A telecommunication system routs real-time information traffic from an originating digital radio unit served by an originating network to a terminating unit served by a terminating network via an intermediate network interconnecting the originating and terminating networks. The originating digital radio unit has an encoder/decoder (e.g., a vocoder) for generating digital wireless frames from information that is input thereto. The originating network includes an originating node with an encoder/decoder for performing wireless-specific conversion of the digital wireless frames to digital wireline (e.g., PCM) traffic. The intermediate network includes an originating-end interface node with an encoder/decoder for compressing the digital wireline traffic for transport across the intermediate network. Optimization of communications routed between the originating and terminating units is achieved by routing the digital wireless frames without wireless-specific conversion being performed at the originating node of the originating network nor compression conversion being performed at the originating-end interface node of the intermediate network, such that the rate of information traffic throughput is maximized.
A communications network includes a data network that is coupled to a wireless access network and other devices. The wireless access network enables access by mobile stations of the data network. Each mobile station, and optionally, one or more of other devices coupled to the data network, contains an adaptive multi-rate codec (coder/decoder) that can be set to operate at a plurality of rates. Based on the selected one of the plurality of rates, the mobile station or other device sets a quality-of-service (QoS) indicator value in a packet carrying data, such as real-time data, over the wireless access network and/or data network. By varying QoS requirements using the QoS indicator value for different codec rates, bandwidth requirements are varied so that more efficient usage of the data network is provided.
A process (111,101) of sending packets of real-time information at a sender (311) includes steps of initially generating at the sender the packets of real-time information with a source rate (s11) greater than zero kilobits per second, and a time or path or combined time/path diversity rate (d11), the amount of diversity (d11) initially being at least zero kilobits per second. The process sends the packets, thereby resulting in a quality of service QoS, and optionally obtains at the sender (311) a measure of the QoS. Rate/diversity adaptation decision may be performed at receiver (361') instead. Another step compares the QoS with a threshold of acceptability (Th1), and when the QoS is on an unacceptable side of said threshold (Th1) increases the diversity rate (d11 to d22) and sends not only additional ones of the packets of realtime information but also sends diversity packets at the diversity rate as increased (d22). Increasing the diversity rate (d11 to d22) while either reducing or keeping unchanged the overall transmission rate (sij+dij) is an important new improvement in even solely-time-diversity embodiments.
A Voice Over Internet Protocol (VOIP) receiver (630), operating in conjunction with a transmitter (606) receives a sequence of voice packets representing a speech utterance transmitted over a VOIP wireless interface (112). A receive packet buffer (120) buffers the received sequence of voice packets after receipt and before playback of reconstructed speech. A processor (650), operating under program control, determines a transmission buffer (108) delay of a first packet in the sequence of packets representing the speech utterance. The control processor (650) further sets a prescribed amount of delay in the receive packet buffer (120) based upon the transmission buffer (108) delay so that the transmission buffer delay+receive buffer delay=a predetermined total delay. The status of the receiver buffer (120) is monitored and tracked by or fed back to the transmitter side to minimize receive buffer (120) under-runs by use of CDMA soft capacity (200), link dependent prioritization (300), real-time packet prioritization (400) and/or variation of vocoder (624) rates (500).
A voice gateway is used to transmit voice signals through the Internet. Upon receiving a connection request, the gateway calculates an end-to-end delay time and selects a proper route according to the calculated delay time to improve the quality of voice transmission. A gateway (10) on the caller side multicasts a connection request to gateways (20) on the receiver side. Each of the receiver gateways returns a response to the caller gateway. According to the response, the caller gateway calculates a compression/decompression allowance and selects a route that employs a compression rule whose compression or decompression time is shorter than the compression/decompression allowance.
Apparatus, and accompanying methods for use therein, for a telephony gateway intended for use, e.g., paired use, at opposite ends of a data network connection, in conjunction with at each end, e.g., a private branch exchange (PBX) for automatically routing telephone calls, e.g., voice, data and facsimile, between two peer PBXs over either a public switched telephone network (PSTN) or a data network, based on, among other aspects, cost considerations for handling each such call and called directory numbers, monitoring quality of service (QoS) then provided through the data network and switching ("auto-switching") such calls back and forth between the PSTN and the data network, as needed, in response to dynamic changes in the QoS such that the call is carried over a connection then providing a sufficient QoS.
