Abstract: The invention relates to a technique for controlling establishment of a tandem free operation (TFO) for transfer of TFO frames including content frames between a first and a second transcoder unit without transcoding of the content frames. A method aspect of the technique comprises the steps of transmitting, in case a remotely used codec does not meet a codec compatibility condition, a first transparent mode message indicating to in-path equipment a transparent mode for subsequent transparent TFO negotiation, and transmitting, in the transparent mode, a TFO configuration frame indicating a local TFO codec list.

Abstract: The invention relates to a technique of operating a call control node controlling at least one section of a call path. The call path includes between two opposite edge nodes a multi-section harmonization path along which codec selection is to be harmonized. A method embodiment of the technique, wherein the call control node is a transfer node in the harmonization path between the edge nodes, comprises the steps of determining if the call control node is a transfer node of the harmonization path; determining if a codec used for the at least one section controlled by the call control node fulfils a predefined harmonization criterion; and providing, in case the used codec does not fulfill the harmonization criterion, a harmonization trigger indication to at least one of the edge nodes of the harmonization path for initiating harmonization.

Abstract: A mobile station receives data for sending to a destination device and determines whether to send the data to the destination device via either short message service (SMS) over service access point identifier 0 (SAPI-0) or SMS over SAPI-3, where SAPI-0 and SAPI-3 comprise different service access points associated with a data link layer of a mobile network. The mobile station selectively sends the location information to the destination device via either SMS over SAPI-0, or via SMS over SAPI-3, based on the determination.

Abstract: The invention is related to a method for transmitting data in a telecommunications network and a device, e.g. a network node, utilizing that method, where said data transfer requires the application of algorithms for coding and/or decoding (“codec”) on said data, where at least one of said network nodes provides a list (25) of available codecs and transmits it to a further network node, where said further network node chooses from the received list (25) and from a corresponding set of codecs available to the further network node a combination of codecs to be used when transferring said data, and where the received list (25) not only comprises direct codecs (28) but transcoding codecs (29), i.e. codecs requiring transcoding, as well.

Abstract: In order to simplify the setup of telephone conference with distributed conference devices, each conference participant applies speech encoding to its own speech signal in order to reduce the bandwidth of its speech signal thus that the bandwidth compressed speech signal shows at least half of the bandwidth of the data transport channel. Each compressed speech signal is inserted to the data transport channel and that each conference participant extracts the compressed speech signals of the other parties from the data transport channel. Each participant then decodes the compressed speech signals and each conference participant sums up all decoded speech signals to a sum signal.

Abstract: Techniques for discontinuous transmission (DTX) and fast in-band signaling of configuration changes and protocol messages in speech communications systems provide cost efficiency in terms of radio transmission capacity, in terms of fixed line transmission, and in terms of implementation effort. An exemplary method for performing discontinuous transmission (DTX) in a communications system in which source data is interleaved for transmission from a first component in the system to a second component in the system includes the steps of detecting periods of source data inactivity, and transmitting silence descriptor (SID) frames from the first to the second component during the periods of source data inactivity, certain of the transmitted SID frames being interleaved using a different interleaving algorithm as compared to that used for source data. For example, the source data can be block diagonally interleaved, and certain of the SID frames can be block interleaved.

Abstract: A method of modifying a call setup message for a call from a first terminal to a second terminal, the call setup message being sent from the first terminal to a network node via an intermediate node which handles at least some signaling or traffic associated with the call, wherein the network node is responsible for negotiating call setup requirements for the call to the second terminal, the method comprising the steps of: intercepting the call setup message at the intermediate node; identifying one or more preferences or capabilities associated with the intermediate node which may impact the way in which the intermediate node can handle signaling or traffic associated with the call; modifying the call setup message based on information relating to said preferences or capabilities; and sending the modified call setup message to the network node, in order that the node can effect said negotiations based on the modified message.

Abstract: A method and system are disclosed for call case controlled block error rate (BLER) or bit error rate (BER) target setting in WCDMA and other CDMA technologies for Mobile to Mobile (MTM) and Mobile to Public Switched Telephone Network (MTPSTN) communication systems, whereby first an overall BLER/BER target value is defined.

Abstract: In a radio communication system in a mobile to mobile call with one good (AMR102) and bad (AMR515) radio link, the good radio (AMR102) link is forced by the poor (AMR515) link to use a more robust AMR coded mode (AMR515) and thereby using excessive power (232). A capacity loss in such system is avoided by adjusting the power level (212) for the connection with the good link (AMR102).

Abstract: The invention is related to a method for transmitting data in a telecommunications network and a device, e.g. a network node, utilising that method, where said data transfer requires the application of algorithms for coding and/or decoding (“codec”) on said data, where at least one of said network nodes provides a list (25) of available codecs and transmits it to a further network node, where said further network node chooses from the received list (25) and from a corresponding set of codecs available to the further network node a combination of codecs to be used when transferring said data, and where the received list (25) not only comprises direct codecs (28) but transcoding codecs (29), i.e. codecs requiring transcoding, as well.

Abstract: A speech analyzing stage (12) and a method for analyzing a speech signal is described. The speech analyzing stage (12) is part of an automatic speech recognition system (10) and is adapted for analyzing in the spectral domain a speech signal sampled at one of at least two different system sampling rates. The speech analyzing stage (12) comprises a first spectral analyzer (18a) for analyzing the speech signal up to a first frequency (flowest) which is preferably derived from the lowest system sampling rate (2×flowest) and a second spectral analyzer (18b) for analyzing the speech signal at least above the first frequency (flowest).

Abstract: A method and a device for initiating the bypassing of a pair of transcoding operations that are performed in series by a first transcoder arranged together with a first communication terminal on a local side of a communication network and by a second transcoder arranged together with a second communication terminal on a distant side of the communication network are described. The method includes receiving from the distant side information about an encoding format currently in use on the distant side and about encoding capabilities of the distant side and transmitting to the distant side information about an encoding format currently in use on the local side and about encoding capabilities of the local side to enable on one or on both sides a change of the encoding format currently in use prior to initiating the bypassing of the transcoding operations.

Abstract: In a radio communication system in a mobile to mobile call with one good (AMR102) and bad (AMR515) radio link, the good radio (AMR102) link is forced by the poor (AMR515) link to use a more robust AMR coded mode (AMR515) and thereby using excessive power (232). A capacity loss in such system is avoided by adjusting the power level (212) for the connection with the good link (AMR102).

Abstract: A method and system for communicating information includes evaluation regarding routing of calls performed at a call control point in an ITSP and/or wireless network. A PIC identity associated with another party's preferences is acquired and sent to the call control point. The PIC identity identifies the type of the carrier network (e.g., circuit switched or VoIP) and is used to make further routing decisions. If a circuit-switched carrier is identified, the IP homing leg is terminated at the voice gateway in the recipient's HPLMN and information is sent over to the GMSC where normal, circuit-switched routing procedures. If a VoIP carrier is identified then the called user's roaming number is retrieved from the HLR, and the call is further routed directly over the IP domain towards a voice gateway at the visited network minimizing the number of transcodings. Additional transcoding steps can be avoided if a single encoding is agreed upon according to “tandem free operation” (TFO).

Abstract: The present invention relates to an improved transcoding apparatus (1) for use in a switching network of a telecommunication system, said transcoding apparatus (1) including a) a plurality of transcoding units for source encoding and decoding data, for example speech data, wherein at least one transcoding unit (11) of said plurality is capable of operating in tandem-free operation mode, b) switching means (12) adapted to switch data through said plurality of transcoding units, c) a transcoder controller (13) for controlling said switching means (12) and said plurality of trnascoding units, wherein said transcoder controller (13) is adapted to d) instruct said switching means (12) to insert one of said at least one transcoding unit (11) into a data path associated with a connection between a mobile terminal of said telecommunicaiton system and said switching network, and wherein said transcoder controller (13) is adapted to e) instruct said one of said at least one transcoding unit (11) to operate in tandem-fr

Abstract: The invention provides method and apparatus for transporting tone signalling information through a GCME link in a GSM network supporting TFO and speech compression modes for the passage of signals through the link, respectively using TFO frames and TRAU-like In a data stream to the local entrance to the GCME link, a tone signal to be transported is detected. The detected tone signal is coded into bits of a TFO or TRAU-like frame for sending through the GCME link, including setting bits to mark the frame as a containing tone signalling information and setting bits to identify the detected tone signal, whereby the tone signalling information is transported in-band through the GCME link as a marked TFO or TRAU-like frame. At the remote exit from the GCME link, the marked frame is detected and the tone signal regenerated in dependence upon the bits of the marked frame.

Abstract: In a method for the establishment of a connection between a first node (RNC1) and a second node (RNC2), both are connectable to a core network which comprises interconnected core network nodes (CN1, CN2). The first node (RNC1) signals a call request to a first core network node (CN1). If the first node is an access node, it sends an initialization according to a framing protocol, wherein a set of parameters for the framing of information sent over the interface between the first access node (RNC1) and the first core network node (CN1). If the first node is no access node, a first control server (MSC1) defines said set of parameters. The set of parameters is transmitted to the first core network node (CN1) and the first core network node (CN1) stores the parameter set. The first core network node (CN1) initializes the connection (Co) to a further core network node according to said protocol and the further core network node stores the parameter set.

Abstract: Techniques for discontinuous transmission (DTX) and fast in-band signaling of configuration changes and protocol messages in speech communications systems provide cost efficiency in terms of radio transmission capacity, in terms of fixed line transmission, and in terms of implementation effort. An exemplary method for performing discontinuous transmission (DTX) in a communications system in which source data is interleaved for transmission from a first component in the system to a second component in the system includes the steps of detecting periods of source data inactivity, and transmitting silence descriptor (SID) frames from the first to the second component during the periods of source data inactivity, certain of the transmitted SID frames being interleaved using a different interleaving algorithm as compared to that used for source data. For example, the source data can be block diagonally interleaved, and certain of the SID frames can be block interleaved.

Abstract: Techniques and systems for identifying coding rates of transmitted frames are described. Unused bits in rate adapted frames are used to carry frame type indicator patterns. Maximal rate frames (i.e., with a highest coding rate) need not include a frame type indicator.

Abstract: A new modem standard provides reliable in-band transmission of TTY/TDD signals via the speech channel of all digital cellular phone systems. The standard includes error correction such as Forward Error Correction (FEC), and also performs interleaving to spread each TTY/TDD character over several adjacent speech frames. The modem standard can be applied to all speech coding standards and all mobile phone systems. The standard can be implemented using a special interface cable or box between a conventional TTY/TDD device and a mobile phone. The interface can also be integrated directly into new mobile phones, and necessary signal processing can be performed by the mobile's DSP. A new Universal TTY/TDD device can also be provided that supports both modem standards: conventional Baudot code as well as the new modem signals, and implemented using specialized hardware, or using conventional hardware such as a laptop computer with specialized software.