摘要:
A variable gain optical amplifier, in which homogeneous gain broadening is dominant, has first and second fixed gain rare-earth doped optical waveguide amplifiers (51, 53) optically in series together with an intervening variable attenuation optical attenuator (52). This arrangement circumvents the problem of gain tilt encountered when operating such amplifiers under variable gain conditions. An alternative form of the module has variable gain waveguide amplifiers, but these are co-regulated so that the aggregate of their gain at a wavelength within the gain spectrum is maintained constant. A further alternative form of module is employed in a concatenation of such modules. In such a concatenation, the gain of individual modules is allowed to vary, but the aggregate of the gain, at a wavelength within the gain spectrum, of all the waveguide amplifiers of all the modules of the concatenation is maintained constant.
摘要:
A method of monitoring quality of service in communications over a packet-based network, involves transmitting test packets across the network and monitoring transmission characteristics such as packet loss and transmission delay for the test packets. A measure of network performance is then dynamically calculated from the transmission characteristics, andis displayed at the endpoint as a dynamic indication of the network performance.
摘要:
A multi cavity comb filter for interleaving or de-interleaving WDM signals has a plurality of stacked optical cavities each having substantially the same thickness. The multiple cavity arrangement provides a comb reflection response and a comb transmission response with broad peaks, so that the filter can be used for transmitting one group of channels and reflecting another group of channels at interleaved positions. The cavities are preferably formed from silicon wafers, so that existing techniques can be employed to obtain specific cavity thicknesses with sufficient accuracy and uniformity.
摘要:
A method of routing calls which require intelligent network (IN) services between a pair of subscribers, at least one of which subscribers is a mobile network subscriber. A mobile network (30) routes a call to an intelligent network (31) via a first path (32) to apply an IN service, and routes the call back to the mobile network via a second (33) or a third (34) path. Calls routed via the second path (33) are completed to the called party. Calls routed via the third path (34) may require further routing to the IN (31) prior to completion, such as where a different called party is involved or where a terminating IN service may be needed.
摘要:
A network arrangement for delivering IP services to subscribers comprises a core network (10), a plurality of label switched media gateways (22,22a) coupled to the network (10) and each providing an interface for one or more subscriber terminals (25,25a,25b). Call servers (11,11a) associated with the network (10) are used to establish connections between pairs of gateways (22,22a), these connections being routed across the core network (10) via tunnels established therein. The tunnels are exclusively reserved for traffic between the label switched media gateways (22,22a) so as to provide security of that traffic from third party access and to provide a guaranteed quality of service. Because traffic is accepted into a tunnel only if bandwidth is available in that tunnel, firm and meaningful quality of service guarantees can be given to users.
摘要:
Modifications to SIP are made which significantly extend the functionality of SIP for example by allowing a service for automatically setting up multi-media conferences to be easily provided. SIP messages are associated with computer software code such as Java byte code, Java applets or mobile autonomous software agents. An example of a mobile autonomous agent is a Java mobile agent. This computer software code may be contained in the body of a SIP message or an address indicating where the computer software code is located is stored in the SIP message. SIP clients are arranged such that on receipt of a SIP message that has been associated with computer software code, that code is executed by a processor associated with the SIP client. For example, in the case that Java applets are contained in a SIP message these are executed by a Java Virtual Machine associated with the SIP client. If a Java mobile agent is contained in the SIP message this executes on a Java Mobile Agent Virtual Machine associated with the SIP client. In one example, such computer software code must always be executed by the processor associated with the SIP client before that SIP client carries out any other actions related to the SIP message. Preferably an indicator is put into the header of a SIP message to indicate that it has been associated with computer software code, and SIP clients are arranged to detect the presence of such indicators. An application programming interface is created in order that the computer software code may control the SIP client and/or any processor associated with that SIP client. In one example, computer software code is associated with SIP messages in order that a service for automatically setting up multi-media conferences is provided.
摘要:
The present invention relates to the detection of a predetermined sequence in a digital bit-stream, and more particularly a method and apparatus for the fast and efficient detection of a start code sequence. As with many packet based bit-streams, packets are identified through the use of a start code. The start code is a unique sequence which occurs only to indicate the start of a packet, and can never occur in the data portion of a bit-stream. Identifying the start of packets is crucial in the processing of packetised bit-streams. In the field of digital broadcasting, a common format of digital video compression is that of the Moving Picture Expert Group (MPEG). MPEG uses a packetised bit-stream and packets are preceded by a start code to enable individual packets to be identified. In any real-time processing of MPEG bit-streams, it is vital to be able to identify the MPEG start codes as quickly and efficiently as possible. Performing this in hardware is a relatively straightforward operation. Detecting start codes using software is also straightforward in a non real-time situation. However, where an end-to-end real-time software solution is required to process MPEG data it may not be possible or desirable to use a hardware-based solution. The present invention overcomes the problems of the prior art and provides a method and apparatus for the fast and efficient detection of the MPEG start code sequence.
摘要:
The present invention relates to digital data transmission and more precisely a method and apparatus for managing a decoder buffer. In data transmission applications, such as digital video broadcasting, large amounts of data are encoded and transmitted to decoders having a decoder buffer. Management of the decoder buffer is very important as transmittal of too little data can result in the decoder buffer underflowing (becoming empty), or if too much data is transmitted the decoder buffer could overflow. Both decoder buffer overflow and underflow result in loss of data and the decoding process is adversely affected. The present invention provides a method and apparatus for managing a decoder buffer such that decoder buffer overflow and underflow are minimised.
摘要:
The invention relates to multiplexing data packets in a data service channel with data in one or more digital video signal channels to form a multiplexed output signal. The data in the data service channel may include control data, conditional access data, electronic program guides, paged data services, service information, broadcast internet information, and business information such as financial share information. The data packets each comprise a time stamp indicating a requested delivery time and the data packets are sorted into a queue in time stamp order. The urgency of the data service channel is calculated as a function of the queue length and requested delivery times. The share of the bit rate of the multiplexed output signal allocated to the data service channel is varied according to its urgency. An error value is calculated for each data packet to represent the error between the expected delivery time and the requested delivery time to the head of the queue and the urgency of the data channel is derived as an average of the error values. The average may be a weighted average.