Abstract: One embodiment of the present invention includes a method for synchronizing packet production and receipt from a digital signal processor (DSP) with polled transmission opportunities in an IEEE 802.11e network. The method comprises producing a packet via the DSP and transmitting the packet from the DSP to a transmission device at a time that is substantially synchronized with a next polled transmission opportunity granted by an access point in an IEEE 802.11(e) network. The method also comprises time stamping the arrival of the packet at the transmission device and transmitting the packet from the transmission device.

Abstract: Determining the reasons for packet loss in a wireless local area network; such as an IEEE 802.11 network. The method uses the strength of received signals at a wireless station, such as the received signal strength indicator (RSSI), noise levels (e.g, non-802.11 energy in the 2.4 GHz frequency range), and packet loss information together to determine the network status. In most implementations, the 802.11 implementation (e.g., the 802.11 driver) is responsible for maintaining statistics on the RSSI and the noise whereas the measure of the packet loss is the responsibility of the application. The network status is determined by the endpoint station independent from the access point.

Abstract: An unconnected power save method reduces the battery consumption of a wireless station in a wireless local area network (WLAN). A wireless station enters a sleep mode between scans for a prospective access point. The sleep duration time is determined using the current position of the station relative to the WLAN network topology.

Abstract: A method for reducing the number of channels that are scanned during a handoff situation for a wireless endpoint station in a wireless local area network (WLAN), such as an IEEE 802.11 network, by tracking past user movements within the WLAN. A suitable sequence of channels to be scanned is generated so that reduce the number of handoffs are reduced and the average time during a handoff is reduced. To reduce handoffs and handoff latencies, the principle of locality (user movement patterns tend to be repetitive) is used. By tracking the past handoffs, future handoffs are predicted.

Abstract: One embodiment of the present invention includes a method for synchronizing packet production and receipt from a digital signal processor (DSP) with polled transmission opportunities in an IEEE 802.11e network. The method comprises producing a packet via the DSP and transmitting the packet from the DSP to a transmission device at a time that is substantially synchronized with a next polled transmission opportunity granted by an access point in an IEEE 802.11(e) network. The method also comprises time stamping the arrival of the packet at the transmission device and transmitting the packet from the transmission device.

Abstract: A package under MGCP protocols for providing control over cost and quality of voice calls on an IP-based network such that an optimal selection of connection parameters under given network conditions can be made by a call agent. A gateway transmits an event to the call agent, which in turn changes a call parameter or modifies a connection option for the call.

Abstract: Disclosed above are various embodiments of VoIP communication systems that utilize low cost IP phones that rely on a centralized VoIP controller for much of the processing. Reducing the processing taking place on an IP phone may reduce the number of components that need to be on the IP phone which may result in a less expensive IP phone in terms of both cost and power. When the IP phone is embodied as a WIPP, the reduced processing may also result in more efficient communication between the WIPP and an AP. The increased communication efficiency may result in less power being used by the WIPP and effectively extend the battery life. Since a number of redundant components have been centralized, the VoIP system as a whole may be less costly. Also, centralized control may provide greater functionality and versatility in the setup and configuration of a VoIP communication system.

Abstract: A technique is described to change the codec or MAC (message authentication code) size in a packet security unit for PacketCable communications during realtime voice transmissions is described. An algorithm that provides fast RC4 key advancing to prevent MIPS (millions of instruction cycles per second) overflow is used to perform codec or MAC size changes. The invention is performed without changing the keying material, where the sender and receiver must continue the RC4 encryption process from its state prior to the codec or MAC size change. A sender needs to preserve continuity of the timestamp across a codec change, since the timestamp reflects realtime. Changing the codec or MAC size is likely to change the frame parameters. To preserve continuity of the RC4 state and the timestamp across the codec/MAC size change, the sender TX and receiver RX generates a new frame number. The new frame number is applied to the first frame generated by the new codec or MAC size.

Abstract: Providing interoperability on a voice communication network by detecting incompatible bit packing payload formats of ADPCM encoded voice data signals in an encoder/decoder. A mismatch of bit packing formats between little endian format and big endian format is determined through the use of a media gateway control protocol package called Real Time Protocol that allows a Call Agent to detect the presence of garbled audio and allows an intelligent gateway to detect the payload format being used by remote endpoints during call setup procedures and then switch the gateway's encoding for the payload to the payload format being used by the remote endpoint. The invention may be applied to an ITU G.726 encoder/decoder.

Abstract: A system to enable TTY communication between a legacy TTY phone and an IP phone over a packet network using a TTY relay protocol established between the IP phone and a media gateway at a central office of the PSTN. The system may also use a voice over Internet Protocol gateway to connect a personal computer to the packet network instead of an IP phone to accomplish TTY communication between software on the computer and a legacy TTY phone.

Abstract: A method for transmitting voice data traffic over a wireless medium using a decision process for the admission of a voice call, such as a call voice over Internet Protocol (VoIP) over a wireless local area network (WLAN). Admission of the call through the network is based upon the priority level of the call and a congestion level of the WLAN in order to increase the quality and the number of calls on a BSS. Paramters such as codec and packetization periods are selected for the call based upon the priority and congestion level. As the level of congestion in the network Increases, only calls above a certain Priority level are allowed into the network.

Abstract: A method for reducing the number of channels that are scanned during a handoff situation for a wireless endpoint station in a wireless local area network (WLAN), such as an IEEE 802.11 network, by tracking past user movements within the WLAN. A suitable sequence of channels to be scanned is generated so that reduce the number of handoffs are reduced and the average time during a handoff is reduced. To reduce handoffs and handoff latencies, the principle of locality (user movement patterns tend to be repetitive) is used. By tracking the past handoffs, future handoffs are predicted.

Abstract: A system and method are provided for improving the throughput of wireless networks, such as IEEE 802.11 networks. Individual devices in the network determine the level of activity on the network. In the event that the network is active, the devices increase throughput by skipping the pending packet transmission and appending its data payload to a future packet transmission. By encoding multiple payloads in a single packet transmission, packet overhead is reduced and MAC layer delays are eliminated. The data payloads are subsequently split back into individual packets for transmission over the wired portion of the network.

Abstract: An unconnected power save method reduces the battery consumption of a wireless station in a wireless local area network (WLAN). A wireless station enters a sleep mode between scans for a prospective access point. The sleep duration time is determined using the current position of the station relative to the WLAN network topology.

Abstract: Determining the reasons for packet loss in a wireless local area network; such as an IEEE 802.11 network. The method uses the strength of received signals at a wireless station, such as the received signal strength indicator (RSSI), noise levels (e.g, non-802.11 energy in the 2.4 GHz frequency range), and packet loss information together to determine the network status. In most implementations, the 802.11 implementation (e.g., the 802.11 driver) is responsible for maintaining statistics on the RSSI and the noise whereas the measure of the packet loss is the responsibility of the application.

Abstract: A system and method for transmitting voice data traffic over a wireless medium, such as a voice call using a wireless Internet Protocol mobile phone on a wireless local area network, to increase the quality and the number of voice calls on a Basic Service Set. Endpoints are capable of dynamically switching to alternate voice coding profiles. The WLAN is surveyed by a wireless endpoint to determine whether a congested network condition exists. Coding profiles may be dynamically switched to decrease the congested conditions and improve call quality. If dynamically switching of profiles in not available then the wireless endpoint gracefully degrades the transmission by periodically dropping packets until network congestion reduces and/or call quality increases.

Abstract: A method for defining a payload format in Real Time Protocol (RTP) messages for voice band data transmission to be transmitted over a packet network to an Internet Protocol (IP) telephony endpoint, such as an IP phone. The invention extends RFC 2833 standards for Real Time Protocol (RTP) message payload format to include voice data transmission. The invention includes receiving voice transmission signals containing a voice band data from the public switched telephone network into a packet network and converting said voice transmission signals into data packets using real time protocol (RTP) and assigning an RTP event code in the RTP payload format for the voice band data applications. Applications include Caller Identification (Caller-ID), Visual Message Waiting Indication (VMWI), and Advanced Digital Subscriber Interface (ADSI).

Abstract: A technique is described to change the codec or MAC (message authentication code) size in a packet security unit for PacketCable communications during realtime voice transmissions is described. An algorithm that provides fast RC4 key advancing to prevent MIPS (millions of instruction cycles per second) overflow is used to perform codec or MAC size changes. The invention is performed without changing the keying material, where the sender and receiver must continue the RC4 encryption process from its state prior to the codec or MAC size change. A sender needs to preserve continuity of the timestamp across a codec change, since the timestamp reflects realtime. Changing the codec or MAC size is likely to change the frame parameters. To preserve continuity of the RC4 state and the timestamp across the codec/MAC size change, the sender TX and receiver RX generates a new frame number. The new frame number is applied to the first frame generated by the new codec or MAC size.

Abstract: The transport of ABCD bits in a digital packet network is facilitated by the inclusion of the ABCD bits into the RTP header of the digital packets. The ABCD bits are first placed in the unused bits of the RTP header. Redundancy of the previous ABCD bits, when applicable, is achieved by restricting the size of fields in the header to reserve space for redundant ABCD bits. The ABCD bits are placed in the reserved portions of restricted header fields.

Abstract: Packet production by a digital signal processor (DSP) is synchronized with modem transmission opportunities in a cable network. The cable modem computes the average waiting time a packet spends waiting to be transmitted by the modem. The DSP re-syncs transfer of packets to the modem by using this average waiting time minus a small amount sufficient to account for the maximum anticipated propagation delay, as the amount to delay the transmission of its next packet to the cable modem. This invention is a mechanism for re-synchronization that is largely independent of any propagation delays inherent in the hardware architecture.