Telefonaktiebolaget Lm Ericsson (publ)

Netsocket, Inc.

Jim Sundqvist

Anders Nohlgren

Fredrik Klas Jansson

Krister Svanbro

Erik Ekudden

Joakim Norrgard

Olov Schelén

Ulf Bodin

Olov Schelen

Lars Westberg

Hans Hannu

Karl Hellwig

Jonas Svedberg

Tomas Frankkila

Magnus Westerlund

Vicknesan Ayadurai

Tomas Svensson

Lars-Erik Jonsson

Tor Bjorn Minde

Bogdan Timus

Anders Tomas Uvliden

Joachim Johansson

Morgan Lindqvist

Zsolt Haraszti

Emil Svanberg

Anders Larsson

Joakim Norrgård

Håkan Lennestål

Fredrik Pettersson

Joakim Norrgáard

Olov Schelèn

Henrik Bergquist

Daniel Lindholm

Erik Lundgren

Wei Wang

An IP network includes a network resource manager having a resource utilization map adapted to manage network resources in an IP network and an application framework having elements for receiving a request for a multicast distribution from a client in the IP network, and elements for requesting network resources from a network resource manager for a Media Quick Start, to start the requested multicast distribution. Further, the network resource manager includes elements for providing the application framework with feedback information relating to network resource availability from the network resource manager, and the application framework includes elements for receiving the feedback information and elements for allowing the Media Quick start or to use another behavior to start the requested multicast distribution based on the received feedback information.

Method and arrangements in an IP network

A method for performing admission control in order to offer assurances on forwarding quality in networks comprising setting a threshold for each link where the threshold defines a maximum sum of forwarding resources requested by applications for their application data flows, ADFs, on the link. The level of the threshold is chosen by utilising knowledge about multiplexing properties of the ADFs on each link and by utilising knowledge about the forwarding resources of the links.

Method and arrangement in a communication system

A method and an arrangement in an IP network provides a scalable solution for reserving resources to obtain a predictable QoS end-to-end in a heterogeneous IP network. The object is achieved by categorizing the destination domain with a domain property label that for example informs about the availability of resources in the destination domain and about how to obtain QoS to the endpoint in this particular domain category.

Method and arrangement for reserving resources to obtain a predetermined quality of service in an IP network

A data network, method and a computer program product, wherein the data network is implemented by a first network level () having a first addressing scheme and at least a second network level () having a second addressing scheme. Each network level provides connectivity over at least one network domain. A first group of Network Resource Managers, NRMs, (b-d)) is arranged to control the resources of the first network level and a second group of NRMs (e-g) is arranged to control the resources of the second network level. The NRMs of the first group (b-d) and second group (e-g) exchange resource requests by using the first addressing scheme and the NRMs (e-g) of the second group perform an address mapping between the first and second addressing schemes.

Arrangements and method for hierarchical resource management in a layered network architecture

A method and arrangement in a communications network. The object is to provide a way of handing recourse management issues and admission control within an IP telephony system. The object is achieved by a topology aware resource manager collecting routing information concerning the IP network, obtaining resource information concerning resources within the IP network, creating a resource map by combing the routing information and resource information, and performing recourse management issues and admission control within the system by the resource map and by interacting with a gatekeeper.

Method for, and a topology aware resource manager in an IP-telephony system

A method for controlling a jitter buffer and a communication apparatus implementing the method and functioning as a first node of a communication system, wherein the jitter butter is adapted to buffer a stream of blocks of compressed speech information generated in a second node in the communication system, transmitted to and received in the first node. Blocks of compressed speech information in said stream of blocks are selected () according to a predetermined rule based on the current buffering delay of the jitter buffer and at least one characteristic of said blocks of compressed speech information. Predetermined actions aiming at reducing buffering delay in the jitter butter are applied () to the selected blocks.

Method and communication apparatus for controlling a jitter buffer

A method for generating speech packets and a communication apparatus implementing the method and functioning as a first node of a communication system. A first stream of digital speech samples having a first sample rate is provided (). If the first sample rate is determined () as not matching a required sample rate, said speech packets are generated () based on a second stream of digital speech samples generated () by performing sample rate conversion of the first stream of digital speech samples.

Method and communication apparatus generation packets after sample rate conversion of speech stream

The present invention relates to a receiver system in a communication system supporting packet-based communication (e.g., an IP-network), including a receiver, speech decoder and a jitter buffer for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller is provided for keeping information about the functional size of the jitter buffer and for providing the speech decoder with control information, such that the speech decoder, based on that information, provides a dynamic adaptation of the size of the jitter buffer using the received encoded, packetized speech signal. The invention also relates to a method of adapting the function size of the jitter buffer of a receiver system.

System for handling variations in the reception of a speech signal consisting of packets

Methods and apparatus are provided for reserving resources in a wireline network from a wireless network. A resource reservation request is made by a service client to a service broker. The service broker contacts a bandwidth broker to determine the available resources in the wireline network. The service broker also contacts a geographical domain server for information related to wireless access for the requested reservation. Using information provided by the geographical domain server, the service broker contacts a radio bearer broker to determine the resources available in a particular wireless network. Based upon the information provided by bandwidth brokers and radio bearer brokers, the service client can reserve the requested resources.

Method and apparatus for network service reservations over wireless access networks

Audio artifacts due to overrun or underrun in a playout buffer caused by the sampling rates at a sending and receiving side not being at the same rate are reduced. An LPC-residual is modified on a sample-by-sample basis. The LPC-residual block, which includes N samples, is converted to a block comprising N+1 or N&#x2212;1 samples. A sample rate controller decides whether samples should be added to or removed from the LPC-residual. The exact position at which to add respective remove samples is either chosen arbitrarily or found by searching for low energy segments in the LPC-residual. A speech synthesiser module then reproduces the speech. By using the proposed sample rate conversion method the playout buffer can be continuously controlled. Furthermore, since the method works on a sample-by-sample basis the buffer can be kept to a minimum and hence no extra delay is introduced.

Method and apparatus in a telecommunications system

The invention is concerned with a method and an apparatus, wherein information data is sent between at least two transceivers in a telecommunication system. The information data is transmitted from the sender of a transceiver to the receiver of one or more other transceivers in form of digital signals having a given sampling frequency. The signals are played out by the receiver in a controlled way. The invention is mainly characterized by estimation of the sender's sampling rate at the sending side of a transceiver, transmitting the estimation to the receiving side of an another transceiver, and controlling the playout of the information data at the receiving side by means of the sampling rate estimated at the sending side to avoid delays and/or interrupts in the presentation. The invention is especially suitable in connection with packet based networks wherein the information data is sent between the transceivers in the telecommunication system in form of packet data frames, such as audio frames.

Method for sending information in a telecommunication system

An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such &ldquo;interaction,&rdquo; for instance, may take the form of updating states in one model using the other model.

Forward error correction in speech coding

A decoder improves delayed packet concealment in a packet network by using two decoder sections. A first decoder section bases its decoding during the concealment phase on erroneous filter states and a set of speech parameters, whereas a second decoder section bases its decoding on saved and updated filter states and the same speech parameters. The outputs of the two decoder sections are thereafter combined to form the final speech signal. This decoding strategy produces a speech signal with smooth transitions from delayed to non-delayed packets and uses information from the most recent packets for speech generation.

Delayed packet concealment method and apparatus

A source encoder schedules its access to a network IP stack. During that scheduled period, the source encoder blocks the transport of non-real-time TCP and UDP packets. After at least one real-time packet has been transported through the stack, the source encoder allows the non-real-time packets to access the physical layer until the beginning of the next scheduled period. Consequently, the packet transport delay is reduced significantly, which results in a substantial increase in throughput.

Internet protocol stack for real time applications

In packet communication paths that include header compression, header fields of packets to be communicated are altered. The alteration of the header fields does not disturb their functionality, and is transparent to the header compression scheme of the packet communication path. The altered header fields are provided for compression by the header compression scheme, resulting in improved performance of the header compression scheme. Performance improvements can also be achieved in packet communication paths that do not use header compression, by violating the integrity of header fields in packets to be transmitted over the packet communication path.

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Luleå, Sweden

Jim Sundqvist