MXPA05013371A - Specific stream redirection of a multimedia telecommunication - Google Patents

Abstract

It is proposed a method for establishing a IP-telecommunications between two participants with a specific control of data stream from that multimedia IP-telecommunications while maintaining the IP-telecommunications between the first and the second participant terminal originally targeted. This is achieved by setting up a connection for the transmission of packets comprising multimedia data from the IP-telecommunications between a terminal from the first participant and a terminal from the second participant both connected to the IP-network while using a sniffer to analyze the header of the data packets from that IP-telecommunications received via the IP-network by the first or second participant. It is provided on that first or second participant terminal of a possibility to initiate a redirection of the analyzed packets corresponding to a specific data stream defined in the header of the respective packets towards a further terminal interconnected with that first or second participant terminal.

Description

REDIRECTION OF SPECIFIC CURRENT OF A MULTIMEDIA TELECOMMUNICATION Field of the invention The present invention relates to a method for establishing IP telecommunications between two participants. In addition, the present invention also relates to a software code executable by computer for the control of data ets received by a participating terminal from an IP telecommunication between two participants as well as to a client computer that will be connected to the IP network and which comprises a communication unit for a participant to carry out IP telecommunications with a second participant. The invention is based on a priority application EP 05 290 779.7 which is incorporated herein by reference.

BACKGROUND OF THE INVENTION IP telephony (Internet Protocol) also known as Voice over Internet Protocol (VoIP) is becoming increasingly popular. This evolution supports the basis of multimedia telecommunications such as IP-based videoconferencing. IP telecommunications are based on the use of Internet Protocol to transmit, for example, voice ets over REF: 168301 an IP network. Normally, the connection of a call is made by two endpoints that open communication sessions with each other. In the Public Switched Telephone Network (PSTN), "the base network for telecommunications oriented by connections, the public (or private) switch connects logical channels through the network to complete calls. VoIP, this connection is a stream of multimedia (audio, video or both) carried in real time.This connection is the carrier channel and represents the voice and / or video content that is being supplied.There are two competing standardized protocols for VoIP operations , ITU-TH.323 and the Session Initiation Protocol (SIP) of the IETF These two protocols describe the signaling and control of multimedia conferencing over et-based networks in many different ways. The ITU is a et-based multimedia communications system i, which is a set of specifications.These specifications define various signaling functions, as well as as media formats related to aged audio and video services.

The H.323 standards were generally the first to classify and resolve aspects of multimedia provision over LAN technologies. H.323 networks consist of access centers (media) and entrances of access centers. The access centers serve as both an H.323 termination endpoint and an interface with non-H.323 networks, such as the PSTN. The access center entries function as a central unit for call admission control, bandwidth management and call signaling. In comparison to that, the Session Initiation Protocol (SIP; RFC 3261) is part of the IETF's multimedia data and control protocol framework. SIP is a powerful client-server signaling protocol used in VoIP networks. SIP handles the establishment and conclusion of multimedia sessions between speakers; These sessions can include multimedia conferences, telephone calls and distribution of various media. It is based on the use of invitations to create Session Description Protocol (SDP) messages to carry out the exchange of capabilities and to establish the use of call control channels. These invitations allow participants to agree on a set of compatible media types. SIP supports user mobility by commanding and redirecting requests to the user's current location. Users can inform the server of their current location (IP address or URL) by sending a registration message to a record. The SIP client-server application has two modes of operation: SIP clients can send a signal through a proxy server or redirect. The main components of a VoIP network are very similar in functionality to those of a circuit switched network and are based on three main pieces, call the media access centers, the media access controllers / signage (access center entry) and the IP network itself. The media access centers are responsible for the origin of the calls, the detection of calls, the analogue digital conversion of voice and the creation of voice packets (CODEC functions). In addition, media access centers have optional features, such as voice compression (analog and / or digital), echo cancellation, mute suppression and statistics accumulation.The media access center forms the interface that the content It is used in such a way that it can be transported over the IP network.The media access centers are the sources of carrier traffic.Typically, each conversation (call) is a single IP session transported by a Real-Time Transport Protocol ( RTP, which runs on a User Diagram Protocol (UDP / IP) or on a Transmission Control Protocol (TCP / IP) .The controllers of media access centers (similar to the entries of H.323 access centers) host the signaling and control services that coordinate the functions of the media access center.The controller of the media access center has responsibility for some or all of the coordination of signaling of calls, translations of telephone numbers, consultation of guests, management of resources and services of signaling access centers to the PSTN (access center SS7). The Real-Time Transport Protocol (RTP) provides end-to-end provisioning services for data with real-time features, such as interactive audio and video. Services include identification of payload type, sequence numbering, time stamping and supply monitoring. The RTP protocol provides features for real-time applications with the ability to reconstruct synchronization, loss detection, security, content supply and identification of coding schemes. Media access centers that digitize voice use the RTP protocol to provide voice (bearer) traffic. For each participant, a particular pair of destination IP addresses defines the session between the two endpoints, which translates into a single RTP session for each telephone call in progress. RTP is an application service integrated in UDP, so it has no connections with a better effort supply. As part of its specifications, the RTP Useful Load Type field includes the coding scheme that the media access center uses to digitize the voice content. This field identifies the RTP payload format and determines its interpretation by the CODEC in the media access center. A profile specifies a static mapping by default of payload type codes to payload formats. These mappings represent the ITU-G series of coding schemes as well as the corresponding ones for video. In US 2002/0194606 a system and method for communication between videoconferencing systems and computer systems is described. That system includes a videoconferencing unit and a processor. The video conferencing unit is a system that captures audio and video information, and creates data in a format suitable for the RTP protocol. The processor receives the data and reassembles it in a format suitable for standard media in computer systems. More specifically, the stage of reassembling the data in a format suitable for standard media in computer systems can be achieved by first determining whether a data box contains audio or video data, then temporarily storing the audio data or video data. as appropriate. The data is then created in a format suitable for standard media in computer systems. Once the data is properly formatted and reassembled, they can be sent as an email attachment or stored on a server. This system and method are not suitable for a specific control of data stream from an IP telecommunication.

BRIEF DESCRIPTION OF THE INVENTION In view of the foregoing, an objective of the present invention is to provide a method for establishing an IP telecommunication between two participants with a control of data streams specific to IP multimedia telecommunications while maintaining IP telecommunications. between the first and second participating terminals originally selected. It is also an object of the present invention to provide a software code executable by computer for the control of data packets received by a participating terminal from IP telecommunications between two participants. Moreover, an object of the present invention is to provide a client computer that will be connected through the IP network and which comprises a communication unit for a participant to carry out IP telecommunications with a second participant, the client computer comprises a computer-readable medium having a computer program recorded therein, the computer program comprises codes that provide specific control of data streams to that of IP telecommunications by multimedia. This objective is achieved according to the invention by applying the steps of: • establishing a connection for the transmission of packets comprising multimedia data from the IP telecommunications between a terminal of the first participant and a terminal of the second participant both connected to the IP network; • use a sniffer to analyze the header of the data packets to the IP telecommunications received by the IP network by the first or second participant, the sniffer possibly but not exclusively is a unit implemented in that first or second participating terminal; • provide in the first or second participant terminal a possibility to initiate a redirection of the analyzed packets corresponding to specific data streams defined in the header of the respective packets towards an additional terminal interconnected with the first or second participating terminal while maintaining the Multimedia IP telecommunications between the first and second participating terminals usually being the first one selected. In an alternative of the modality according to the invention, the specific data that will be redirected correspond to the stream of videos that come from those multimedia IP telecommunications. This can be applied appropriately for IP multimedia telecommunications that are part of a teleconference providing possibilities of sharing documents among the participants. In another alternative according to the invention, the specific data stream can be redirected to a dedicated port of the first or second participating terminal. In this way, the specific data stream redirected to that dedicated port does not necessarily have to be preceded by some signaling data since the additional terminal that will be connected to that dedicated port is already adapted to process that specific data stream. For example, if the specific data stream is part of a video stream coming from multimedia IP telecommunications, that additional terminal connected to the dedicated port would suitably be a terminal with a visual presenter on which the corresponding images would be visually presented. to the video stream. According to another aspect of the invention, its objective is achieved by a software code executable by computer for the control of data packets received by a participating terminal from an IP telecommunication between two participants. That code comprises a code that provides a possibility for that participating terminal to initiate a redirection of the data packets corresponding to a specific data stream. This is achieved after recognizing the characterization of that specific data stream normally defined in the header of the respective packets. This can be done by a certain sniffer that is possible but not necessarily part of the software code executable by computer. The use of this computer executable software code allows to adequately control in a separate way the different specific IP telecommunications currents. This can be particularly advantageous when IP telecommunications are part of a teleconference to thereby release, for example, the visual presentation of the initially selected terminal of the video portion of IP telecommunications when transferring that video to an additional terminal connected to that terminal. This software code executable by computer can be installed in the caller or in the call or even in both participating terminals used for IP telecommunications. Suitable developments of the invention are described in the dependent claims, in the following description and in the figures.

BRIEF DESCRIPTION OF THE FIGURES An exemplary embodiment of the invention will now be further explained with reference to the accompanying figures, in which: Figure 1 is a schematic view of an architecture with the different stages for the establishment of IP telecommunications as used for the present invention. Figure 2 is a schematic view of the same architecture as that of Figure 1 with a different sequence of steps for establishing IP telecommunications as used for the present invention. Figure 3 is a schematic view of an embodiment according to the present invention. Figure 4 is a flow diagram of an embodiment according to the present invention.

DETAILED DESCRIPTION OF THE INVENTION In Figure 1 a typical architecture is shown as used when implementing the present invention. The main characteristic for carrying out IP telecommunications between two participants, that is to say, a caller and a call, are the terminal of the caller and of the call respectively 1 and 2 both connected by means of their access center 11, 12 respectively to the network IP 3. Normally an additional constituent is an access center entry (H.323) or a call agent as a media access center (SIP) controller connected to the IP network 3 is involved when an IP telecommunication is established. There are several ways to establish these IP telecommunications. If H.323 is selected as reference, then the sequence of steps is as follows: at the start or entry terminals 1, 2 of the caller and called register respectively are registered with access center entry 13 through their service centers. respective access 11, 12. When the caller dials the destination telephone number of the call at his terminal 1, that request is sent to the access center entrance 13 through the access center 11. Optionally, the access center entrance 13 authorizes IP telecommunications to be completed 1. The access center entry 13 maintains a record of the bandwidth requirements for those telecommunications. Then, the caller sends a call-start message 15 to the called party followed by the exchange of capabilities 16 (CODEC parameters, media stream installation) with the call. The call is informed from its access center 12 that it has an incoming call by a start message such as a common telephone sound but also any other type of message possibly presented visually on the call terminal screen 2. Optionally a request for Resource reservation protocol is sent directly between both access centers 11, 12. And then both participants open an RTP 17 session between themselves. Figure 2 shows the same architecture as in Figure 1, with a different sequence of stages when establishing IP telecommunications that correspond to the use of the SIP alternative standardized protocol. The difference with Figure 1 appears in the sequence of steps when an IP telecommunication is initiated. When the caller dials a call destination telephone number at his terminal 1, the media access center 11 then notifies the call agent 13 that the call is entering. The call agent 13 consults the telephone number (or universal resource locator) and directs the access center 12 to which the call 2 terminal is connected to create an RTP connection (IP address and port number) between both access centers of media 11 and 12. The call agent informs the destination media access center of the incoming call in 12 for example with a sound or any other message. Finally, both media access centers 11 and 12 open an RTP 17 session between themselves when both participants (caller and called) open IP telecommunications. When using the standardized SIP protocol, the call agent or the media access center controller 13 can be replaced by a SIP proxy server. This server after consulting the phone number or URL will send the invitation to the called party normally by any form of an email address. As an alternative, the calling agent 13 can be replaced by a SIP redirect server. This server after consulting the telephone number or URL to register the called party will then send a destination address back to the caller in a manner similar to that shown in figure 1 with 14. In the latter case, the caller sends an invitation directly to the call using the email address of the called party. Subsequently, the SIP clients, that is, terminals 1, 2 of the respective caller and called, open an RTP session among themselves when the called user opens (lifts) the IP telecommunications. According to the present invention, when an IP telecommunication is established between the terminal 1 of the caller and the terminal 2 of the called one, a sniffer can be activated to analyze the header of the received packet by means of the RTP session by the terminal of the called one. to the caller. The sniffer is able to extract RTP session parameters such as RTP ports, caller / responder IP addresses and dynamic CODEC types from for example the SIP session that precedes the data flow in RTP. This sniffer can be implemented in the so-called terminal 2 although other implementations are conceivable. Together with the sniffer, a software code in the form of a software code executable by the computer is provided to the called one, which comprises codes that provide the possibility for the calling terminal to initiate a redirection of the data packets corresponding to a data stream. specific. The characterization of that specific data stream is obtained by the sniffer when analyzing the header of the respective packets. As shown in Figures 1 and 2, an additional terminal 4 is connected to the call terminal 2. This interconnection may preferably be by means of a dedicated port such as a Universal Serial Bus but other interfaces such as a Bluetooth or WLAN are also conceivable. This additional terminal 4 is preferably an IP-type terminal such as an IP telephone with a visual presenter. In this way, the call has the possibility to redirect any specific data stream corresponding to, for example, the video current coming from those multimedia IP telecommunications between the terminal of the caller 1 and the terminal of the call 2 to that additional terminal 4. This is particularly suitable for releasing the visual display of the called terminal 2 of that video stream that is being displayed visually on the screen of that additional terminal 4. The use of a dedicated port such as a USB port has the advantage that no specific signaling is required before the specific data stream when it is redirected to the additional terminal 4 to alert that additional terminal 4 of the current content of the current. redirected data. Figure 3 shows a modality according to the invention that preferably but not exclusively is adapted for IP telecommunications within teleconferences. This figure shows an H.323 call control signaling as well as the RTP flow (CODEC video H.261, H.263 and others) between the caller terminal and the caller. The use of an RTP sniffer (directred) allows to clearly identify the different packet transferred by means of RTP between the caller and the caller. In particular, it is then possible to identify the packets corresponding to the video current so as to be able to differentiate the different current (audio or video) according to its characterization by the sniffer. This differentiation can be used by an RTP switch that is possible but not exclusively implemented in the call terminal 2. Figure 3 also shows a computer executable software code normally used in a teleconference, for example, that allows to share documents such as doc files or presentations among the participants. This software code executable by computer in the form of a Software Development Team (SDK) here with the network session example has a direct access to the RTP switch. In the terminal of the call 2 is also implemented some code possibility that is part of that computer executable software code comprising a code that provides the possibility to initiate the redirection. This code can be in the form of a Java application, for example, with some icons that allow the call when it is started to redirect the data packets that correspond to a specific data stream defined in the header of the respective packets to the additional terminal 4 here interconnected via the USB port. In the example shown in figure 3 the specific data that will be redirected corresponds to the video stream that will be reproduced on the screen of the additional terminal which in this case is an IP terminal with a visual presenter. The call can then use the screen of its terminal 2 to visually present the document that will be shared without necessarily being limited by the video stream of those IP telecommunications. In Figure 4 a flow chart of an implementation according to the invention is shown. First, the specific development team starts to control the sharing of documents during a teleconference together with the Java application that allows to redirect specific data streams. IP telecommunications are established. When the RTP session is established the RTP flows through the RTP sniffer and the switch. The Java application in the mode according to Figure 4 in a similar way as in Figure 3 gives the call (or the caller or even both participants depending on the implementation) the option to redirect the video stream to its terminal that in that case it is the screen of the PC or the additional IP device. If the RTP sniffer finds the video stream then it is sent respectively to the SDK (here a network session) or to a dedicated USB port to be then presented visually. If the RTP sniffer is not able to find the video stream, for example, the RTP np session is fully established or a code not known to that sniffer is used, then the Java application gives the user the ability to initiate redirection in a later moment. It is clear from the above description that the solution proposed in accordance with the present invention can be implemented in a similar manner on the caller side. It is noted that in relation to this date, the best method known to the applicant to carry out the aforementioned invention, is that which is clear from the present description of the invention.

Claims (7)

1. Having described the invention as above, the content of the following claims is claimed as property: 1. A method for establishing IP telecommunications between two participants, characterized in that it comprises the steps of: • establishing a connection for the transmission of packets comprising data of multimedia from those IP telecommunications between a terminal of the first participant and a terminal of the second participant both connected to the IP network; • use a sniffer to analyze the header of the data packets of the IP telecommunications received by the IP network by the first or second participating terminal; • provide in that first or second participant terminal a possibility to initiate a redirection of the analyzed packets corresponding to specific data streams defined in the header of the respective packets towards an additional terminal interconnected with that first or second participating terminal while maintaining the IP multimedia telecommunications between the first and second participating terminals.
2. 2. The method according to claim 1, characterized in that it is adapted for specific data corresponding to the video stream coming from the IP telecommunications.
3. 3. The method of compliance with the claim 1, characterized in that IP multimedia telecommunications are part of a teleconference that provides possibilities of sharing documents among the participants.
4. 4. The method of compliance with the claim 1, characterized in that the specific data stream is redirected to a dedicated port of the first or second participating terminal.
5. 5. A software code executable by computer for the control of data packets received by a participating terminal from an IP telecommunication between two participants, characterized in that it comprises a code that provides a possibility for that participating terminal to initiate a redirection of the packets of data corresponding to a specific data stream defined in the header of the respective packets towards an additional terminal interconnected with that participating terminal while maintaining multimedia IP telecommunications between the two participating terminals.
6. 6. The computer executable software code according to claim 5, characterized in that the code is part of a software code executable by computer to carry out teleconferences through the IP network that provides possibilities for sharing documents among the participants .
7. 7. A client computer for being connected to the IP network and characterized in that it comprises a communication unit for a participant to carry out IP telecommunications with a second participant, the client computer comprises a computer readable medium having a program of computer recorded therein, the computer program comprises codes that provide a possibility to initiate a redirection of the packets coming from those IP telecommunications corresponding to a specific data stream defined in the header of the respective packets towards an additional terminal interconnected with that client computer while maintaining multimedia IP telecommunications between that communication unit and the second participating terminal.