Abstract: This invention relates to maintaining or re-establishing a communications session in a wireless communications system. The present invention provides a method of re-routing a call between a user device and a third party device, the method comprising: setting up a call with the third party device using a first wireless connection between the user device and an access point; monitoring a quality parameter of the first wireless connection at the access point; automatically re-routing the call from the third party device to the user via a second wireless connection between the user device and a base station in response to determining that the quality parameter of the first wireless connection has fallen below a predetermined threshold.

Abstract: In order to keep a pathway for time critical data packets (for example for VoIP calls) a dummy flow of data packets is established between a home hub router (20) and a broadband remote access server (25) being the first gateway to the IP network. Should a call be made from the telephone (23), the dummy flow is replaced for the duration of the call by the packets generated for the VoIP call. The dummy flow is adjusted by increasing or decreasing the number of data packets transmitted in order to ensure that a minimum bandwidth having a tolerable delay is available. The dummy flow works by causing non time critical (i.e. P2P data packet flows) to back off and no modification is required to routers and the like incorporated into the network.

Abstract: When a mobile station MS1 determines during communication through a first network (1), that a second network (7) is available that offers better operational parameters for the communication, the mobile station sends a handover announcement (S 5.3) to the first network to offer it an opportunity to improve the operational parameters for the communication through the first network. The first network (1) may in response offer updated operational parameters (S5.6) with a view to encouraging the communication to continue through the first network. A decision (S5.7) whether to make a handover to the second network can then be made by comparing the updated operational parameters for the first network with the operational parameters for the second network.

Abstract: For users to access network services such as video streaming from a device, users usually have to register themselves with the service first. Most registration mechanisms require a user to input a username, password, date of birth and other details. When implemented through a web interface, this mechanism is relatively easy for the service provider to provision. However, the problem is that users are required to manually enter significant amounts of information which can be both time consuming, especially on a mobile device where there usually no QWERTY input device, and susceptible to unintentional errors. The present invention proposes an automated registration process that does not require a user to enter any details manually except for the initial request to subscribe to a service. The process gathers information automatically about the user and the device used, which is then stored and used for user authentication during subsequent service requests following the initial subscription request.

Abstract: Faults resulting in reception of a still, but unknown, frame are recognised by comparing each received frame of the video signal with its predecessor, incrementing a counter in the event that the difference between the frames falls below a threshold; and generating an alarm signal in the event that the count of the counter exceeds a predetermined count. Other types of fault such as loss of signal (i.e. reception of just noise) are recognised by incrementing the counter whenever the difference exceeds a threshold. Similar results may be obtained by instead the monitoring quantisation step size and/or number of transmitted bits of a digitally coded signal, and noting that it falls below, or exceeds, a threshold. A preferred option is to compute a complexity measure, being a monotonically increasing function (e.g the product) of the quantisation step size and of the number of coded bits and compare this with the threshold value.

Abstract: A video sequence (4) is subjected to a signal compression process, in which the video sequence is divided (14) into a plurality of segments such that in each segment the number of bits required to code each frame in uncompressed form falls within a range having a predetermined magnitude, and a quantisation level is selected (17) for the encoding (19) of each segment such that the overall bit rate of the segment corresponds to a predetermined value. This value may be pre-set, or may be set in response to an input from the transmission network (3) or remote decoder (2). The quantisation level Q is determined according to a function of the number of bits per frame R, determined by analysis (10,11,12,13) of the entire sequence prior to transmission.

Abstract: A video stream is digitally encoded such that the rate at which individual segments of data are encoded varies according to the amount of data required to generate each segment. Frames are selectively omitted from transmission (32) such that the cumulative frame rate does not fall below a predetermined value. This process can be used to ensure that the next frame to be displayed is always available in the buffer store 6 associated with the decoder (2). The decoder (2) is arranged to identify where frames have been omitted from the decoded transmission, and to perform a resynchronisation (7) on the decoded stream by comparison between time stamps in the video stream and a corresponding audio stream. Resynchronisation may be performed by extending the durations of individual frames, or by repeating frames.

Abstract: Recorded material such as video is transmitted in compressed form to a receiver, which has a buffer for smoothing differences between the data rate received and that consumed by a decoder that follows. The whole of the recording is analysed to determine a point at which to commence playing such that no buffer underflow can occur; the decoder commences playing only when this point has been reached.