunications participants in a telecommunications network is disclosed. In one aspect, a method of determining a time to permit a communication session by telecommunications participants to be conducted includes causing a telecommunications processor to retrieve, from a data storage: a free time value that is representative of a free time attributed to a participant in the communications session, a funds balance held by the participant, a pre-stored cost per unit time value, and a representation of a billing pattern for the participant. The method also includes causing the telecommunications processor to determine a maximum time to permit the communication session to be conducted as a function of the free time value, the funds balance, the cost per unit time value and the billing pattern. The telecommunications processor initiates ending the communication session when the time to permit the communication session to be conducted expires.
摘要:
The present invention provides for routing calls between disparate domains, such as a circuit-switched subsystem and a multimedia subsystem. When a user element is homed in a first domain and roaming in a second domain, an incoming call will arrive at a gateway node in the first domain. As a result, a message identifying the user element and indicating that an incoming call has been received at first gateway node for the first domain is sent to a continuity control function (CCF) residing in the multimedia subsystem. The CCF will create and effect delivery of an inter-domain routing number to the first gateway node. The inter-domain routing number is used by the first gateway node to route the call to the second domain. The inter-domain routing number may be associated with a second gateway node of the second domain.
摘要:
A process and apparatus to facilitate communication between callers and callees in a system comprising a plurality of nodes with which callers and callees are associated is disclosed. In response to initiation of a call by a calling subscriber, a caller identifier and a callee identifier are received. Call classification criteria associated with the caller identifier are used to classify the call as a public network call or a private network call. A routing message identifying an address, on the private network, associated with the callee is produced when the call is classified as a private network call and a routing message identifying a gateway to the public network is produced when the call is classified as a public network call.
摘要:
An integrated base station provides uplink and downlink wireless connectivity between user equipment and a wireless communication network. In order to accommodate certain features of the network, a bypass switch of the integrated base station is employed so that the baseband module of the base station is bypassed as to those features. In this case, selected data obtained from the RF transceiver of the integrated base station is processed by equipment in the back end system instead of by the baseband module. A first operation mode enables the baseband module to process the selected data, while a second operation mode bypasses the baseband module so that the back end system performs the processing. The bypass switch may be permanently activated. The bypass process allows for processing of the selected data in an efficient manner without requiring replacement of the integrated base station, thereby providing an efficient and cost-effective solution.
摘要:
A communication system includes a communication apparatus that processes a packet according to control information set by a control apparatus; a first control apparatus that controls the communication apparatus by setting the control information in the communication apparatus; and a second control apparatus that operates in concert with the first control apparatus. The first control apparatus transmits information necessary for interoperation between the first and second control apparatuses to the second control apparatus via the communication apparatus.
摘要:
Telephone call routing in networks is provided by forwarding routing data other than origination identification and destination identification in-band with calls, and using the in-band data at call destinations to do further routing. In some embodiments negotiation is accomplished between routers at different points in the network based on the in-band routing data. Practice of the invention extends to intelligent telephony networks and as well to simulated telephone calls between computers in wide area data networks, such as the Internet and Intranets.
摘要:
A caller ID based call routing feature. A processing system in the public switched telephone network (PSTN) receives first identifying information for identifying the source of a telephone call and associates additional information stored in a memory with the first identifying information. The additional information may then be transmitted to the subscriber via the Internet for display. Another feature is a branch calling feature where the subscriber may program a processing system within the PSTN to forward an incoming call to two or more end units (e.g., telephones) simultaneously. If the call at an end unit is answered, answer supervision signaling is transmitted back to the processing system which then terminates all other calls. The processing system then connects the calling party to the subscriber. The branch calling may be made for any combination of local, long distance, and cellular telephone numbers.
摘要:
A system and method blend particular quality considerations into the process of expanding the route choices in a route table, such that more calls can be successfully routed while maximizing certain objectives. The quality considerations drive how additional routes are chosen for the route table and how call volumes are allocated to the chosen routes so that overall profitability can be maximized within the governing constraints of quality targets, route quality, predicted traffic, and route capacity.
摘要:
A telephone network access device switchable between a VOIP service and a PSTN service includes an RJ-11 port, a voice splitter, a first relay, a second relay, a SLIC chip, a CPU, and a ring detection unit. The ring detection unit outputs a first switch signal in response to detecting a ring signal of a PSTN communication. The second relay connects the telephone to the VOIP network in response to a power on event. The CPU controls the first relay and the second relay to connect the telephone to the PSTN network in response to the CPU receiving the first switch signal, and controls the second relay to reconnect the telephone to the VOIP network when the PSTN communication ends.