Nokia Corporation

Nokia Telecommunications Oy

Nokia Solutions and Networks Oy

Nokia Siemens Networks Oy

Nokia Networks Oy

Olli Sakari Kirla

Tommi Koistinen

Jyri Suvanen

Johan Haeggström

Teemu Himanen

Antti Pekka Einari Kurittu

Henrik Lepanaho

Speech transcoding in packet networks may be useful when both incoming and outgoing speech streams of the transcoding entity are packet based. This can be any transcoding entity having packet interfaces. A method can include omitting jitter buffering before decoding in a transcoder and omitting bad frame handling in a decoding stage of a transcoder. The method can also include freezing a decoder and the encoder when a packet is not received. The method can also include sending packet loss information from the decoder to the encoder as side information when the packet is not received. The method can further include setting an outgoing packet stream to permit detection of missing packets by a downstream decoder upon receiving a valid packet after the packet is not received.

Speech transcoding in packet networks

The invention allows improving the control of transmission of the silence detection (SID) frames to prevent annoying effects in the music sent to the calling terminal instead of ringback tones. This can be achieved by disabling the discontinuous transmission functionality during the period of call establishment. There are many different ways to accomplish the idea behind the present invention.

Method and system for enhancing the discontinuous transmission functionality

The invention relates to a network element () and a method for enhancing the quality of digitised analogue signals transmitted in parameterised coded form via a digital network. In order to enable an enhancement of the quality of the digitised analogue signals on network side, the network element comprises means () for extracting signals from and insert signals into the network, first processing means () for processing the extracted parameters in the parameter domain with functions suitable to enhance the quality of the digitised analogue signals and second processing means () for processing the extracted parameters in the linear domain with functions suitable to enhance the quality of the digitised analogue signals. Moreover included analysing and selecting means () determine the expected enhancement of quality in the different processing domains and cause a corresponding insertion of processed signals back into the network. The proposed method comprises corresponding steps.

Audio signal quality enhancement in a digital network

The invention is related to a controller of a communication system that includes a module for exchanging messages indicating a traffic channel change between communication system units before hard handover, a module for changing a downlink traffic channel, a module for detecting a communication system unit on a handover target channel, a module for changing an uplink traffic channel, and a module for completing a hard handover process.

Hard handover method and controller

The invention relates to a method for transcoding audio signals in a communications system. In order to improve the inter-operability between units () capable of handling wideband audio signals and units () or network components () capable of handling narrowband audio signals, it is proposed that first, an audio signal is received in a network element () of a communications network via which said audio signal is transmitted. Next, it is determined in said network element () whether a transcoding of the received audio signal is required. In case a narrowband-to-wideband transcoding of the received signal is required, the received narrowband audio signal is transcoded into a wideband audio signal in the network element (). The generated wideband audio signal is then forwarded to the receiving terminal (). The invention equally relates to a corresponding communications system and its components.

Method for transcoding audio signals, transcoder, network element, wireless communications network and communications system

The present invention proposes a method of providing information of the echo path of a speech connection in a Packet Data Network, said method comprising the steps of requesting delay information by means for triggering echo removal, said delay information being obtained by signaling a non-speech measurement data unit to said Packet Data Network by an echo estimation means, and measuring the delay of the echo of said measurement data unit by said echo estimation means; replying the echo delay of said measurement data unit from said echo estimation means; generating delay information including at least said echo delay of said measurement data unit; and adjusting a shift register being related to an echo removing device by the use of said delay information such that the performance of said echo removing device is optimized in view of its echo removing capability.

Delay measurement system in a packet network

A method for changing a data transfer rate on a communications connection, wherein information is processed using a processing method at a first point of the communications connection. The processed information is transferred to a second point of the communications connection where the information is restored using a restoration method that corresponds with the processing method. The processing method is selected from a given set of processing methods and the restoration method is selected from a given set of restorations methods. The processing method is selected so that the processed information is produced at a rate that is less than or equal to the lowest of the data transfer rates before and after the change in the data transfer rate.

Method for changing the route of a data transfer connection and for increasing the number of connections over a data transfer link

In a method for performing a handover with the view of changing a telecommunications channel utilized by a connection between a mobile station and a mobile services switching center in a mobile communications system, to make sure that the handover causes as little interference as possible to the ongoing connection, the allocation of a new radio channel to the connection involves checking whether a predetermined trigger condition is met, which happens if either the speech coding method or the data transfer rate changes, or if they both change at the same time, in connection with the handover, and directing the switching function of the mobile services switching center to perform the handover if the checks show that the handover is a BSC-internal handover where the trigger condition is met.

Handover performed within a base station controller

A method and an apparatus for echo cancellation in a digital data transmission system in which system the end of a transmission link to which sound returns as an echo is the far end and the end of the transmission link from which an echo is reflected back is the near end, and in which a speech coding method is used on the echo path at least for a far-end signal transmitted from the far end to the near end, the method comprising the following steps: the echo originating from the near end is estimated with an adaptive linear filter on the basis of the far-end signal, and the echo estimate is subtracted from the near-end signal transmitted from the near end to the far end, whereby the speech-encoded far-end signal is decoded in an echo canceller, and the echo originating from the near end is estimated on the basis of said decoded far-end signal.

Echo cancellation in digital data transmission system

The invention relates to an acoustic echo suppressor and a non-linear processor for an echo canceller in a 4-wire data transmission network. An echo canceller is typically controlled by comparing the power levels of the near-end and far-end signals. The transfer function of an acoustic echo path is usually very non-uniform. According to the invention, the spectrum of the far-end signal is treated, before the signal power level is calculated, with a weighted filter, which models the effect of the transfer function of acoustic echo. As a result, the double talk dynamics is improved, since high-energy vowels of far-end speech, which normally cause low-energy vowels of near-end speech to be clipped in an echo suppresseor, will be attenuated.

Growing community of inventors

Espoo, Finland

Olli Sakari Kirla