Abstract: There is provided a method in a packet based network system for node-to-node transmission of data packets comprising timing packets and non-timing packets, which is directed to a mechanism for providing a delay variation compensation in a timing system or timing sensitive signal transport in a packet based network without participating in the timing signaling of the timing packets or timing sensitive packets themselves. The method comprises associating the data packets with different levels of transmission priority Pr, Pl, assigning highest (or highest available) transmission priority Pr to the timing packets, separately queuing the timing packets in different buffers 401, 402, and providing first opportunity transmission of the timing packets regardless of transmission priority level of non-timing packets waiting to be transmitted. The advantage of the method is that timing-sensitive traffic thereby experiences reduced buffer delay variations.

Abstract: There is provided a method for providing edge node fault management and bandwidth, BW, control for transmission of a data stream in a packet-based network. The method comprises adaptively transmitting a data stream DS by controlling an outgoing data stream being transmitted to one or more client devices by analyzing at least one data stream characteristics of a corresponding received data stream of the one or more receiving client devices and providing a transmission strategy for the outgoing data stream based on said analyzed data stream characteristics. Thereby accurate adjustment of consumed BW and an associated recovery scheme based on applying adaptive bitrate, ABR, to maintain highest achievable performance when it comes to video quality and also affect on other services, are provided.

Abstract: There is provided a method in a communication network system of e.g. IP type for live distribution of media content capable of node-to-node time-transfer for synchronizing a respective local clock of an originating server and at least one client device or other server by means of two-way time transfer. The media content is sent as a data stream using unicast or multicast via respective communication links over a network. The method comprises employing a mode in which two-way time transfer is selectively provided, selectively meaning that two-way time transfer is employed such that the amount of per client specific time information being exchanged by the server and client devices is controlled, e.g. by time limiting the two-way time transfer, or during that mode reducing the occurrence of full two-way time transfer based synchronization of the client devices.

Abstract: There is provided a method and corresponding node and system for playout buffering and retransmission which can handle Over-The-Top (OTT) and Video-On-Demand (VOD) distribution to a large number of client devices, e.g. a server serving a number of client devices with e.g. the same TV-channel. The presented method comprises transmitting a data stream DS from a server 101 to at least two client devices 151-153, where a predetermined portion of client shared data of the data stream is buffered in a shared buffer. Upon receiving per client requests for retransmission of data from the client devices, requested data is retrieved from the shared buffer and retransmitted to the requesting client device. The method further comprises removing buffered data from said shared buffer based on if a time of interest for the buffered data has passed. The inventive concept is applicable in distribution of e.g. media data as in typical OTT and VOD, which is transported over unicast.

Abstract: There is provided a mechanism for providing Adaptive Bit Rate adjustments of media streams like video for live Over the Top distribution without the need to monitor buffer fill levels, which is advantageous for live video distribution over IP networks. An embodiment of the invention is based on monitoring change in the received bitrate level at a client device, and providing a predicted future bandwidth characteristics of the network based on the change. By monitoring the received bitrate and its rate of change over time, a prediction of (near future) available bandwidth in the network is advantageously provided. A drastic drop of the received bitrate may for instance indicate that the available bandwidth will become too low to sustain a currently received media stream, and the system can accordingly initiate a decision to request a lower video quality.

Abstract: There is provided a method for providing edge node (110) fault management and bandwidth, (BW), control for transmission of a data stream in a packet-based network. The method comprises adaptively transmitting a data stream (DS) by controlling an outgoing data stream being transmitted to one or more client devices (151, 152, 153) by analyzing at least one data stream characteristics of a corresponding received data stream (DS1, DS2, DS3) of the one or more receiving client devices (151, 152, 153), and providing a transmission strategy for the outgoing data stream based on said analyzed data stream characteristics. Thereby accurate adjustment of consumed (BW) and an associated recovery scheme, i.e. FEC and/or retransmission, to maintain highest achievable performance when it comes to video quality and also affect on other services, are provided.

Abstract: There is provided a method in a distribution network (200), comprising a central master node 101 for distributing a received media stream DO comprising packets of a first type over the distribution network to a multiple of client devices 151,152,153. The method comprising providing a common master-client shared time reference Tref, and determining a mean intermediate arrival time for received packets of the first type, which is utilized for providing playout time information for the packets together with the time reference Tref.

Abstract: There is provided a mechanism for providing Adaptive Bit Rate adjustments of media streams like video for live Over the Top distribution without the need to monitor buffer fill levels, which is advantageous for live video distribution over IP networks. An embodiment of the invention is based on monitoring change in the received bitrate level at a client device, and providing a predicted future bandwidth characteristics of the network based on the change. By monitoring the received bitrate and its rate of change over time, a prediction of (near future) available bandwidth in the network is advantageously provided. A drastic drop of the received bitrate may for instance indicate that the available bandwidth will become too low to sustain a currently received media stream, and the system can accordingly initiate a decision to request a lower video quality.

Abstract: A method to provide a cyber attack resistant and fault tolerant precision clocking scheme for wide area critical infrastructure networks through what is called Distributed Time Source Validation, DTSV, is provided, which is a distributed algorithm and signaling mechanism for a network to detect a compromised time source or sources in a multiple master clock system. The method includes providing a local clock signal, receiving in a node R1 an external clock signal from an external source, C or S1, estimating based on the local clock signal and the external clock signal timing parameters associated with the first node and the external source, comparing the timing parameters to detect any Mutual Clock Discrepancy (MCD) between the first node and the external source, and distributing any detected MCD in the network.

Abstract: There is provided an improved redundancy method for a communication network. The invention is based on the insight that a hitless failover can be performed by at the receiving node using a single common indexed play-out buffer in which a selective reconstruction of the transported data stream utilizing individual pieces, i.e. frames or packages, of a data stream received over multiple links is performed.

Abstract: Example embodiments provide a compression method for TDM frames transported over a packet network. The method is based on mapping of allocated TDM slots but not all un-allocated slots into a transport structure, which thereby reduces the needed amount of data transported over packets and hence reduces the required packet rate and packet capacity during transport of the TDM frame. The mapping is further arranged to allow dynamic changes of packet rate and packet capacity, thus the mapping is performed such that a working size of the transport TDM frame is allowed to increase or decrease over time.

Abstract: There is provided a node for facilitating time distribution in communication networks, and more specifically for time synchronization in digital television (DTV) distribution network. The node comprises an interface, a clock for establishing a local time, and a time-locked loop. The interface is configured for interconnecting the node to at least one neighboring node over an isochronous transport link for transmission and reception of repetitive frames comprising time information. The time-locked loop is configured for, based on remote time information received via the interface and local time information from the clock, synchronizing the clock to the clock of one of the at least one neighboring node. This facilitates that the node, or a corresponding synchronous network comprising nodes according to the inventive concept, is rather insensitive to network delays. In this way the requirements on the network infrastructure are reduced. In particular, there is no need for dedicated networks.

Abstract: There is provided an improved redundancy method for a communication network. The invention is based on the insight that a hitless failover can be performed by at the receiving node using a single common indexed play-out buffer in which a selective reconstruction of the transported data stream utilizing individual pieces, i.e. frames or packages, of a data stream received over multiple links is performed.

Abstract: There is provided a node for facilitating time distribution in communication networks, and more specifically for time synchronization in digital television (DTV) distribution network. The node comprises an interface, a clock for establishing a local time, and a time-locked loop. The interface is configured for interconnecting the node to at least one neighboring node over an isochronous transport link for transmission and reception of repetitive frames comprising time information. The time-locked loop is configured for, based on remote time information received via the interface and local time information from the clock, synchronizing the clock to the clock of one of the at least one neighboring node. This facilitates that the node, or a corresponding synchronous network comprising nodes according to the inventive concept, is rather insensitive to network delays. In this way the requirements on the network infrastructure are reduced. In particular, there is no need for dedicated networks.

Abstract: The present invention relates to methods and apparatuses for allocating time slots to circuit-switched channels established to comprise one or more respective time slots in a recurrent frame of a time division multiplexed network. According to the invention, a time slot allocated to said channel is associated with a selected level, of at least two available levels of priority. Decisions as to whether or not to deallocate said time slot from said channel is then based upon a comparison of said selected level of priority and a level of priority associated with a request for a time slot for another channel.