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.

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.

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 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 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 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.

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.

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.

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.