EP1714473A1

EP1714473A1 - Multimedia packet transmission link setup using an interactive voice response system

Info

Publication number: EP1714473A1
Application number: EP05716643A
Authority: EP
Inventors: Klaus Hoffmann
Original assignee: Siemens AG; Nokia Siemens Networks GmbH and Co KG
Current assignee: HMD Global Oy
Priority date: 2004-02-11
Filing date: 2005-02-08
Publication date: 2006-10-25
Also published as: WO2005079049A1; DE102004006756A1; CN1918892A; US20070172051A1; DE102004006756B4

Abstract

The invention concerns a communication set up between subscribers (A, B) by means if an interactive voice response system (IVR). The generation of a recall of the subscriber (A) is controlled by signalling messages serving to warn the subscriber (B) of a call. A direct link TDM, RTP/RTCPA/B in the multimedia network (IN) is set up between the two subscribers (A, B) at latest when the communication is set up. Thus, it is possible to provide intelligent IVR network power characteristic of the PSTN network power in a multimedia packet transmission multimedia network (IN), such that a recall is properly displayed. The direct TDM, RTP/RTCPA/B link further enables the IVR system to be significantly free of load

Description

Establishment of a packet-oriented multimedia connection with the participation of an interactive voice response system

In the past, two main types of communication networks for transmitting information have emerged: packet-oriented (data) networks and line-oriented (voice) networks. In the course of the convergence of these two network types, convergent multimedia networks have emerged. Combining these different network types creates hybrid networks.

Line-oriented networks - also called voice networks, telephone networks or Public Switched Telephone Network (PSTN) - are designed for the transmission of continuously flowing (voice) information, referred to in the technical field as (voice) connection, conversation or call. The information is usually transmitted with high quality of service and security. For example, for speech, a minimal - e.g. <200 ms - Delay without fluctuations in the delay time (delay jitter) is important, since speech requires a continuous flow of information when played back in the receiving device. A loss of information cannot therefore be compensated for by retransmitting the non-transmitted information and usually leads to acoustically perceptible disturbances in the receiving device (e.g. crackling, distortion, echo, silence). In the professional world, the transmission of speech is also generally referred to as real-time (transmission) service or as real-time service.

Packet-oriented networks - also called data networks - are designed for the transmission of packet streams, which are also referred to in the technical field as data packet streams, session or flow. A high level of service usually does not have to be guaranteed here become. Without guaranteed quality of service, the transmission of the data packet streams takes place, for example, with delays that fluctuate over time, since the individual data packets of the data packet streams are usually transmitted in the order in which they are accessed by the network, ie the more packets to be transmitted from a data network, the greater the time delays. In the professional world, the transmission of data is therefore also referred to as a transmission service without real-time conditions or as a non-real-time service.

The packets usually differ depending on the type of packet-oriented network. For example, they can be configured as Internet, X.25 or frame relay packets, but also as ATM cells. They are sometimes referred to as news, especially then when a message is delivered in a packet.

The Internet is a well-known data network. Because of the Internet protocol IP used there, this is sometimes also called the IP network, although this term is to be understood broadly and includes all networks in which the IP protocol is used. The Internet is designed as an open (wide area) data network with open interfaces for connecting (mostly local and regional) data networks from different manufacturers. It provides a transport platform that is independent of the manufacturer.

Connections are communication relationships between at least two participants for the purpose of a - mostly mutual, ie bi-directional - information transfer. The participant initiating the connection is usually referred to as an 'A participant'. A subscriber connected through an A-subscriber is called a 'B-subscriber'. In a connectionless network, connections represent at least the unambiguously logical relationship between A and B subscribers, ie according to this view, for example, the connectionless flows in the Internet represent logically abstracted connections (eg A-subscriber = browser and B-subscriber = web server). In a connection-oriented network, connections also represent clear paths through the network on a physical level, along which the information is transmitted.

Signaling is used to coordinate network components with one another, but not for the "actual" transmission of information in the above sense. The information transmitted for signaling is usually referred to as signaling information, signaling data or simply signaling. The term is to be understood broadly. For example, also includes the messages for the control of registration, admission and status (RAS), the messages for the control of user channels of existing calls (e.g. in accordance with the H.245 standard) and all other similarly designed messages. The "actual information" is also called user information, payload, media information, media data or simply media to distinguish it from the signaling. Communication relationships leading to. Serving transmission of the signaling are also referred to below as signaling connections. The communication relationships used to transmit the user information are e.g. Speech connection, user channel connection or - simplified - user channel, bearer channel or simply called bearer.

In this context, out-of-band or outband means the transmission of information in a different way / medium than that provided in the communication network for the transmission of signaling and useful information. In particular, this includes a local configuration of devices on site, which e.g. is done with a local control facility. In contrast, at in-band

Information in the same way / medium, possibly logically separated from the signaling and user information considered tions, transmitted.

In the course of the convergence of voice and data networks, voice transmission services and increasingly also broadband services such as Transmission of moving picture information also implemented in packet-oriented networks, i.e. The transmission of the real-time services which have hitherto usually been line-oriented is carried out in a convergent network - also called voice-data network or multimedia network - in a packet-oriented manner, i.e. in packet streams. These will too

Realtime packet streams called. The transmission of voice information over a packet-oriented IP network is also identified with 'VoIP' (Voice over IP).

The international standardization committees IETF (Internet Engineering Task Force) and ITU (International Telecommunications Union) describe several distributed architectures for multimedia networks, which initially start from homogeneous multimedia networks.

At ITU, the basic standard H.323,., Defines the transport of voice, data and video streams over an IP network. Audio and video streams are transmitted in accordance with the RTP / RTCP protocol. The connection control is caused by the H.225 protocol, which enables the signaling, registration and synchronization of media streams. The H.323 architecture primarily provides the following types of functional units:

- terminal, e.g. a terminal in a Local Area Network (LAN), for bi-directional real-time communication with other end devices, gatekeeper for performing the connection control,

- Media Gateway (MG) at the interface to other networks for converting H.323 formats into the formats of these networks,

- Media gateway controller (MGC) for controlling media gateways, in particular their respective transmitted bindings, using the H.248 protocol and for converting between different signaling protocols.

At the IETF, session initiation protocol (SIP) standardizes telephony over the Internet, which enables interactive connections to be made available over the Internet. SIP supports the control of connections and the translation of SIP addresses into IP addresses. SIP is based on comparatively intelligent endpoints, many of which perform the signaling function themselves. If a connection is established using SIP, a description of the bearer is usually exchanged between the two sides of the connection. The Session Description Protocol (SDP) according to the RFC2327 standard is used for this. This use includes described in the standard RFC3264: "An Offer / Answer Model with the Session Description Protocol (SDP)". The following bearer data are particularly important:

IP address of the Bearer Connection RTP / UDP port of the Bearer Connection (depending on whether there is voice or data transmission) Codec (s) that can (can) be used for voice or data transmission

- Bearer Connection stream mode

A SIP proxy server can be used for a connection setup, for example if the connected endpoints do not know each other. It can also be designed to evaluate, change and / or forward a received request for a client (eg an IP telephone, a PC or a PDA). MG and MGC are also provided at the interface to other networks. The MGCP protocol (Media Gateway Control Protocol) is used to control the MG. bindings, using the H.248 protocol and for converting between different signaling protocols.

IP address of the Bearer Connection RTP / UDP port of the Bearer Connection (depending on whether there is a voice or data transmission) Codec (s) that can be used for voice or data transmission Stream mode of the Bearer Connection

A SIP proxy server can be used for a connection setup, for example if the connected endpoints do not know each other. It can also be designed to evaluate, change and / or forward a received request for a client (eg an IP telephone, a PC or a PDA). MG and MGC are also provided at the interface to other networks. The MGCP protocol (Media Gateway Control Protocol) is used to control the MG. It is common to both architectures that the connection control level and the resource control level are functionally clearly separated from each other and are usually even implemented on different hardware platforms.

The connection control level is used for the controlled activation, control and deactivation of network services. For this purpose, it can include dedicated connection controllers, to which the following functions can be assigned: - Address translation: conversion of E.164 telephone numbers and other alias addresses (e.g. computer names) to transport addresses (e.g. Internet addresses).

- Admission control: check whether and / or to what extent use of the communication network is permitted. - Alias Address Modification: Return of a modified alias address, which is used by endpoints e.g. be used to establish a connection.

- Bandwidth Control: Management of transmission capacities, for example by controlling the permissible number of devices that can use the communication network at the same time - _^ .

- Connection authorization: admissibility check for incoming and outgoing connection requests.

- Connection control signaling: switching and / or processing of signaling messages.

- Connection Management: management of existing connections.

- Dialed Digit Translation: Translation of the dialed digits into an E.164 telephone number or a number from a private numbering scheme.

- Zone Management: Registration of (e.g. VoIP-enabled) devices and provision of the above functions for all devices registered with the Connection Controller.

Examples of connection controllers are those from the ITU in the H.323 gatekeeper or the SIP proxy. The Resource Control level is used for the regulated implementation of activated services. To control network resources (e.g. transmission nodes), it can include resource controllers, to which the following functions can be assigned: - Capacity Control: Control of the volume of traffic fed to the communication network, for example by checking and possibly limiting the permissible transmission capacity of individual packet streams.

- Policy Activation: Reservation of (transmission) resources in the communication network.

- Priority management: Preferred transmission of priority traffic flows, e.g. with the help of priority indicators, which are provided in priority packets.

If a larger communication network is divided into several domains - also called 'zones' - a separate connection controller can be provided in each domain. A domain can also be operated without a connection controller. If several connection controllers are provided in a domain, only one of them should be activated. From a logical point of view, a connection controller should be seen separately from the facilities. Physically, however, it does not have to be implemented in a separate connection controller device, but can also be implemented in each end point of a connection (for example in the form of an H.323 or SIP terminal, media gateway, multipoint control unit) or in a primarily for program-controlled data processing Device (for example: computer, PC, server) can be provided. A physically distributed implementation is also possible.

An alternative example of a connection controller is a media gateway controller, to which the optional functions connection control signaling and connection management are usually assigned. Furthermore, the assignment of a signaling conversion function to implement different (signaling) protocols is conceivable, for example in the Boundary of two different networks, which are combined to form a hybrid network, may be required.

The resource controller is also known as a 'Policy Decision Point (PDP)'. It is implemented, for example, within so-called edge routers - also called edge devices, access nodes or, when assigned to an Internet service provider (ISP), also called provider edge routers (PER). These edge routers can also be designed as media gateways to other networks to which the multimedia networks are connected. These media gateways are then connected both to a multimedia network and to the other networks and are used internally for the implementation between the different (transmission) protocols of the different networks. The resource controller can also be designed only as a proxy and forward information relevant to the resource controller to a separate device on which the relevant information is processed in accordance with a function of the resource controller.

The exchange of signaling messages takes place in these networks either by means of a connection controller (Connection Controller Routed Signaling - CCRS) or directly between the end devices (Direct Endpoint Routed Signaling - DERS). Which connection is used can be individually determined for each terminal and for each transmission direction for each connection.

With CCRS, all signaling messages are transmitted by at least one call controller. All facilities send and receive signaling messages only via the call controller. A direct exchange of signaling messages between the facilities is prohibited.

With DERS, copies of selected signaling messages can be transmitted to the connection controller, so that a connection controller also has knowledge of the connections between the end devices in this variant can. However, these connections are not actively influenced or verified by himself.

In summary, the function split between the two levels can be described in such a way that the resource control level is assigned only the functions that are required for the transmission of useful information, while the connection control level includes the intelligence for controlling the resource control level. In other words, the facilities at the resource control level have as little network control intelligence as possible and can subsequently be implemented in an economically particularly advantageous manner on separate hardware platforms. This is a particularly nice advantage because of the higher number of installations in this level compared to the connection control level.

Combining different networks creates hybrid networks in which different protocols are used. In order for all devices in such a network to be able to communicate with one another without restrictions (eg IP-based telephones compatible with PSTN and vice versa), there is interworking between the respective protocols (eg SIP and H.323 in packet-oriented multimedia networks or ISUP and DSS1 in line-oriented PSTNs Networks) required. This interworking can be moved widely and includes, in addition to the pure interworking of the bearers, the interworking of features such as call hold, call waiting (call waiting), call redirect (call forwarding), 3PTY (three-party conference), CONF (conference without quantitative restrictions on conference participants) or IVR (Interactive Voice Response).

Interworking between two different protocols can be effected indirectly or directly. With indirect interworking, a further, third protocol is switched between the two protocols - for example the protocol BICC (Bearer Independent Call Control) according to the Standard Q.1902 or the SIP_T protocol (SIP for Telephones), which is described in the RFC3372 standard. In contrast, direct interworking takes place directly between the two different protocols, ie without the use of an intermediate protocol.

Both in convergent multimedia networks and in hybrid networks, which e.g. By combining a convergent multimedia network with a conventional line-oriented voice network, new technical problems arise due to the new or different technologies used in the respective network types when information is transmitted - especially in real-time packet streams.

It is an object of the invention to recognize at least one of these problems and to enrich the prior art by providing at least one solution.

The invention is based on the knowledge that during the evolution of hybrid networks which result from the interconnection of proven line-oriented networks with modern multimedia networks, many of the performance features which have long been established in line-oriented networks do not or at least not be fully supported.

One reason for this is seen in the large number of new interworking interfaces and protocols, of which the previous features are not yet or not fully supported.

Furthermore, the invention is based on the knowledge that the differentiated specifications for bearer handling in the different PSTN networks and multimedia networks do not match. For example, in PSTN networks and H.323 networks the partner is signaled that their own sending direction is blocked, while in SIP networks the partner is signaled that the partner is (from the point of view of the signaling party) remote

10 Transmission direction has to be interrupted, since in SIP networks only the own transmission direction, but not the own reception direction, is separated and thus each SIP subscriber suppresses his own transmission direction himself by deactivating his transmitter. On the other hand, some signal tones, such as the ringback, are generated at the B subscriber in PSTN networks and transmitted through the network to the A subscriber, while in multimedia networks the beep should preferably only be generated at the A subscriber.

According to the knowledge of the invention, these divergences mean that the intelligent network services of the PSTN network, such as the prepaid services - also called Interactive Voice Response IVR - have to be integrated into the complex structure of the growing networks when used in a packet-oriented multimedia network. According to the knowledge of the invention, it is no longer possible to transmit ringback tones across the packet-oriented multimedia networks as before. According to the knowledge of the invention, it is also undesirable to permanently connect the connections established with the aid of IVR systems via the IVR systems.

The IVR service is currently standardized at the IETF. In the previous draft standard draft-ietf.sipping-3pcc-03.txt, however, there is no reference to the findings of the invention. The problem of ringback tones is not discussed. As a result, this function is currently lost if the B subscribers are assigned to the multimedia network and therefore do not feed ringback tones into the payload stream.

In principle, it would be conceivable to create a ring back tone as a precautionary measure for the A subscriber. This solution is problematic, however, if the B subscriber cannot be reached at all or the connection to the B subscriber fails for other reasons, because this would simulate the A subscriber a state of the connection setup which does not correspond to the facts , From the operator's point of view, this is not the case

11 acceptable and cannot be offered commercially.

A solution to this problem situation on which the invention is based is specified in the patent claims.

There are a number of advantages associated with this solution:

By announcing the ringing to the B subscriber, a ringback can be displayed to the A subscriber.

Due to the notification with the aid of signaling messages, it is no longer necessary to send a ringback tone to the B subscriber.

- By coupling the ringback with the signaling messages, the A-subscriber is only shown the ringback if there is actually ringing at the B-subscriber. The simulation solution is no longer prone to errors. The solution is therefore acceptable from the operator's point of view and can also be offered commercially.

- The implementation of an IVR system in a multimedia network increases the acceptance of these modern networks.

Further advantageous refinements of the invention result from the subordinate or subordinate claims.

By switching the two connections of the IVR system to a direct connection between the two participants, the previous "looping through" of the bearer by the IVR system is no longer necessary, which considerably relieves the load on the IVR system.

By forming the first signaling message as a SIP re-INVITE message, the recommendation of the IETF standard RFC3311, chap. 5.1 fulfilled, according to which an existing connection (here the first connection, in the standard

12 dard "confirmed dialogue") an UPDATE message could also be sent, the retransmission of an INVITE, which in this case is also referred to as "re-INVITE", but is recommended. The specification of detailed SIP messages also has the nice advantage that the further development of the draft standard draft-ietf-sipping-3pcc-03.txt is made considerably easier.

By transmitting a ringback tone to an A subscriber who is assigned to a line-oriented network, the line-oriented terminals can continue to be used unchanged. This is the nice advantage of a seamless connection of the two networks to form a hybrid overall network. By displaying an information window, the method can be optimally used on devices with a display, e.g. Computers, mobile phones, etc. can be customized.

The invention is explained below on the basis of further exemplary embodiments, which are also shown in the figure. It shows:

1 shows an exemplary arrangement for carrying out the method according to the invention with a hybrid communication network, consisting of two packet-oriented multimedia networks and a line-oriented voice network, which are connected by intermediary media gateways, media gateway controllers and SIP proxies, and one end point each common features in each of the three networks

1 shows an exemplary arrangement for carrying out the method according to the invention. It comprises a line-oriented network PSTN _A and two multimedia networks IN _B and IN _I VR, which are preferably designed as integrated voice-data networks SDN. The PSTN _A , IN _B and IN _{1VR networks} are combined to form a hybrid network. The networks are IN

13 preferably designed as IP networks and each include a SIP proxy SP _B or SPIVR as a call controller. It is obvious to the person skilled in the art that the invention can of course be used in any packet-oriented networks IN, such as, for example, the Internet, intranet, extranet, H.323 network with a gatekeeper as a call controller, a local area network (LAN) or a company internal network (Corporate Network), for example, a Virtual Private Network (VPN).

_A subscriber A is connected to the network PSTN _A using a conventional telephone T and to the network IN _B a subscriber B using a SIP-capable telephone, for example a SIP client SC implemented in software. An Interactive Voice Response System IVR is assigned to the IN _IVR network. A first connection is provided between subscriber A and the IVR system, which comprises a bearer an end-to-end user channel TDM, RTP / RTCP _{A / IVR} . Furthermore, a second connection is provided between the system IVR and subscriber B, which acts as a bearer, an end-to-end user channel

RTP / RTCP _I VR _/ B includes. Finally, a direct connection is provided between subscriber A and B, which as bearer comprises an end-to-end user channel TDM, RTP / RTCP _{A / B.}

The combination of the line-oriented Bearer TDM with the packet-oriented Bearer RTP / RTCP is done by an intermediate Media Gateway MG for conversion between different, network-specific user channel technologies RTP / RTCP (Real Time [Control] Protocol) and TDM (Time Deviation Multiplex), the connection the signaling SS7 of the network PSTN with the signaling SIP of the networks IN is effected by interposed media gateway controllers MGC _{A / B} and MGC _I VR. In this case, the controller MGC _{A /} _B effects direct interworking between the different network-specific signaling protocols ISUP of the network PSTN and SIP _{B of} the network IN _B. On the other hand, a BICC or SIP_T protocol is used between the MGC _{A / B} and MGC _I VR controllers

14 Interworking between the different signaling protocols ISUP of the network PSTN and SIPTV _{R of} the network IN _IVR are used.

The gateway MG is controlled by the controller MGC _{A / B assigned to it} by a - preferably internationally standardized - protocol, for example MGCP (Media Gateway Control Protocol) or H.248. It is usually implemented as a separate unit which runs on a different physical device / hardware platform than the MGC _AB controller assigned to it.

It should be emphasized that the embodiments of the invention shown in this way, despite their sometimes very detailed representation of concrete network scenarios, are only of an exemplary nature and are not to be understood as restrictive. It is clear to the person skilled in the art that the invention works in all conceivable network configurations, in particular other interworking scenarios. In particular, the SIP protocols can be replaced by protocols from the H.323 family or other protocols with the same effect.

An exemplary embodiment of the invention is explained below in which the PSTN subscriber A, as a feature, expands a connection to the SIP subscriber B with the aid of the IVR system.

First of all, a first connection TDM, RTP / RTCP _{A /} IR is set up between subscriber A and the system IVR, which is assigned to the packet-oriented network IN _IV R. Because subscriber A is assigned to the line-oriented network PSTN, its line-oriented signaling ISUP is mapped to the packet-oriented signaling SIP and SIP_T and its line-oriented bearer TDM is mapped to the packet-oriented bearer during the transition between the networks

RTP / RTCP _{A / I} V _R implemented (and vice versa). For example, the SIP signaling SIP: Invite when interworking between

15 the ISUP protocol and the SIP protocol are mapped to the ISUP signaling 0: IAM. Likewise, the ISUP signaling 0: ACM and 0: ANM, with which the ringing of the telephone T and the acceptance of the call by the subscriber A are indicated, are mapped to the SIP messages 180: ringing and 200: OK. The first connection set up in this way comprises at least one bearer TDM, RTP / RTCP _IVR (usually bidirectional in a telephone conversation) for transmitting information between subscriber A and the system IVR.

Subsequently, the system IVR is informed by subscriber A of the data required for authenticating subscriber A and for identifying subscriber B. For example, subscriber A transmits a passcode and the subscriber B's number on the bearer TDM, RTP / RTCP _{A / IVR} .

In the further course, the second connection RTP _IVR / _B to subscriber B is set up by the system IVR using the communicated data. This leads to a ringing being displayed for the B-subscriber. This is communicated to the IVR system. The system IVR then sends a first signaling message to subscriber A. After receiving this message, subscriber A is informed of the ringing at subscriber B using a ringback. In this exemplary embodiment, subscriber A is assigned to network PSTN, so that the transmission of a ringback tone is desirable. This ringback tone is preferably generated in the media gateway MG. For this purpose, the first signaling message is received by the media gateway controller MGC _{A / B} and translated by the latter into an instruction for generating the ringback tone, which is transmitted to the media gateway MG using the MGCP protocol. Alternatively, the ringback tone could also be transmitted directly by the IVR system instead of the first signaling message, provided the IVR system knows which network type subscriber A is assigned to. This could be known, for example, on the basis of fixed default settings

16 on .

As soon as subscriber B accepts the established connection, subscriber A is informed of this with the aid of a second signaling message. This is also received by the Media Gateway Controller MGC _{A / B} and translated into an instruction to switch off the ringback tone, which is transmitted to the Media Gateway MG using the MGCP protocol. In the alternative embodiment, the ringback tone would be switched off immediately by the IVR system. This ends the ringback for subscriber A. In this way, ringback and ringing are consistently coordinated.

With the acceptance of the second connection RTP / RTCP _{TVR / B} at the latest, the two connections TDM, RTP / RTCP _{A /} IVR and RTP / RTCPIVR _/ B in the packet-oriented network IN are advantageously switched to a direct connection TDM, RTP / RTCP _{A / B.} This is achieved, for example, by transmitting the IP addresses of subscribers A, B in the relevant messages. For example, the.IP addresses could be transmitted in SIP messages INVITE, re-INVITE, 180 RINGING, 200 OK or ACK and there in matching SDP attributes.

The sender of the second signaling message depends on when the two connections are switched to a direct connection. If it occurs rather early at the start of establishing the second connection, the sender is <probably immediate subscriber B. If it occurs rather late, the immediate sender is more the IVR system, which previously received a corresponding (indirect) message from subscriber B - eg a SIP message 200 OK - has been communicated.

Summarized and generalized to subscriber A, which are assigned to any network, the individual steps from the establishment of the second connection can be shown as follows:

17 put :

It is clear to the person skilled in the art that the invention works with all relevant network configurations, in particular all interworking scenarios TDM <= IP. It is also clear to the person skilled in the art that the invention can also be used if there is no ISUP, BICC between the PSTN subscribers (ISDN, analog subscriber or also mobile subscriber) and the SIP or SIP-T subscribers. The above-mentioned method would then normally be carried out within switching centers. Interworking of NGN (Next Generation Network) participants such as VoDSL (Voice over Digital Subscriber Line), H323, etc. with SIP or SIP-T is also possible.

In conclusion, it should be pointed out that the description of the components of the communication network relevant to the invention is in principle not to be understood as restrictive. It is particularly obvious to a relevant specialist that terms such as subscriber, gateway, controller, etc. are to be understood more functionally than physically.

18 All functional units can in particular be partially or completely implemented in software / computer program products P and / or distributed over several physical devices.

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Claims

claims

1. Method for establishing a connection between a first subscriber (A) and a second subscriber (B) under

Mediation of an Interactive Voice Response System (IVR), which is assigned to a packet-oriented communication network (IN _ΪVR ), with the following steps:

Establishing a first connection (TDM, RTP / RTCP _{A /} I _VR ) between the first subscriber and the IVR system,

Establishing a second connection (RTP / RTCP _{IVR / B} ) between the IVR system and the second subscriber,

- Information of the second participant about the establishment of the second connection using a ringing, - Notification of the ringing to the IVR system,

Notification of the ringing to the first subscriber by sending a first signaling message from the IVR system,

Information of the first participant about the ringing at the second participant with the help of a ringback,

Acceptance of the second connection by the second subscriber,

- Notification of acceptance to the first participant by sending a second signaling message, - termination of the ringback.

2. The method according to claim 1, in which the second subscriber is identified taking into account data from the first subscriber, in particular whose call number is communicated to the IVR system by the first subscriber.

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3. The method according to any one of the preceding claims, in which at the latest with acceptance of the second connection in the packet-oriented network, the two connections of the IVR system on a direct connection (TDM, RTP / RTCPA _/ B) between the two participants without mediation by the IVR system be converted.

4. Method according to one of the preceding claims, in which, in the event that a SIP protocol according to one of the IETF standards RFC2543, RFC2543bis0x, RFC3261 or RFC3372 is used in the packet-oriented network, the first signaling message as re-INVITE with a " Alert Info "and the second signaling message is designed as re-INVITE without an" Alert Info ".

5. The method according to any one of the preceding claims, in which, in the event that the first subscriber is assigned to a line-oriented network, a ringback tone is transmitted instead of or as a result of the first signaling message and instead of or as a result of the second signaling message, the transmission of the ringback tone is set.

6. The method according to any one of the preceding claims, wherein the announced ringing is displayed to the first participant with the aid of an information window.

7. Arrangement - in particular packet-oriented, integrated multimedia network (IN) or hybrid network (IN, PSTN) - comprising means which are set up to carry out all steps of a method according to one of the preceding method claims.

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8. Device comprising

- Means that are set up to carry out those steps of a method according to one of the preceding method claims that are caused by the device, - Means that are used to carry out prescribed interactions of the device with other devices, of which the remaining steps of the method be carried out.

9. The device according to the preceding claim, which is designed as a computer program product (P) and the means are designed as program codes that are executed by at least one processor to carry out the method.

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