Abstract: The present invention proposes a transmission method for transmitting data in a telecommunication network (NW), said telecommunication network comprising at least one access network (RAN) adapted to transmit data to/from at least one terminal device (MS) and a core network (CN, MSC, GW/TC), which are connected to each other via a first interface (Iu), wherein said data are exchanged via said first interface (Iu) using a real-time transfer protocol (RTP). Preferably, said real-time transfer protocol complies with ITU-T H.225 and ITU-T H.323 Recommendations. Consequently, transmission resources available on said first interface can be effectively and economically used, since the RTP protocol conforming to ITU-T H.323 Recommendation is adapted to handle discontinuous transmission.

Abstract: The invention is concerned with transmission of speech information over mixed networks, particularly over a segment using packet transmission in an otherwise circuit switched data transmission network. According to the invention, PCM transmission lines are connected with a packet based network such as an IP network in such a way, that data from one or more channels of a first PCM transmission line is collected in the payload part of a data packet. Further, the destination address of the data packet is constructed in such a way, that the packet network entity being connected to the second PCM transmission line can determine based on the destination address of the packet, to which channel or channels of the second PCM transmission line the data should be transmitted.

Abstract: The invention relates to a data transfer method in a digital cellular radio network, the method comprising channel coding the information to be transferred for transmission. In order to implement a data rate of 14.4 kbit/s in GSM type of cellular radio systems by employing one time slot only for data transmission, the channel coding according to the method of the invention comprises grouping bits to be transmitted in blocks having the minimum size of 288 bits, carrying out convolutional coding for the blocks with a code rate of ½ by using GSM convolutional coding polynomes, and puncturing the bits obtained by deleting bits from each block so that blocks containing no more than 456 bits will be obtained.

Abstract: A telecommunication network and a routing method, wherein user data are routed via an access network (2) to a gateway device (5) of a core network (7) connected to the access network (2) via a first interface. The control of the gateway device (5) is performed by supplying a control information from the core network (7) to the gateway device (5) via a second interface, and the user data are routed directly between the access network (2) and the gateway device (5) via the first interface. Thus, the user data are routed directly from the access network (2) to the gateway (5), whereas the control can be performed via a dedicated and open control interface. Processing resources can be shared between several applications and trunking gains can be utilized. Furthermore, real-time transport capabilities can be provided by a simple method of connecting the Internet to the mobile network.

Abstract: Data are transmitted between a core network (CN) and at least one access system (BSS) of a GSM based access system type via an Iu interface (Iu-I/F). Also, the present invention aims to provide accordingly adapted network elements. By virtue of the present invention, substantial savings concerning resources can be obtained by supporting data formats of GSM coded speech on the open, i.e., not network operator proprietary, Iu interface according to UMTS/3rd generation specifications. Thus, GSM transcoders and 3rd generation transcoders as well as GSM base station controllers and 3rd generation radio network controllers can be integrated, so that the same hardware resources can be used for both systems, i.e., for GSM as well as for 3rd generation.

Abstract: The invention concerns a method for boosting data transmission in a telecommunications system and a mobile communications system. The invention is characterised in that the telecommunications network uses at least on a part of the transmission path between the fixed station, e.g. a base transceiver station, and the transcoder unit speech coding at a lower transmission rate than the transmission rate of the speech coding used on the transmission path between the fixed station and the terminal equipment. The speech parameters received from the terminal equipment are converted for the speech coding method used on the transmission connection between the fixed station and the transcoder unit and vice versa. In the network transcoder unit it is possible to reconvert speech parameters received from the direction of the terminal equipment e.g. into speech parameters of the speech coding used on the transmission path between the terminal equipment and the fixed station.

Abstract: The invention relates to a data transfer method in a digital cellular radio network, the method comprising channel coding the information to be transferred for transmission. In order to implement a data rate of 14.4 kbit/s in GSM type of cellular radio systems by employing one time slot only for data transmission, the channel coding according to the method of the invention comprises grouping bits to be transmitted in blocks having the minimum size of 288 bits, carrying out convolutional coding for the blocks with a code rate of ½ by using GSM convolutional coding polynomes, and puncturing the bits obtained by deleting bits from each block so that blocks containing no more than 456 bits will be obtained.

Abstract: The invention relates to a data transfer method in a digital cellular radio network, the method comprising channel coding the information to be transferred for transmission. In order to implement a data rate of 14.4 kbit/s in GSM type of cellular radio systems by employing one time slot only for data transmission, the channel coding according to the method of the invention comprises grouping bits to be transmitted in blocks having the minimum size of 288 bits, carrying out convolutional coding for the blocks with a code rate of ½ by using GSM convolutional coding polynomes, and puncturing the bits obtained by deleting bits from each block so that blocks containing no more than 456 bits will be obtained.

Abstract: A method carries out statistical multiplexing in a telecommunications network comprising a transmitting unit for transferring channels containing information from at least two traffic sources, a receiving unit and a transmission link between them. The information to be transmitted in the telecommunication system is transferred over the transmission link in transmission frames.

Abstract: An arrangement and method for executing different applications associated with several connections using at least one processing unit is provided. The arrangement includes a controller and a database that has several applications. The at least one processing unit and the controller are connected to the database. The controller allocates an available processing unit for each connection at the beginning of the connection, identifies the applications required by the connection concerned, and loads the applications from the database to the allocated processing unit.

Abstract: A method and a transmitter (100) and a receiver (102) for transmitting messages (114) in a digital telecommunications system. The information (104) to be sent is encoded in an information encoder (106) of the transmitter (100) into frames that are transmitted via a transmission path (108) to the receiver (102). A good state and a bad state have been defined for the frames, and a bit pattern corresponding to each message (114) has been defined. The messages(114) are encoded with a message encoder (116) of the transmitter (100) and transmitted to the receiver (102). The messages (114) are transmitted together with the information (104) via a common transmission path (108), inserting the bit pattern corresponding to the message (114) in the frame, forming the frame corresponding to the time of transmitting the message (114) as bad, and in short sequences, preferably only one frame at a time.

Abstract: An arrangement for cancelling an echo returning to a first mobile station (MS1) from a second mobile station (MS2). The arrangement comprises tandem free operation (TFO) and an acoustic echo canceller (AEC) to which criteria have been defined, upon fulfillment of which the echo canceller replaces an uplink speech signal of the second mobile station (MS2) by comfort noise (CN) in a second coding format. In accordance with the invention, the echo canceller (AEC) generates additional information (AI) if the tandem free operation (TFO) is active and if criteria for replacing the speech signal with the comfort noise are fulfilled at the same time. In response to said additional information (AI), a first transcoder (TRAU1) encodes the speech signal to be transmitted to the first mobile station (MS1) by using the comfort noise (CN) that is generated by the echo canceller (AEC) and that is in the second coding format.

Abstract: Data are transmitted between a core network (CN) and at least one access system (BSS) of a GSM based access system type via an Iu interface (Iu-I/F). Also, the present invention aims to provide accordingly adapted network elements. By virtue of the present invention, substantial savings concerning resources can be obtained by supporting data formats of GSM coded speech on the open, i.e., not network operator proprietary, Iu interface according to UMTS/3rd generation specifications. Thus, GSM transcoders and 3rd generation transcoders as well as GSM base station controllers and 3rd generation radio network controllers can be integrated, so that the same hardware resources can be used for both systems, i.e., for GSM as well as for 3rd generation.

Abstract: A telecommunication network and a routing method, wherein user data are routed via an access network (2) to a gateway device (5) of a core network (7) connected to the access network (2) via a first interface. The control of the gateway device (5) is performed by supplying a control information from the core network (7) to the gateway device (5) via a second interface, and the user data are routed directly between the access network (2) and the gateway device (5) via the first interface. Thus, the user data are routed directly from the access network (2) to the gateway (5), whereas the control can be performed via a dedicated and open control interface. Processing resources can be shared between several applications and trunking gains can be utilized. Furthermore, real-time transport capabilities can be provided by a simple method of connecting the Internet to the mobile network.

Abstract: The present invention proposes a transmission method for transmitting data in a telecommunication network (NW), said telecommunication network comprising at least one access network (RAN) adapted to transmit data to/from at least one terminal device (MS) and a core network (CN, MSC, GW/TC), which are connected to each other via a first interface (Iu), wherein said data are exchanged via said first interface (Iu) using a real-time transfer protocol (RTP). Preferably, said real-time transfer protocol complies with ITU-T H.225 and ITU-T H.323 Recommendations. Consequently, transmission resources available on said first interface can be effectively and economically used, since the RTP protocol conforming to ITU-T H.323 Recommendation is adapted to handle discontinuous transmission.

Abstract: The invention concerns a method for boosting data transmission in a telecommunications system and a mobile communications system. The invention is characterised in that the telecommunications network uses at least on a part of the transmission path between the fixed station, e.g. a base transceiver station, and the transcoder unit speech coding at a lower transmission rate than the transmission rate of the speech coding used on the transmission path between the fixed station and the terminal equipment. The speech parameters received from the terminal equipment are converted for the speech coding method used on the transmission connection between the fixed station and the transcoder unit and vice versa. In the network transcoder unit it is possible to reconvert speech parameters received from the direction of the terminal equipment e.g. into speech parameters of the speech coding used on the transmission path between the terminal equipment and the fixed station.

Abstract: The invention relates to a synchronization method in a telecommunication system that includes a transmitting unit, a receiving unit and a transmission link between these two. The information to be transmitted in the telecommunication system is transferred from the transmitting unit to the receiving unit in a data frame that includes an information section and a synchronization section. The bits of the synchronization section are in each consecutive data frame always in the same state as in the previous data frame, and the synchronization section consists of at least one synchronization bit in a selected logical state. In the method, the transmitting and receiving units of the data frame are mutually synchronized with synchronization bits.

Abstract: In a mobile speech communication system including a base station having a data connection to a mobile services switching center through a transcoder unit, where the data connection includes a circuit-switched data transmission connection between the base station and transcoder unit to exchange speech parameters transmitted as TRAU frames, the data connection also being partly composed of a packet-switched data transmission connection to improve the data transmission connection, a converter is arranged on the data transmission connection between the base station and the transcoder unit for converting a TRAU frame received from the circuit-switched data transmission connection into a packet to be transmitted over the packet-switched connection, wherein the conversion includes a removal of bits included in the TRAU frame which are not needed on the packet-switched connection.

Abstract: In some call-switching systems, transmitted speech samples may be sent back (back-looping) to the sender from a telephone switching center if the switching center is unable to send the speech samples forward. When supplementary information is sent with the speech sample, back-looping may cause a problem if the sending device interprets the signal as originating from another device. To avoid this problem, transmitted samples are subjected to a check value calucation such as a checksum or a cyclic redundancy check. The check value is stored in memory and any received samples are subjected to a check value calculation and compared to the stored samples. If the comparison yields a match, then the device will know that the received samples originated from itself and can be properly handled.

Abstract: A method and device for eliminating acoustic echo in a digital mobile communications system, which system includes a mobile station and a fixed mobile network in which speech signal transmission occurs on a radio path in an uplink direction from the mobile station to the fixed mobile network, and in a downlink direction from the fixed mobile network to the mobile station, and in which a speech coding method is employed on the radio path.