Abstract: In a method of improved media frame transmission in a communication network. Initially a plurality of “original” or regular media frames are provided for transmission. According to the invention, robust representations of the provided regular media frames are generated and stored locally. Subsequently, one or more of the regular media frames is/are transmitted. The invention detects an indication of a loss of a transmitted media frame, and the idea is to transmit, in response to a detected frame loss, a stored robust representation of the lost media frame and/or a stored robust representation of a subsequent, not yet transmitted, media frame to increase the media quality.

Abstract: User equipment in a wireless communication system monitors scheduling information and locally detects a change in link data rate of the uplink channel based on the monitored scheduling information. In this way, a change in link data rate can be detected directly without significant delay. This direct or early detection of a rate change is then combined with an appropriate system reaction. The information of the detected change in link data rate is preferably utilized for adapting the application data rate of an IP application running in the user equipment. As an alternative, or as a complement, data packets are classified based on relative importance and selected for transfer of information over the uplink channel based on the classification of data packets and in dependence on the detected change in link data rate.

Abstract: The present invention sends multiple versions of a multimedia packet to the base station, and, based on the radio channel and traffic characteristics, an appropriate version of the multimedia packet is selected to send to the mobile station at a given time. In this way, source transmission is improved to instantaneous conditions. The steps of the present invention are performed in conjunction with RTP used for multimedia transmission over internet protocol (IP) networks. In a first embodiment, the multiple versions are sent to the base station in the same RTP packet, and the base station strips out the extraneous versions. In a second embodiment, the base station receives multiple RTP packets having identical information in the packet header in many fields, and selects an appropriate one among these for transmission to the mobile station, discarding the rest.

Abstract: An apparatus for controlling utilization of RTCP bandwidth for compound and non-compound RTCP packets is described. This apparatus includes a bandwidth detector determining available RTCP bandwidth and a channel quality detector determining channel quality. A bandwidth divider connected to the bandwidth detector and the channel quality detector divides the available RTCP bandwidth between compound and non-compound RTCP packets based on the determined channel quality.

Abstract: A method is proposed for determining the round-trip delay for sending multimedia data, such as voice or video data, between a first network node and a second network node over a communications network, where data is encapsulated in data units comprising at least one frame of encoded data. The method includes the steps of: said first network node sending a request for reconfiguration of the data unit structure to the second node; the first node detecting a reconfiguration response from the second node, where the reconfiguration response comprises at least one data unit of data having a structure that corresponds to the request for reconfiguration. The first node then measures the time elapsed between sending the reconfiguration request and detecting the reconfiguration response and determines a value for roundtrip delay using said measured time. In accordance with a first embodiment, the request for reconfiguration is a request to change the encoding mode of the multimedia data.

Abstract: In a method of improved session negotiation between first and second clients in a cellular telecommunication system, the two clients negotiating S10 and agreeing upon a first codec type for a session. The first and second clients then initiating the session and exchanging S20 media data frames according to the first codec type. Subsequently, during the initiated session, at least one of the first and second clients providing S30 an indication for a second codec type in at least one subsequent media data frame. Finally, upon receiving and recognizing said indication, the other of said first and second clients switching S40 to the indicated second codec type in a next media data frame, thereby enabling the first and second clients to exchange subsequent media frames utilizing the second codec type.

Abstract: The present invention sends multiple versions of a multimedia packet to the base station, and, based on the radio channel and traffic characteristics, an appropriate version of the multimedia packet is selected to send to the mobile station at a given time. In this way, source transmission is improved to instantaneous conditions. The steps of the present invention are performed in conjunction with RTP used for multimedia transmission over internet protocol (IP) networks. In a first embodiment, the multiple versions are sent to the base station in the same RTP packet, and the base station strips out the extraneous versions. In a second embodiment, the base station receives multiple RTP packets having identical information in the packet header in many fields, and selects an appropriate one among these for transmission to the mobile station, discarding the rest.

Abstract: A packet scheduler reduces or “compresses” the packet transmission delay jitter or delay range where packets experience little or no scheduling delay before transmission. As a result, the number of packets that experience little or no delay is reduced. A preferred example way of compressing the packet transmission delay jitter is to reduce the transmission priority of low delay packets. Compressing the delay jitter is particularly desirable for services like VoIP that require low packet transmission delay jitter.

Abstract: In a method of providing a backward and forward compatible speech codec payload format, the following steps are included: providing S10 an RTP package; including S20 payload according to a first codec into the provided RTP package, and appending S50 payload according to a second codec into the provided RTP package. In addition, at least one unused bit is located S30 in the included first codec payload, and the located at least one unused bit is designated S40 as a codec compatibility bit. Finally, the designated at least one codec compatibility bit is utilized S60 to provide an indication of the presence of the appended second codec payload.

Abstract: A control logic means preferably for a receiver comprising a jitter buffer means adapted to receive and buffer incoming frames or packets and to extract data frames from the received packets, a decoder connected to the jitter buffer means adapted to decode the extracted data frames, and a time scaling means connected to the decoder adapted to play out decoded speech frames adaptively. The control logic means comprises knowledge of whether a state recovery function is available and is adapted to retrieve at least one parameter from at least one of the jitter buffer means, the time scaling means, and the decoder, to adaptively control at least one of an initial buffering time of said jitter buffer means, the knowledge of the availability of the state recovery function, and a time scaling amount of said time scaling means from the time scaling means or the decoder.

Abstract: The present invention is to select an adaptation scheme for the transmission of the encoded media that results in a satisfactory performance of the transmitted encoded media. A difference from the prior art is that each adaptation scheme defines a set of different transmission formats, wherein each transmission formats is a combination of at least two of the parameters the source codec bit rate, the packet rate, the number of frames of each packet (referred to as frame aggregation), and the level of redundancy. By using the different transmission formats, the transmission can be adapted to different operating scenarios and the performance is hence improved.

Abstract: It is a basic idea to determine the characteristics of a jitter profile measured over a plurality of packets, and classify the jitter profile based on the determined characteristics as well as information representative of the particular access used for communication between the sender and the receiver. The classified jitter profile is then matched to an appropriate action for media layer adaptation so that a proper action for media layer adaptation can be initiated.

Abstract: A receiver includes a detector for detecting a change in source of incoming media during an on-going communication session, and means to provide a reset signal in order to reset decoder states of a decoder in response to such a detected change before decoding new incoming media. In this way, a state mismatch can be avoided without the need for several active decoder instances in the receiver, leading to substantial savings with respect to overall complexity, memory usage and power consumption. This also means that media distortions can be eliminated or at least reduced when the decoded media is finally rendered by a player.

Abstract: An apparatus for controlling utilization of RTCP bandwidth for compound and non-compound RTCP packets is described. This apparatus includes a bandwidth detector determining available RTCP bandwidth and a channel quality detector determining channel quality. A bandwidth divider connected to the bandwidth detector and the channel quality detector divides the available RTCP bandwidth between compound and non-compound RTCP packets based on the determined channel quality.

Abstract: In order to efficiently handle the switch between user media and announcement media, a basic step is to first determine a configuration of the user media. Next, a configuration of the announcement media to be presented is determined based on the determined user media configuration. Subsequently, the announcement media is configured according to the announcement media configuration, and the configured announcement media is finally sent to the intended user. In this way, the overall appearance or sound of the announcement will be virtually the same as or at least similar to the overall appearance or sound of the user media, preferably without distortions. This allows the user to perceive the announcement as clearly as possible.

Abstract: In a method of improved media frame transmission in a communication network. Initially a plurality of “original” or regular media frames are provided for transmission. According to the invention, robust representations of the provided regular media frames are generated and stored locally. Subsequently, one or more of the regular media frames is/are transmitted. The invention detects an indication of a loss of a transmitted media frame, and the idea is to transmit, in response to a detected frame loss, a stored robust representation of the lost media frame and/or a stored robust representation of a subsequent, not yet transmitted, media frame to increase the media quality.

Abstract: The present invention sends multiple versions of a multimedia packet to the base station, and, based on the radio channel and traffic characteristics, an appropriate version of the multimedia packet is selected to send to the mobile station at a given time. In this way, source transmission is improved to instantaneous conditions. The steps of the present invention are performed in conjunction with RTP used for multimedia transmission over internet protocol (IP) networks. In a first embodiment, the multiple versions are sent to the base station in the same RTP packet, and the base station strips out the extraneous versions. In a second embodiment, the base station receives multiple RTP packets having identical information in the packet header in many fields, and selects an appropriate one among these for transmission to the mobile station, discarding the rest.

Abstract: User equipment in a wireless communication system monitors scheduling information and locally detects a change in link data rate of the uplink channel based on the monitored scheduling information. In this way, a change in link data rate can be detected directly without significant delay. This direct or early detection of a rate change is then combined with an appropriate system reaction. The information of the detected change in link data rate is preferably utilized for adapting the application data rate of an IP application running in the user equipment. As an alternative, or as a complement, data packets are classified based on relative importance and selected for transfer of information over the uplink channel based on the classification of data packets and in dependence on the detected change in link data rate.

Abstract: A control logic means preferably for a receiver comprising a jitter buffer means adapted to receive and buffer incoming frames or packets and to extract data frames from the received packets, a decoder connected to the jitter buffer means adapted to decode the extracted data frames, and a time scaling means connected to the decoder adapted to play out decoded speech frames adaptively. The control logic means comprises knowledge of whether a state recovery function is available and is adapted to retrieve at least one parameter from at least one of the jitter buffer means, the time scaling means, and the decoder, to adaptively control at least one of an initial buffering time of said jitter buffer means, the knowledge of the availability of the state recovery function, and a time scaling amount of said time scaling means from the time scaling means or the decoder.

Abstract: A packet scheduler reduces or “compresses” the packet transmission delay jitter or delay range where packets experience little or no scheduling delay before transmission. As a result, the number of packets that experience little or no delay is reduced. A preferred example way of compressing the packet transmission delay jitter is to reduce the transmission priority of low delay packets. Compressing the delay jitter is particularly desirable for services like VoIP that require low packet transmission delay jitter.