FR2934733A1 - Packetized multimedia stream i.e. video data, transporting and routing method for internet, involves receiving and duplicating multimedia streams, and simultaneously transmitting streams towards destinations through packet-switched networks - Google Patents

Abstract

The method involves receiving input packetized multimedia streams i.e. video data, from sources, and duplicating the multimedia streams, irrespective of its format. The multimedia streams are simultaneously transmitted towards destinations through packet-switched networks or by packet switching type telecommunication access links under the form of multimedia signals, where the number of the streams is different for each signal. Independent claims are also included for the following: (1) an equipment for transporting and routing packetized multimedia streams from sources to destinations in real time (2) a system for transporting and routing packetized multimedia streams from sources to destinations in real time.

Description

METHOD AND SYSTEM FOR VIDEO CONTRIBUTION, TRANSMISSION AND DISTRIBUTION OVER IP

The present invention relates to a method and a system for transmitting one or more multimedia signals from one or more sources to one or more destinations over a transmission network. In particular, this method is directed to a method and a system for transporting video data over a communication network using the Internet Protocol (IP). The field of the invention is that of telecommunications and in particular telecommunications applied to the field of multimedia.

The invention applies more particularly to the transport of multimedia data over an Internet type communication network using the IP protocol. The multimedia data concerned are in particular the sound and / or the moving image.

Indeed, broadband over IP is becoming widespread in Europe, Asia, North America and Africa. These are made possible by access methods such as optical fiber, as well as very economical access methods such as ADSL (Asymmetric Digital Subscriber Line in English) or satellite IP for example. We are witnessing, according to the markets, more or less rapidly an increase in access rates. In parallel, the quality of service (QoS or Quality of Service in English) on IP networks improves, allowing the conveyance of isochronous flows such as voice or video. This is especially true in Virtual Private Networks (VPNs or Virtual Private Networks) based on IP and MPLS (Multiprotocol Label Switching) technologies. On the other hand, the QoS mechanisms expected in IPv4 (Internet Protocol version 4) will be generalized and extended in IPv6 (Internet Protocol version 6). RTP stands for Real-time Transport Protocol, transport level protocol (level 4 in the OSI model). Although protocol of the transport layer, this protocol adds a number of application functions, such as the transport of the reference clock for example. It has been designed to carry real-time streams, typically audio and video, over IP networks. Source means the entity that provides a signal corresponding to a content, a chain for example. The signal in question will be called source signal. The term destination is the entity that receives a signal corresponding to a content, a chain for example. The signal in question will be called destination signal. In addition, the word flow designates by extension any flow of data or stream of packets if these data are packaged. The flow fragment word designates a portion of the data or packets of an original stream, possibly encapsulated in another data or packet format. Transport Stream or TS: this is the format commonly used in the field of digital television for the transport of multiplexed audiovisual data streams. The TS format is defined in ISO 13818-1. Reference is made to the notions of protocol layers, for example network layer, transport, applicative by reference to the OSI protocol description model. To date, the state of the art is shown in Figure 2. To distribute their programs, most architectures in place are of the type: for m sources and n destinations, these architectures require mxn permanent links, usually provided on dedicated optical fibers or dedicated radio-relay systems. These architectures are therefore very expensive and do not allow an evolution of the means of distribution of the different sources to the different destinations, especially when there must be a change of formats or protocols between the source and the destination. In the current state of the art, it is however possible to concentrate the streams on a site but the switching is done only in the video domain. On the other hand, no mutualisation of the links is possible and the logic is that of the 1 multimedia signal = 1 link. To overcome these various defects, the method according to the invention allows the transport and real-time routing of packaged multimedia streams from n sources to m destinations, characterized in that it can receive one or more streams of input from one or more sources and duplicate any input stream, regardless of its format, and send it simultaneously to one or more destinations across packet transmission networks. Advantageously, said packetized multimedia streams are: transported through telecom access links of the packet transport type and through one or more extended telecom networks of the same type. Advantageously, the adaptation between the video signal in its electrical form and the paquetized flux or flows representing it is performed at the end of the link on the source side of the specialized gateways. In a particular embodiment, the adaptation between the packetized stream (s) and the video signal in its electrical form that they represent is carried out at the end of the link on the destination side by specialized gateways. Advantageously, said adaptation gives rise to 1 to several paquétisées multimedia streams.

Advantageously, said adaptation is made from 1 to several multimedia streams paquétisés. In another particular embodiment, said packetized streams remain in the packetized form between the source and the destination, without ever returning in the form of a multimedia signal between these two ends. Advantageously, said multimedia streams from different sources or with different destinations may or may not use the same links and telecom networks by taking advantage of the transport and network multiplexing layer of these links. Advantageously, each input multimedia signal will be transported in the form of 1 or more paquétisées multimedia streams, the number of multimedia streams may be different for each signal.

Advantageously, each output multimedia signal is transported in the form of 1 or more paquétisées multimedia streams, the number of multimedia streams may be different for each output signal and may also be different from the number of multimedia streams of the corresponding source signal. In another embodiment, said method may duplicate a multitude of input streams having different formats or not.

Advantageously, each input source is represented by 1 or more streams and each destination to which it is duplicated is represented by 1 or more streams, this number being different for each source and each of the duplicated destinations. Advantageously, said duplicate flows can be sent to the recipients in different or different formats and different or different from the format of the source stream. Advantageously, each source and each destination implements mechanisms for improving the quality of service such as FEC mechanisms, jitter suppression, packet reordering, etc. Advantageously, each source and each destination implements transmission mechanisms of a stream through a multitude of different physical paths to have a transmission rate greater than the unit rate of a single line. Advantageously, each source and each destination implements mechanisms for transmitting a stream through a multitude of different physical paths in order to have redundancy mechanisms between these lines. Advantageously, the encapsulation formats, the quality of service enhancement devices and / or the transmission devices on multiple physical paths can be used or not, with different or different parameters, different between each of the 30 sources. and each of the destinations of this source. Advantageously, all the operations of duplication, format change and change of transport characteristics are performed by one of the equipment located in a site within the extended network or networks and that said changes are made in full on packetized streams. . In a particular embodiment, all the duplication operations are divided between several duplication and format change equipment, these units can be in several sites distributed in the network or networks Advantageously, the fact that the routing of multimedia streams to the duplication systems is ensured by the network addressing mechanisms of the transport networks used, thus making it possible to take advantage of the modularity by overcoming the IP infrastructures to propose a system of duplication and format change that can be modified in stages. The invention also comprises a device for transporting and real-time routing of multimedia signals from n sources to m destinations for the implementation of the method and comprising at least interfaces (2) to the communication stack (3). ) network and transport characterized in that each input multimedia signal will be transported via said communication stack in the form of 1 or more paquétisées multimedia streams, the number of multimedia streams can be different for each signal. In a particular embodiment, said equipment comprises at least equipment (118) reassembling fragments of streams carried on several lines to restore either an IP stream for example, possibly in a format different from the original format, or a multimedia signal, ASI, SDI or HD-SDI for example (115).

Advantageously, it comprises at least gateways (112) which ensure the format conversion of a packetized stream, for example under IP to a traditional multimedia signal (ASI, SDI or HD-SDI for example).

Finally, the invention denotes a system for transporting and real-time routing of multimedia signals from n sources to m destinations for the implementation of the method and comprising at least interfaces (2) to the communication stack (3). network and transport, at least equipment (118) ensuring the reassembly of the stream fragments conveyed over several lines to restore either an IP stream, possibly in a format different from the original format, or a multimedia signal, at least gateways (112) which provide the format conversion of a packetized stream, for example under IP to a traditional multimedia signal (ASI, SDI or HD-SDI for example) and at least the gateway functions (105) which make it possible to ensure the fragmentation of the stream over several physical lines to allow going beyond the bit rate of a single line and at least one of the equipment located in a site within the extended network or networks; All changes are made on paquétisées feeds to ensure all duplication operations, change of format and change of transport characteristics. Other features and advantages of the invention will appear more clearly on reading the following description of a preferred embodiment, given as a simple illustrative and nonlimiting example, accompanied by 7 figures illustrating the general principle of the invention. invention. The present invention provides a simpler architecture, less expensive and less complicated to maintain. The number of lines used is m + n at worst in this case and if some sources are on the same site, additional pooling is possible as shown in Figure 1. The advantages of the present invention are: significantly reduces the number of links, by pooling the uplinks and downlinks to multiplex, advantageously at the IP level, several signals on the same physical link and by duplicating and switching the multimedia streams on the central point to distribute them to different destinations. The present invention also makes it possible to have modifiable modularity by step (scalable in English) infinitely higher than that of traditional video switching solutions. These are made in the video field, with 1 BNC cable for each channel. Matrix sizes are very rarely greater than 512x512 and when they are of such size, they are of a large size and cabling aspect also becomes problematic. The solution we offer allows, on a Gigabit Ethernet port, to receive or transmit several tens of multimedia streams (typically between 50 and 200 depending on the rate), each unit can accommodate multiple Gigabit Ethernet interfaces. On the other hand, simply add units to the central site to increase capacity. Indeed, it is the IP addressing mechanisms of the input and output networks that make it possible to share the processing of flows between several units, thus making the extension of the capacity very simple. In a conventional solution, the doubling of the capacity results in complementary wiring whereas it is not. not the case in our solution. • To be able to propose an adaptation of the incident format and the delivery format according to the customer's needs, with: the possibility of having a different format for each customer. In a traditional solution, only two formats exist: SDI and ASI. In our solution we offer all encapsulation formats available in IP. • In addition to the previous point, and to improve the modularity by step, the problems of interfaces with the traditional world of video is deported to the edges of the network. Thus, only video streams over IP are transported and duplicated within the solution, including for the duplication and adaptation functions of the encapsulation format. The present invention is compatible with all multimedia formats over IP to allow interconnection with all equipment on the market. • Feeds from each source and each destination are treated independently in terms of transport creating an isolation between the two. This makes it possible to adapt the transport format to the characteristics of each customer and to the characteristics of his network. This makes it possible, for example, to apply error correction mechanisms for one target and not another, or to apply different parameters. • To accommodate a variety of network configurations, fragmentation and reassembly mechanisms are also available, with the ability to apply this mechanism differentially to sources and destinations. These mechanisms make it possible to fragment an original flow towards several physical lines, so as to have the sum of the flows of all the lines to convey this flow, as well as a fault tolerance by ensuring the redundancy of a line. on the other. • Compared with existing solutions in the network or audiovisual domains, the present invention makes it possible to serve IP video clients directly, including IPTV, and at the same time enables them to serve traditional video customers by offering them, at the same time end of links, the ability to deliver or obtain traditional video formats. On the other hand, the processing, formatting, duplication and routing are all done in the IP domain to offer unparalleled levels of performance in terms of integration and scalability. Let us now describe in detail the overall functioning of the system as represented in FIG. 3. In a first example, we seek to convey a number (Ni) of source multimedia signals (101, 102 and 103b) to a certain number (M1) of destination multimedia signals (114, 115 and 116).

These source signals may be of traditional video, SDI or ASI type (101), directly in the form of an IP multimedia stream (103b) or one of the two types (102). These source signals are adapted to the IP network by using on the source site, conversion equipment from the traditional format to IP (104) or a traditional format or IP to a more suitable IP format (105).

The gateway functions (104) can be rendered by various pieces of equipment on the market which will make it possible to transform a multimedia signal (101 or 102) into multimedia streams packaged in various formats (ProMPEG COP3 with or without FEC, TS / UDP for example). The functions of gateway (105) will also allow to ensure the fragmentation of the flow on several physical lines to allow to go beyond the flow of a single line. The gateway functions (105) will also make it possible, if the source signal (102) is already in the packetized multimedia flow format, to change, if necessary, the encapsulation of this stream and to provide, if necessary, said fragmentation function. . Some sites may have several sources that will first be transported on a local network (106), Ethernet type for example, before being transported over a fiber-type IP link (107) for example. Some sites will broadcast on such a link directly. Other sites will broadcast on an aggregate of IP lines (108), for example xDSL lines whose bit rates have been added by the fragmentation / reassembly technology. Finally, other sites will also use fibers (107) to route streams already packetized format such as TS / UDP or ProMpeg COP 3 for example. All of these streams, via the access links (107 and 108), will then transit via one or more extended networks (103) IP type or Ethernet for example. They will be conveyed to one or more sites or modules (109) called datacenters in which the format adaptation and duplication functions will be provided. The streams will be received on a local area network at the module (106) of said data center (109), said network allowing interconnection via local links (110a) to (110z) of Fast or GigabitEthernet type, for example, to RSW equipment (111). ) described later. Said local network is composed of routers, switches and firewall to ensure the switching, redundancy, access control and security of this network. This equipment will ensure, for each incoming stream, the duplication of this stream to the different destinations of this stream and the adaptation of its encapsulation format to the desired format in reception. For example, one could receive a stream in COP3 RTP format with FEC and have to transmit it to at least two targets, one in RTP COP3 without FEC for example and the other directly in the format IPTV (TS / UDP) for example. Said RSW adapter replicator modules (111) installed in said data centers (109) will allow these operations. The resulting flows will again pass through said local area network (106) and then be transmitted through wide area networks (113), which may or may not be the same as the IP or Ethernet destination networks (103) for example. , connected via fiber-type IP or Ethernet links (117) for example These streams will be transported to their destination via access links (117) or an application link aggregate (119). The access links, in particular the broadband access links, will be able to carry several streams and will therefore be shared. In this case, some sites will use a local network (116) and some not, to route the flows to gateway equipment (112). The gateways (112) will provide the format conversion of a packetized stream, for example under IP to a traditional multimedia signal (ASI, SDI or HD-SDI for example).

Advantageously, the equipment (118) ensures the reassembly of the stream fragments conveyed over several lines to restore either a stream in IP for example, possibly in a format different from the original format, or a multimedia signal, ASI, SDI or HD- SDI for example (115). Some other sites will directly receive the packetized stream (116), in their IP form for example, without transformation. The streams will be rendered for the other sites in video formats, for example ASI, SDI or HD-SDI (114). We will now describe the detail of the operation of the RSW adapter replicator module (111) as shown in FIG.

Paquetized multimedia streams (110a to 110z) are from or to the network. It can be multimedia streams in various encapsulation formats. They may or may not implement quality of service mechanisms and lost packet recovery (FEC for example). Each multimedia stream corresponds to a network stream (in the transport / socket sense). Each source, corresponding to each source signal, can be at the origin of one or more paquétisé multimedia streams. This is for example the case when using functions of fragmentation and reassembly on several physical lines. There may also be several multimedia streams per source, for example in the case of using the FEC COP 3. These streams are transported from their source on one or more IP networks and / or Ethernet.

There is a number (N2) of packaged multimedia streams (110a to 110z) at the input. Ni being the number of multimedia sources, we will have N2 z N1. Each stream (110i) is attached to only one source.

The network interfaces (2) represent physical interfaces with the IP network. Typically, it is Fast or Gigabit Ethernet interface with physical connections copper twisted pair or fiber optics for example. The system comprises from 1 to I physical interfaces. Each physical interface accepts 0, 1 or more packaged media streams. The network and transport protocol stacks (3) are network sense protocol (OSI model) management stacks of the term for processing the network layer and the transport layer. For example IP for the network layer and TCP or UDP for the transport layer. The Format Streaming and Format Adaptation Module (4) makes it possible to accept various input IP multimedia flow formats (without being limited to TS / UDP format, TS / RTP / UDP compliant with the ProMPEG COP3 or no, additional FEC flow conforming to ProMPEG COP3 or not, etc.). It ensures the standardization of the input packets to have, whatever the type of input, only RTP packets containing media streams in TS format. There are as many of these modules as multimedia sources to process (Ni sources therefore N1 modules). The source media streams are received from the network and transport protocol stack (3) via an internal communication bus (10) to the module (4). In the case where several streams are destined for said module (4), these flows coming from the same source, the protocol stack addresses these multiple flows to said module (4). This will be the case for example in the presence of fragmentation / reassembly, FEC COP 3, etc. In any case, said module (4) performs one or more of the following functions, adapted to the input format: • Timestamp: puts a date and time on the packet at the reception, if the packet is not already timestamped • If necessary, reassemble stream fragments if the fragmentation and reassembly function is implemented for transmission over several physical links. • Reorder the packets in case these packets have a number sequence (such as RTP). where they arrive disordered • Detect lost packets (if allowed by the starting protocol, RTP for example) • Recover lost packets by an FEC mechanism when possible (COP3 compliant stream for example) • Proceed to the dice -encapsulation to extract the useful multimedia content (extraction of the TS packets for example) • Re-encapsu.l the useful multimedia content in a RTP packet having 20 o The original timestamp if it exists, the timestamp generated in default o The original serial number if it exists, the serial number generated in default. In summary, said stream receiving and format adaptation module (4) receives a plurality of input multimedia streams corresponding to a single content, from a single source, with variable formats, extracts from these streams the content Multimedia transported 30 - and generates a new multimedia stream, in a common format holes modules (4) (TS / RTP compliant COP3 for example). The queue (5) contains the packets, for example RTP, at the output of the reception and adaptation module (4) and is at the input of the duplication matrix (6). This queue is of the FIFO type and it also allows the insertion of a packet at a given location of the queue by the stream receiving and matching module (4) to allow the packets to be reordered. Said duplication matrix (6) ensures the duplication of the packets on N1 inputs to M1 outputs. This element is detailed below. It receives packets of said RTP Queue (5) via an internal bus (10). It only processes RTP packets. The queue (7) of output RTP packets is the dual element of (5) with the same properties. It receives the packets, in RTP format for example, from said duplication matrix (6) and delivers the packets to the adaptation and transmission module of the streams (8). From the packets received from said queue (7), in an RTP format for example, said module (8) for format adaptation and stream transmission ensures the transmission of packets to the network destination. All packets received by said module (8) are in the same format, for example TS / RTP conforming to COP3. Said module (8) ensures on the one hand the adaptation to the output format by re-encapsulating the multimedia content contained in the incident packet to the format of the output packet. On the other hand, it provides, depending on the output format, the addition of FEC redundancy information, as well as the addition of sequence information (number sequence) and timestamp. The sequence number number and timestamp information are directly derived (copying being the most efficient form) of the incident packet, for example in the RTP format. This information is added if the output format requires it and takes advantage of it. The incident stream can on the other hand be fragmented into 2 or more stream fragments if the network to which the target is connected requires it. Finally, the timestamp information of the incident stream is used to suppress transmission jitter by conventional jitter suppression methods. The packets are delivered to the communication stack (3), for example in UDP sockets, to be sent over the network. For each destination, we will have 1 or more flows depending on whether the QoS enhancement functionalities (FEC, etc.) are implemented or not, as well as the fragmentation functions of the stream into several stream fragments, for example . The communication stack (3) then delivers the packets over the network through the appropriate physical interface (2). The communication stack provides, according to IP, the routing to the correct physical interface (2).

Advantageously, the physical interfaces (2) manage packets in VLAN (IEEE 802.1Q) format. The different modules are interconnected by internal communication buses (10), for example PCI, PCI Express, Compact PCI ... A generic processor (RISC architecture, x86 for example), ensures the data transfer of one module to another. The different processing modules are implemented by specific processors or are implemented as a task of the previous processor. The buffers are implemented in a fast access memory (for example RAM) accessible by said communication bus (10) and the processors. Said adaptation module (4) makes it possible to use the best format and parameters for the network connecting the source. The module (8) makes it possible to use the best format and the best parameters for the network connecting the target. These modules apply Quality of Services strategies, flow protection, dejittering, etc., differentiated according to the source and each destination. Given that even for the same source there are separate transmit branches, a differentiated strategy for each of the targets of the same source is applied.

In summary, the RSW adapter replicator system processes N2 (0, 1 or more) multimedia streams (110i) as input from the I interfaces (2) (1 to several interfaces), transported over a local or extended network from their source to the next. to the RSW adapter duplicator module via an IP or Ethernet network. These N2 multimedia streams (1) represent N1 sources, each source can use 1 or more multimedia streams (1) to carry each single multimedia content (N2 z N1). All the multimedia streams are sent by the interfaces (2) to the network and transport communication stack (3) (in the sense of the OSI network model) via an internal communication bus (10). According to their network and transport addressing (for example IP address and UDP or TCP port), the streams are addressed by the communication stack (3), via an internal bus (10), to flow reception modules and format adaptation (4), which provides traditional reception processing such as packet reordering, and more specific processing such as lost packet recovery, fragmented stream reassembly for example. Said modules (4) finally re-encapsulate previously processed multimedia streams as indicated previously in RTP packets for example, the essential being that they are in a format common to all sources and that this format can support the transport timestamp information and sequence number. These packets are transmitted to said input queue (5) of said duplication matrix (6) via said internal bus (10). There is N1 (N1 being the number of sources) reception lines composed of the succession of elements (10), (4), (10), (5), (10): a line for each source. The flows coming from these lines are transmitted to the duplication matrix which will, for the Ni sources and thus reception lines, duplicate the RTP packets to Ml destinations.

Duplicate packets are sent to transmission lines composed of said elements (10), (7), (10), (8). The elements are: transmitted by said duplication matrix (6) via the internal bus (10) to a queue (7). This queue delivers the packets to the format and transmission adaptation module (8) of the stream. This module will adapt the stream to the common format (the standardized stream, in RTP format for example), to all the destinations, possibly using different formats and adding, if necessary, redundancy, timestamp and network number information. sequence for example. Said module (8) also ensures the release of the packets according to the RTP timestamp so as to eliminate the IP jitter induced during the incident transmission and this by using conventional dejittering methods.

Advantageously, said module (8) implements flow fragmentation mechanisms. The flows resulting from the passage in said module (8) are sent to said communication stack (3) which will route and forward the packets to the corresponding physical interfaces (1). Each transmission line will use 1 to several multimedia streams for each of the destinations to be reached. The set of Ml lines of previous emissions (consisting of the succession of elements (10), (7), (10), (8)) will emit M2 multimedia streams to M1 different destinations. These M2 multimedia streams will be sent through the I network interfaces, each interface managing 0, 1 or more multimedia streams.

Now let us describe said duplication matrix noted (6) in FIG. 4 and detailed in FIG. 5. Said duplication matrix (6) is delimited by the frame containing the duplication table (11) and the duplication processors (9). The duplication processors are typically implemented on generic processors RISC type or x86 for example. One or more replication processors (9) can be implemented on the same physical processor. The duplication table (11) is typically implemented as a shared memory between the processors. This duplication matrix manages packets, for example RTP packets. The packets are received from a queue (5) via an internal bus (10) on a duplication processor. Each reception line, in the sense indicated above, has a duplication processor. The duplication matrix will thus have N1 duplication processor. For each packet entering a duplication processor (9), a copy of this packet is sent via said internal bus (10) to 0, 1 or more output queues (7). Each output queue (7) is associated with a transmission line and there are M1 transmission lines. The set of duplication processors has a duplication table that has N1 rows (inputs) and M1 columns (outputs). Booleans are set to true in this table to indicate the output lines to which each duplication processor must copy these packets.

For example, if the processor n must send these packets to the transmission lines a, b and c, the values at the intersection of the line n and the columns a, b, and c are set to true. In this line, the other values are wrong.

This table must respect the constraint that each column must not have more than one value set to true. For each new packet in said input queue (5), the duplication processor traverses the row of the corresponding matrix and duplicates this packet in said output queue (7) attached to the transmission lines for which the value of the matrix is set to true. Advantageously, other mechanisms, based on linked lists for example, are used to implement this function. However, its principle remains the same and the constraints remain the same to ensure proper operation of the system. Let's now describe the process of fragmentation and reassembly of flows. The fragmentation method is implemented in the gateways (105) or RSW adapter replicator modules (111), and the reassembly method is implemented in the RSW adapter replicator module (111) or in the gateways (118). This process covers different functions that can be implemented separately from each other. Said aforementioned elements implement said mechanisms and implement them only when the configuration requires it. The fragmentation and reassembly method makes it possible to transmit, between the aforementioned modules, a multimedia stream using one or more physical paths. Each path consists of at least one access link and at least one wide area network. The access links will be mostly different for different paths. The incident multimedia stream is fragmented at the source to these different paths and these different fragments are reassembled at their destination to render a single stream.

Since the gateways and modules (105), (111) and (118) can generate or receive several streams for each source and destination, the fragmentation and reassembly method is applied or not to each multimedia stream handled by these entities.

The method thus makes it possible to exceed the flow limits imposed by a single connection line. The user can therefore achieve the desired rate by multiplying the number of physical path between the transmitter and the network or between the network and the receiver.

In addition, the fact that the connection between the receiver and the network or between the transmitter and the network consists of a plurality of physical paths makes it possible, in particular, to use at least a portion of the available bit rate of the lines as a backup from one another, which increases the guarantee of routing and the continuity of operation of the overall system. Advantageously, the method comprises a counter-reaction, from the receiver module to the transmitter module, making it possible to regulate the transmission of the packets of the multimedia stream, said counter-reaction being carried out by sending from the receiver module to the transmitter module of at least one indicator relating to the transmission of packets on at least one physical path. These indicators can be transmitted using for example TCP / IP. These indicators can for example include the following data, characteristics of the quality of the transmission: the number of packets received and the bit rate, the number of packets lost, the number of packets received disordered, the jitter, the latency of the line, and - the number of desynchronization of the flow, These indicators are calculated by conventional methods. Advantageously, the counter-reaction can be performed on at least one physical path. Advantageously, the feedback can cause an adjustment of a data transmission rate on at least one path according to at least one indicator relating to the transmission. It may also, for example, cause a variation of a global encoding bit rate on at least one internal or external equipment provider of the multimedia source if this is possible, for example an MPEG-2 or MPEG-4 encoder. FIG. 6 represents a general scheme of use of the splitting and reassembly function, implemented in a transmission equipment (105) typically to the RSW (111) or RSW (111) to the receiving equipment ( 118). The incoming multimedia stream (F) is fragmented by the fragmentation module (211) into several stream fragments. This stream will be re-encapsulated if it does not contain timestamp and sequence number information. For example, said stream is re-encapsulated in an RTP format. The resulting stream will be referred to as the original reencapsulated stream. We will use the same name if no re-encapsulation was necessary (case of a flow (F) in RTP format for example). Each stream fragment can then be conveyed over a separate physical link (213) composed of a transmission line and a modem. The splitting of the multimedia data stream into a plurality of stream fragments and / or the distribution of data packets over a plurality of paths are for example performed by a scheduler using a weighted Round Robin scheduling. This scheduling is well known to those skilled in the art. For more information see sections 2.3.2 and 2.3.2.1 of Prioritization Algorithm Study in a Service Differentiation Internet, Octavio Napoleon MEDINA CARJAVAL, March 2001, or Advantageously, splitting packets of data is dynamically adjusted based essentially on at least one fixed or variable effective transmission rate on at least one link. Advantageously, the splitting of data packets is dynamically adjusted as a function also of the quality indicator of the transmission over a given path, for example a value of transmission jitter, latency or packet loss rate on at least one path .

The packets are re-encapsulated, for example using RTP or a specific format, to at least add sequence number information specific to each path. These streams will be named re-encapsulated streams by path. The original sequence number and timestamp information is preserved during this re-encapsulation. The packets are then conveyed through one or more local or wide area networks (214) and are received on one or more links (215) at the destination site. The packets are routed through these links (215) to the stream fragment reassembly module (212). The reassembly module (212) delivers a reassembled flow (F) identical to the original flow. The stream is reassembled and reordered thanks to the sequence number present in the packets conveyed (in the original re-encapsulated stream in case of re-encapsulation or in the original stream in the opposite case), for example RTP packets. If the stream has been re-encapsulated by the fragmentation module (211), it will be extracted to be restored to the original format. Figure 7 is a block diagram of splitting and reassembly of the multimedia data stream taking into account the transmission quality in accordance with the method according to the invention, the latter taken into account being an advantageous version of the invention. The input packetized media stream (222) is distributed according to a weighted Round Robin algorithm (220) between multiple paths (221). The distribution is based on weights that are transmitted by the fragmentation control module (233) via a local communication (235), for example an inter-process communication or a local TCP-IP connection.

The packets sent on the different paths (221) are advantageously re-encapsulated to have sequence numbers specific to each path, which makes it possible to very effectively determine the lost packet rate per path. These are the re-encapsulated flows by path.

Upon reception at the end of the paths (221), a quality estimator (232) measures the transmission quality on each of the paths. These measurements can include, among other things, jitter, lost packet rate, latency, and so on. Stream fragments from each path in a re-encapsulated path format are de-encapsulated and reassembled using the original sequence number (that of the original re-encapsulated flow) written in the packets. If the original stream has been re-encapsulated (original reencapsulated stream), it is unencapsulated and restored to the original format. The resulting stream (223) is identical to the original stream (222). The data of the estimator (232) is periodically transmitted to the transmitter via a communication link (231) using for example a protocol over TCP / IP. The quality and quality data of each path are received by the fragmentation control module (233). This module, based on the quality information received from the estimation module (232) and on the basis of the constraints such as the maximum bit rate of the lines or the ideal target distribution indicated by the user during the initial parameterization, dynamically determines: the weights of the round Robin weighted scheduler (220), the maximum possible throughput (sums of the maximum flow rates of the flux fragments). The first information is transmitted by a local call (235) to the scheduler (220) which immediately applies this new rate distribution between the different links (221).

The second information is transmitted to an encoding module (236) which may be internal or external to the entities concerned, namely the end gateways (105 or 118) or external to them but interconnected therewith. It enables the coding module to change its encoding rate according to the one it receives via the link (237). The links (237) and (235) can be implemented in the form of TCP-IP communications or inter-process communications for example. The fragmentation control module (233) determines the assignment strategy based on pre-determined logic or fuzzy logic, for example. This logic 10 will involve the different quality indicators with adjustable weights. This logic will also make it possible to manage cases of redundancy francs between the lines with strategies of replacement of a line by a free line in case of interruption of the service on the first. 15

Claims (24)

REVENDICATIONS1. A method for real-time transport and routing of packetized multimedia streams from n sources to m destinations, characterized in that it can receive one or more input streams from one or more sources and duplicate any stream any format and send it simultaneously to one or more destinations through packet transmission networks. 2. Transport and routing method according to claim 1, characterized in that said packetized multimedia streams are conveyed through telecom access links packet transport type and through one or more extended telecom networks of the same type. 3. A method of transport and routing according to any one of claims 1 and 2, characterized in that the adaptation between the video signal in its electrical form and the fluxed or fluxes representing it is carried out at the end of the Source side connection of specialized gateways. 4. A method of transport and routing according to any one of claims 1 and 2, characterized in that the adaptation between the packetized or fluxes and the video signal in its electrical form that they represent is carried out at the end. from the destination side link through dedicated gateways. 5. A method of transport and routing according to claim 3, characterized in that said adaptation gives rise to 1 to several paquétisées multimedia streams. 6. A method of transport and routing according to claim 4, characterized in that said adaptation is made from 1 to several multimedia streams paquétisés. 7. Transport and routing method according to any one of the preceding claims, characterized in that said packetized streams remain in the packetized form between the source and the destination, without ever returning in their form of multimedia signal between these two ends. 8. transport and routing method according to any one of the preceding claims, characterized in that said multimedia streams from different sources or with different destinations may or may not use the same links and telecom networks taking advantage of the transport and network multiplexing layer of these links. 9. A method of transport and routing according to any one of the preceding claims, characterized in that each input multimedia signal will be transported in the form of 1 or more multimedia streams packetized, the number of multimedia streams may be different for each signal. 10. transport and routing method according to any one of the preceding claims, characterized in that each output multimedia signal is transported in the form of 1 or more packaged multimedia streams, the number of multimedia streams may be different for each output signal and may also be different from the number of multimedia streams of the corresponding source signal. 11. Transport and routing method according to any one of the preceding claims, characterized in that it can duplicate a multitude of input streams having different formats or not. 12. Transport and routing method according to any one of the preceding claims, characterized in that each input source is represented by 1 or more streams and each destination to which it is duplicated is represented by 1 or more streams, this number may be different for each source and each of the duplicate destinations. 13. Transport and routing method according to any one of the preceding claims, characterized in that said duplicate flows can be sent to the recipients in different or different formats and different or different format of the source stream. 14. Transport and routing method according to any one of the preceding claims, characterized in that each source and each destination implements mechanisms for improving the quality of service such as FEC mechanisms, removal of jitter , packet reordering, etc. 15. transport and routing method according to any one of the preceding claims, characterized in that each source and each destination implements mechanisms for transmitting a stream through a multitude of different physical paths to dispose of a transmission rate greater than the unit rate of a single line. 16. A method of transport and routing according to any one of the preceding claims, characterized in that each source and each destination implements mechanisms for transmitting a stream through a multitude of different physical paths to have mechanisms redundancies between these lines. Transport and routing method according to one of the preceding claims, characterized in that the encapsulation formats, the devices for improving the quality of service and / or the transmission devices on multiple physical paths. can be used or not, with different or different parameters, different between each of the sources and each destination of this source. 18. A method of transport and routing according to any one of the preceding claims, characterized in that all operations of duplication, format change and change of transport characteristics are provided by one of the equipment located in a site to within the extended network or networks and that said changes are made entirely on packetized streams. 19. A method of transport and routing according to any one of the preceding claims, characterized in that all the duplication operations are distributed between several duplication and format change equipment, these units can be in several sites distributed in the network (s) 20. Transport and routing method according to any one of the preceding claims, characterized in that the routing of multimedia streams to the duplication systems is provided by the network addressing mechanisms of the transport networks used, allowing to take advantage of the modularity by mitigating the IP infrastructures to propose a system of duplication and format change that can be modified in stages. 21. Equipment for transporting and real-time routing of multimedia signals from n sources to m destinations for the implementation of the method according to claim 1, comprising at least interfaces (2) to the communication stack (3). ) network and transport characterized in that each input multimedia signal will be transported via said communication stack in the form of 1 or more paquétisées multimedia streams, the number of multimedia streams can be different for each signal. 22. Equipment for transporting and real-time routing of multimedia signals from n sources to m destinations for the implementation of the method according to claim 21, characterized in that it comprises at least equipment (118) ensuring the reassembly of the stream fragments conveyed over several lines to restore either an IP stream for example, possibly in a format different from the original format, or a multimedia signal, ASI, SDI or HD-SDI for example (115). 23. Equipment for transporting and real-time routing of multimedia signals from n sources to m destinations for the implementation of the method according to any one of claims 21 to 22, characterized in that it comprises at least gateways (112) which provide the format conversion of a packetized stream, for example under IP to a traditional multimedia signal (ASI, SDI or HD-SDI for example). 24. System for transporting and real-time routing of multimedia signals from n sources to m destinations according to claims 101 to 24, characterized in that it comprises at least interfaces (2) to the communication stack (3). network and transport, at least equipment (118) ensuring the reassembly of the stream fragments conveyed over several lines to restore either an IP stream, possibly in a format different from the original format, or a multimedia signal, at least gateways (112) which provide the format conversion of a packetized stream, for example under IP to a traditional multimedia signal (ASI, SDI or HDSDI for example) and at least the gateway functions (105) which ensure fragmentation flow over a plurality of physical lines to allow going beyond the rate of a single line and at least one of the equipment located in a site within the extended network (s) and that said The operations are performed entirely on packetized streams to ensure all operations of duplication, format change and change of transport characteristics.