199 results about "Voice traffic" patented technology

The present invention generally to a multimode terminal including a wireless local area network (WLAN) system and a Bluetooth system that avoids radio interference between the two systems by collaborative coexistence methods that include time-sharing, combined frequency and time-sharing, and forward looking combined frequency and time-sharing between the WLAN system and the Bluetooth system. The coexistent multimode terminal and the method of coexistence provide WLAN transmission / receptions that are not impacted when there is no Bluetooth traffic, Bluetooth transmissions / receptions that are not impacted when there is no WLAN traffic, Bluetooth and WLAN transmissions / receptions that are provided fair access to the medium when both Bluetooth and WLAN traffic are present, and high priority Bluetooth traffic, for example, voice traffic, that has priority over non-high WLAN traffic.

A system for simultaneously transporting internet protocol voice traffic and power to an internet protocol telephony transport using the existing twisted pair infrastructure of a traditional phone system. The system includes a twisted pair broadband switch. The twisted pair broadband switch connects on one end to an internet protocol telephone voice switch using Ethernet wiring and connects on another end to a media converter using a twisted pair wiring on the existing twisted pair infrastructure. The media converter condition the voice traffic between a twisted pair interface and an Ethernet interface to allow connection to an internet protocol telephone. Power is simultaneously transported with the voice traffic over the same twisted pair wiring of the existing twisted pair infrastructure.

In a complete network-in-a-box system acting as an enterprise network demarcation point, packets such as voice, video and data packets, are routed over common network connections, such as LAN and WAN. The packets are mapped from a public address field (such as an IP address) and port number to a private address field and port number, the mapping process typically being handled by a NAT (Network Address Translation). The packets are also prioritized, by marking the packets for priority queuing and routing, and configuring the bandwidths of the WAN traffic and the voice traffic to predetermined quantities and configuring the address fields of the voice devices. Simultaneous transmission of the various packets can be limited to predetermined quantities, typically by utilizing a CAC (Client Access Control). Secure firewalls are also included as well as a performance test client application that provides a defined workload generated across the WAN interface for capacity planning measurements and allows remote monitoring of the QoS (Quality of Service) data, such as latency, jitter, lost packets and MOS scores. Optionally, a simple, common remote management interface is included, allowing service providers to configure, upgrade and manage the system. Additionally, address fields can be provided to voice, video or data devices attached to a LAN port. VPN authentication and encrypted sessions can be tunneled through the firewall for access to an internal network by using a VPN terminator. For power outages and other emergency purposes, additional ports that connect to PSTN (Public Switched Telephone Network) analog telephone lines as well as other analog telephones or devices, can be provided. Another advantageous element is that most of the above components or features may be enabled or disabled.

A system that incorporates the subject disclosure may perform, for example, a method for detecting signal interference in a first segment of a plurality of segments of a radio frequency spectrum of a wireless communication system, determining according to the signal interference a measure of quality of service of the first segment for transmitting voice traffic, comparing the measure of quality of service to a desired measure of quality of service measure for voice traffic, determining from the comparison that the first segment is not suitable for voice traffic, and notifying a system that the first segment is not suitable for voice traffic. Other embodiments are disclosed.

An apparatus and method for transferring a message from a calling party to an unavailable telephonically called party. The method comprises the steps of inititating a call to the called party over a communications channel, identifying that the called party is not available and instructing the calling entity to record the message at the calling entity and after recording to transmit the message to the called entity as a data message via a data channel for reception by the called entity. The system reduces the use of the voice / telephone communications networks by allowing a voice message to be transmitted through data channels, reducing the amount of voice traffic on the voice communication lines. The fact that the original voice message was actually stored with the calling entity after the telephone voice communication had been discontinued is transparent to both the calling party and the called party. Greetings for the called party can be stored at a messaging system of the calling entity.

A method, apparatus, and software for providing lifeline service during power failure affecting Customer Premises Equipment (CPE) in a Digital Subscriber Loop (DSL) arranged to carry voice traffic in band rather than on a separate analogue POTS band. The arrangement provides a reduced service, capable of supporting at least one voice call, to operate during such power failure and, where a single call is in progress during power failure, that call may be maintained during the transition from DSL to analogue POTS lifeline service. Where a call is in progress upon power restoration, the lifeline POTS service may be maintained until completion of the call so as not to interrupt a potential lifeline call.

A fixed wireless system (FWS) utilizing Orthogonal Frequency Division Multiplexing (OFDM) communication techniques is spectrally efficient and responsive to communications involving both voice and high speed data, such as Internet data. The FWS includes a wireless base unit; a plurality of fixed wireless remote units; a plurality of wireless data traffic channels available between the wireless base unit and the plurality of fixed wireless remote units; and a plurality of wireless voice traffic channels available between the wireless base unit and the plurality of fixed wireless remote units. Each wireless traffic channel is identifiable by a unique combination of frequency and time slots. Each wireless data traffic channel is used for carrying high speed data in addressed data packets to and from the plurality of fixed wireless remote units. On the other hand, each wireless voice traffic channel can be assigned and dedicated to a particular voice communication call involving one of the plurality of fixed wireless remote units for carrying voice data of the call.

An apparatus for providing bifurcated voice and signaling traffic over a cable telephony architecture by segregating signaling traffic and voice traffic and transmitting the respective traffic over two different mediums to a switch to establish a phone call.

A method and system for servicing data traffic in wireless local area networks (such as IEEE 802.11 networks) in multiple queues having different levels of priority. These queues include at least a low priority queue for 'best effort' traffic, a medium priority queue for streaming data such as video pictures, and a high priority queue for voice traffic, and are serviced in order of decreasing priority. To prevent the 'best effort' traffic in the low priority queue from being 'starved', a bit is set to indicate when such a condition is likely to occur, and the low priority queue is served first when that bit has been set. Alternatively, low and medium priority traffic is handled on a weighted round-robin basis, and high priority traffic is given strict priority over both. Transmit and receive queues are handled on a rotating priority basis.

In a wireless telecommunications network, a Code Division Multiple Access (CDMA) scheme is applied to data to encode it. The encoded data transmitted in the uplink using an Orthogonal Frequency Division Multiplexing (OFDM) frame structure including a cyclic prefix (CP). The CDMA encoded data may be time multiplexed with Single Carrier-Frequency Division Multiple Access (SC-FDMA) transmissions. The CDMA transmissions may be used for relatively small payloads, such as those associated with voice traffic and control signals, and the SC-FDMA transmissions used for higher date rate transmissions. This enables autonomous transmission, without scheduling, for smaller payloads. A transmitter includes a selector 3 for performing time multiplexing in the uplink, the required mode being indicated by the Hybrid Automatic Repeat Request (HARQ) ID. A first branch 1 carries SC-FDMA data and a second branch 2 carries data to be transmitted as a CDMA scheme. A CAZAC code may be used in encoding the CDMA data.

A communication device according to the present invention enhances transfer of time-critical data between one or more LANs and a device (e.g., edge router, etc.) coupled to a backbone network. A virtual bottleneck in the form of a queue is introduced by the communication device at the customer premises or customer end of a backbone network access line where the network congestion or bottleneck resides. The virtual bottleneck delays and / or discards time insensitive traffic prior to time-critical or voice traffic being delayed in the edge router. This is accomplished by the virtual bottleneck queue including a storage capacity or length less than that of a queue utilized by the edge router. A traffic manager or scheduler controlling the virtual bottleneck dynamically adjusts the virtual bottleneck based on the bandwidth required for time-critical packets to ensure sufficient bandwidth is available for those packets.

A message interworking gateway (1) resides in an overlay IMS network (2) associated with a home SS7 network (10). The gateway (1) intercepts requests for routing information (MAP-SRIs) from either the home SMSC (11) or a foreign SMSC. If the message recipient is available on the IMS overlay network (2) the gateway (1) performs protocol translation AND routes the translated message to the recipient. If not, the gateway returns the SRI to the home (10) network. The gateway operates without updating the home HLR (12) with user presence data and hence voice traffic can be kept on circuit switched technology without being routed to the IMS overlay network (2).

A method and system for providing remote voice communication service via a wireless local area network (WLAN). A WLAN access point is arranged with predefined security settings to establish a VPN tunnel over a packet network with a VPN terminator on a remote network. The access point then wirelessly receives voice traffic from a wireless terminal operating in the WLAN and sends the voice traffic over the VPN tunnel to the remote network, for transmission to a call control device, which provides connectivity with a voice transport network. And the access point similarly receives voice traffic via the VPN tunnel from the remote network and sends the voice traffic wirelessly to the wireless terminal. Advantageously, multiple wireless terminals operating on the WLAN can engage in voice calls concurrently via the access point.

A digital subscriber loop modem arrangement is arranged to respond to a fault condition so as to provide at least a voice lifeline service over DSL service on a digital subscriber line in which data and voice traffic are transported on a plurality of carriers. On detection of a power loss at the subscriber end of the line, the number of carriers is reduced so as to maintaining transmission only of those carriers transporting the voice traffic. This reduces the power demand which can now be met from the remote end of the line. A method of operating a digital subscriber loop modem arrangement under a fault condition so as to provide at least a voice over DSL service on a digital subscriber line in which data and voice traffic are transported on a plurality of carriers, the method comprising the steps of: replacing the data with random data, disabling those data carriers on which the random data is transported, and maintaining transmission of those carriers transporting the voice traffic.

A switch capable of handling voice-over-IP (VoIP) traffic between calling devices and called devices. The switch comprises: 1) call application nodes for executing call process server applications, wherein a first call process server application and a similar second call process server application form a first load sharing group server application; and 2) network address translation nodes for executing firewall server applications. A first firewall server application executed on a first network address translation node is associated with a similar second firewall server application executed on a second network address translation nodes separate from the first network address translation node. The first and second firewall server applications form a second load sharing group server application. The second load sharing group server application receives VoIP traffic and selects one of the first and second firewall server applications to verify that the VoIP traffic is authorized to access at least one of the call process server applications in the call application nodes according to a load distribution algorithm.

A method and apparatus for performing traffic engineering, e.g., allocating bandwidth, on a wireless access network are disclosed. For example, the method determines a number of subscriber stations (SSs) that a Base Station (BS) is capable of supporting in accordance with at least one performance objective for voice traffic, wherein the at least one performance objective for voice traffic comprises a type of codec. The method then allocates bandwidth by the base station in accordance with the number of subscriber stations that the base station is capable of supporting.

A system and method for data communication over a cellular communications network that allows the transmission of digital data over a voice channel using a vocoder that operates in different modes depending upon characteristics of the inputted signal it receives. To prepare the digital data for transmission, one or more carrier signals are encoded with the digital data using one of a number of modulation schemes that utilize differential phase shift keying to give the modulated carrier signal certain periodicity and energy characteristics that allow it to be transmitted by the vocoder at full rate. The modulation schemes include DPSK using either a single or multiple frequency carriers, combined FSK-DPSK modulation, combined ASK-DPSK, as well as PSK with a phase tracker in the demodulator. These modulation schemes permit data communication via a CDMA, GSM, or other type of voice traffic channel at a low bit error rate.

A connection between an external host and a host internal to a network (e.g., VPN or LAN) and located behind a firewall and / or NAT-enabled device is established and maintained. A dynamic DNS server is updated with the remote location (RL) information associated with the internal host. This information includes the dialed number (i.e. the number assigned to the internal host), the physical address of the internal host, the private IP address corresponding to the internal host, the public IP address corresponding to the firewall, and the port on which voice data is to be communicated. Each time the internal host's location changes, a new and updated record is sent to the dynamic DNS server. The call placed by an external host reaches its final destination based on the record information in the DNS server table. Once the connection between the external host and the internal host is established, voice traffic is “cut-through” the firewall on this channel associate with the port designated by the RL record.

A method and system for increasing the efficiency of providing bandwidth for voice traffic to a data provider via asynchronous communication mediums is provided. This is generally accomplished by not transmitting any data during the silence periods and playing out background noise (i.e., comfort noise) at the other end, to obtain significant bandwidth savings.

A communications system multiplexes voice communications signals onto one or more transport level connections established through a packetized network, such as the Internet. The invention supports the use of variable-length packets and accommodates variable jitter. The system conforms to real time protocol (RTP) and employs internet telephone gateways (ITGs) to bind users to channel identifiers, to indicate payload type and length, to provide channel identification and time stamps, and to indicate cessation and resumption of voice traffic from a particular user through use of, for example, marker bits.

A method of rate shifting specially suited for the voice traffic, which differentiates poor channel conditions from a heavily loaded channel of a WLAN and adapts to the network traffic condition and channel condition promptly with low rate of false shifting. Determining when to rate shift is based on a combination of the received signal strength indication and the retry rate.

A telecommunications network (10) includes a gateway (18) that receives signaling information in a message based signaling format from a Class 5 softswitch (26). The gateway (18) also receives voice traffic over an inter-machine trunk from a public switched telephone network (12). The gateway (18) places the voice traffic into data packets. The gateway (18) transfers the data packets and the signaling information to an Internet Protocol network (30). The data packets and the signaling information may be transferred over a common physical link and over separate logical links.

Systems, apparatus and methods in a mobile device for saving power by powering down all transceivers not carry traffic are presented. The traffic may be voice and / or data traffic. A mobile device may select single transceiver to carry voice traffic and the same or different transceiver to carry data traffic. A mobile device first determines its position (e.g., a coarse position estimate) then consults a database or map to determine which networks are theoretically available. The mobile device executes a rule against the theoretically available networks to select the single network, then enables the transceiver for the one network to determine if the network is actually available for use. If the database inaccurately states a network is available from a current position but the transceiver shows that the network is actually not actually available, a next network from the database or map and corresponding transceiver are selected.

Methods and systems for providing voice and date services in a femtocell wireless network. The proposed approach integrates IWLAN architecture into femtocell architecture by introducing a gateway to serve both IWLAN and femtocell users. The proposed approach handles the voice and data in a different way so that it enhances the data handling efficiency while re-using existing MSC investment. The proposed approach carries the data traffic from a femtocell base station to the gateway in native IP packet, instead of encapsulating them in 3G data, thus enhancing the efficiency and performance for the data traffic. The data traffic can then be sent to GGSN or directly to packet data network. The approach tunnels voice traffic to MSC through the gateway as in conventional Iu-CS approach.

The invented method of reducing voice frame network inbound traffic congestion first determines whether a first defined threshold level of inbound voice and data traffic is reached. If so, then the method discriminates between inbound voice and data traffic within an input queue. Discrimination is by analyzing the size of each packet and comparing it to predefined packet size criteria and / or by analyzing the rate at which packets of inbound voice and data traffic arrive in the input queue and comparing it to predefined arrival rate criteria. Finally, the method frees space within the input queue for use by inbound voice traffic until the first defined threshold level of inbound traffic no longer is reached. The freeing of space includes selectively discarding data, which discarding preferably continues until a second defined threshold level of inbound traffic is reached, the second defined threshold level being less than the first defined threshold level. The invented apparatus includes decision logic for making such a determination and queue management logic responsive thereto for such discrimination and freeing of space. A user interface preferably is provided that permits a user to define the first and second defined threshold levels, which are implementation- and application-dependent.

The present invention protocol offers guaranteed Quality of Service (QoS) for concurrent calls in a highly dynamic, scalable network. The invention employs a TDMA reservation technique to transmit voice traffic to multiple destinations and a CSMA / CA contention scheme to support data traffic. The present invention operates over a link-state based routing protocol that reliably floods routing and resource reservation information to network nodes. The present invention is suitable for general networking applications that require QoS for multimedia traffic in a mobile, Ad-Hoc network and enables conference calls to be established and operated in that type of network under various conditions. Moreover, the present invention capitalizes on certain properties of a radio, such as a RAKE type receiver, that sums up multiple, identical transmissions from multiple sources. In addition, the present invention enables roaming between and / or within a group or island of network nodes during the lifetime of a call.

A system and method for data communication over a cellular communications network that allows the transmission of digital data over a voice channel of the communications network. Digital data is encoded into DBPSK data using differential binary phase shift keying encoding. The DBPSK data is then sent across the cellular network using a vocoder having a linear predictive or other speech compression codec. At the receiving end, the DBPSK data is demodulated back into the original digital data. This approach permits data communication via a CDMA, GSM, or other type of voice traffic channel at a low bit error rate.

A method and system for providing a private voice call service to a mobile subscriber are provided. A soft switch in a mobile communication system, upon receipt of an origination message including a private voice request information from a calling MS, transmits to a VPN server a private circuit service request message including a service ID for the private voice call service and the network address information of a caller and called. The VPN server establishes a communication path for edge routers connected to calling and called gateways according to network address information. Thus, voice traffic is transmitted between the calling MS and called MS though the communication path of the VPN.

An apparatus and method for transferring a message from a calling party to an unavailable telephonically called party. The method comprises the steps of inititating a call to the called party over a communications channel, identifying that the called party is not available and instructing the calling entity to record the message at the calling entity and after recording to transmit the message to the called entity as a data message via a data channel for reception by the called entity. The system reduces the use of the voice / telephone communications networks by allowing a voice message to be transmitted through data channels, reducing the amount of voice traffic on the voice communication lines. The fact that the original voice message was actually stored with the calling entity after the telephone voice communication had been discontinued is transparent to both the calling party and the called party. Greetings for the called party can be stored at a messaging system of the calling entity.

A switching apparatus for switching packetized voice traffic between a plurality of communication devices, the switching apparatus comprises a multi-layer switch, a plurality of communication ports, control means and ingress processing means, said packetized voice traffic comprises call control packets and medium packets which are exchanged between the communication devices via said communication ports, wherein medium packet traffic from a first communication device to a second communication device is split into a first call segment and a second call segment, the first call segment originates from said first communication devices and terminates at said switching apparatus, the second call segment originates from said switching apparatus and terminates at said second communication device, each medium packet from said first communication device is processed by said ingress processing means of said switching apparatus before onward transmission to said second communication device.