ES2375304T3 - CAPACITY NEGOTIATION IN A TELECOMMUNICATIONS NETWORK. - Google Patents

Abstract

A method of negotiating protocol options between the first (A), second (B) and third (C) nodes in a telecommunications network that uses separate Call Control and Bearer Control protocols, the method comprising: transmitting a first call control message from the first node (A) to the second node (B) specifying the protocol options supported by the first node (A), transmit a second call control message from the second node (B) to the third node (C) specifying the protocol options supported by the two nodes first (A) and second (B); and select a protocol option from the protocol options specified in the second control message.

Description

Negotiation of capacity in a telecommunications network.

Field of the Invention

The present invention relates to the negotiation of capacity in a telecommunications network and in particular, but not necessarily, to the negotiation of a suitable conversation encoder.

Background of the invention

Telecommunications networks today rely heavily on Signaling System No. 7 (SS7) (Signaling System No. 7) as the mechanism to control call connections and to handle the transfer of signaling information between signaling points of the networks. Typically, one or more application and user parts at a given signaling point will make use of the SS7 to communicate with peer and user application parts at some other signaling point. Examples of user parts are ISUP (ISDN User Part-User Part of ISDN) and TUP (Telephony User Part-Part of Telephony User) while examples of application parts are INAP (Intelligent Network Application Part-Part of Application of Intelligent Network) and MAP (Mobile Application Part-Part of Mobile Application). The conventional SS7 protocol stack includes Message Transfer Parts (MTP1, MTP2, and MTP3) that handle formatting and signaling messages for transport on the physical layer as well as various routing functions.

There has been considerable interest in the latter among the telecommunications community in using non-standard signaling transport mechanisms (i.e. non-conventional within the telecommunications industry) in telecommunications networks instead of conventional SS7 mechanisms. The reasons for this are related both to improvements in efficiency as well as potential cost savings. Much consideration has been given, for example, to the use of Internet Protocol (IP) networks to transport signaling information between signaling points. IP networks have the advantage that they make efficient use of transmission resources using packet switching and are of relatively low cost due to the widespread use of technology (as opposed to specialized telecommunication technology). There is also interest in using transport mechanisms that include AAL1 / 2/5, FR etc.

The ISUP standard dealing with the establishment and control of call connections in a telecommunications network is closely linked to the signaling transport mechanism via SS7 and does not lend itself directly for use with other non-standard transport technologies such as IP and AAL2. Thus, several standardization bodies that include ITU-T, ETSI, and ANSI are currently considering the specification of a signaling protocol for call control, which is independent of the underlying transport mechanism. This may appear to be separated from the protocol, Bearer Control functions that merely refer to setting the parameters (including the start and end points) of the "tube" by means of which user plane data is transported between nodes, and that are specific to the transport mechanism. The new protocol, called Transport Independent Call Control (TICC), maintains the functions of Call Control so that the services requested for a call between callers and given calls (for example call forwarding), and total routing of user plane data.

The new network architecture resulting from the separation of the levels of Call Control and Bearer results in the appearance of an air interface between a Call Control entity and a Bearer Control entity, where these entities are called Door Controller Media Link and Media Gateway respectively. The air interface is referred to below as X-CP, examples of which are the MEGACO protocol of the IETF and the H.248 protocol of ITU Study Group 16 (SG 16 - Study Group 16).

Another example for a separation of call and bearer control is EP 0690640, which describes a telephone signaling system that is operative outside the traditional call and data transmission routes to provide start and end signaling.

Traditionally, fixed telephone networks use Pulse Code Modulation to transport user plane data, for example voice, facsimile, etc., between network nodes. Modem cellular networks on the other hand often use one or more encoders / decoders (called "codecs") to compress voice signals for efficient transmission over the air interface and within the cellular networks themselves. Where a telephone call connection extends between two networks (or terminals) that support different or multiple conversation codecs, a negotiation between the terminals can take place to decide on the appropriate codec. If this negotiation is not carried out, the result may be a request to transcode at the interface between the networks, that is, conversion from one form of conversation coding to another. Transcoding is expensive in terms of resources, significantly degrades quality, and introduces a delay in treatment time. Codec negotiation is therefore the preferred option.

In addition to codec negotiation, there is often a need in the conventional telecommunications network to negotiate other functionality and other parameters. For example, it may be desirable to negotiate security capabilities such as voice encryption and data encryption between terminals or nodes in telecommunications networks.

Compendium of the present invention

In accordance with a first aspect of the present invention there is provided a method of negotiating a call capability between signaling points in a telecommunications system, the method comprising:

send a capacity preference or prioritized list of preferences from a start signaling point to an end signaling point or a signaling transfer point, at the Call Control level; Y

return a capacity acceptance from an end signaling point or signaling transfer point at the Call Control level, if the termination signaling point or signaling transfer point accepts a preference sent by the signaling point of start.

It will be noted that in some cases, for example where the termination signaling point or signaling transfer point does not accept a capacity preference (or one of a preference list) sent by the initiation signaling point, a return signal will be returned. non-acceptance message in which case a default capacity is assumed by both points. Alternatively, a default message may be returned indicating that the default codec will be used. If no codec can be agreed, then in certain situations a call may be released due to network incompatibility.

The present invention is particularly suitable for negotiating conversation codec capabilities between signaling transfer points located in different telecommunications networks. For example, in Japanese telecommunications networks, the invention can be used to negotiate the use of one of VSELP, PSI-CELP or μ-law coding, where the coding of μ-law coding is the default coding. However, the invention is also applicable to negotiate other capabilities that include security capabilities (eg voice encryption and data encryption). The protocol used to carry out the negotiation can be TICC, or it can be a specific protocol also used at the CC level, that is, a User Plane Capability Negotiation protocol.

Where the Call Control and Bearer Control levels are controlled by separate protocols, a signaling point reacts to the selection of a capacity at the Call Control level by notifying the Bearer Control level, if the selection affects the level of carrier. If appropriate, notifications can be subsequently sent to the bearer level between bearer switching points to allow for the establishment of appropriate bearer level resources.

Preferably, the signaling point or signaling transfer point is a Media Gateway Controller. More preferably, the Media Gateway Controller communicates with one or more Media Gateways that exist at the Bearer Control level.

Despite the fact that certain options supported by a capacity at the Bearer Control level may be known at the Call Control level, the Call Control level will not necessarily know the current availability of those options at the Control level. Bearer Preferably therefore, upon receiving a capacity preference or a prioritized list of preferences at an end signaling point or a signaling transfer point, the Call Control level conducts a negotiation with the Bearer Control level. to determine option availability at the Bearer Control level. More preferably, this negotiation takes place between a Media Gateway Controller of the Call Control level and a Media Gateway of the Bearer Control level.

Where a prioritized list of preferences is sent from a starting signaling point to a Media Gateway Controller, the Controller preferably modifies the list to remove preferences that it knows are not supported by the associated Media Gateway. The Controller then sends the modified list to the Media Gateway that selects the highest priority preference that the Media Gateway can support at that time. The Gateway can then reserve the necessary resources for that preference and advises the Media Gateway Controller about the preference. The Media Gateway Controller can then return a capacity acceptance to the Startup Media Gateway Controller. In accordance with a second aspect of the present invention a signaling point is provided arranged to negotiate a call capability with another signaling point in a telecommunications system, the method comprising:

means for sending a capacity preference or prioritized list of preferences to an end signaling point or signaling transfer point, at the Call Control level; Y

means for receiving a capacity acceptance from an end signaling point or signaling transfer point at the Call Control level, whose capacity acceptance is sent if the termination signaling point or signaling transfer point accepts a preference sent by the start signaling point.

In accordance with a third aspect of the present invention, a Media Gateway Controller of a telecommunications system is provided, the Media Gateway Controller comprising:

a means for receiving a capacity preference or prioritized list of preferences from a Peer Media Gateway Controller, where said capacity preference or prioritized list of preferences refers to a connection to be established on the telecommunications system;

a means to communicate with a Media Gateway associated with the Media Gateway Controller to determine the availability of the preference or preferences in the Media Gateway; Y

a means for returning a capacity preference acceptance message to said Peer Media Gateway Controller depending on the availability determined in the Media Gateway.

In accordance with a fourth aspect of the present invention, a Media Gateway of a telecommunications system is provided, the Gateway comprising:

a means for receiving a capacity preference or a prioritized list of preferences from a Media Gateway Controller, in which said capacity preference or prioritized list of preferences refers to a connection to be established on the telecommunications system ;

a means to select a preference based on the availability of preferences in the Media Gateway; Y

a means to send the selected preference to said Media Gateway Controller.

According to a first embodiment of the invention there is provided a method of negotiating protocol options between first, second and third nodes in a telecommunications network using separate call control and bearer control protocols, the method comprising:

transmitting a first call control message from a first node to the second node specifying the protocol options supported by the first node;

transmit a second call control message from the second node to the third node specifying the protocol options supported by the two first and second nodes; Y

select a protocol option from the protocol options specified in the second control message.

The said step of selecting a protocol option from the protocol options specified in the second control message can be carried out by the third node.

Each of said call control messages may include a level of preference associated with each specified protocol option.

The first node can be a start node. The third node can be an end node.

The method may also include a step of determining whether the bearer level that determines whether the bearer level between the first and second nodes is affected by said selection stage, and if the bearer level is affected by taking actions to modify the Bearer level parameters between the first and second nodes.

The method may also include determining whether the bearer level between the second and third nodes is affected by said selection stage, and if the bearer level is affected by taking actions to modify the bearer level parameters between the second and third nodes. .

In accordance with another aspect of the present invention there is provided a method of establishing iq of a call connection in a telecommunications system in which the Call Control protocol is independent of the bearer transport mechanism, the method comprising:

negotiate a first conversation codec between a system start signaling point and a first end signaling point;

establish a call connection on the transport mechanism between the start signaling point and said first end signaling point depending on said first conversation codec;

subsequently, negotiate a second different conversation codec between said first termination signaling point and a second new signaling point;

notify the starting signaling point of the second conversation codec; and establish a call connection between the start signaling point and said second end signaling point, where said first end signaling point acts as an intermediate signaling point and where the first mentioned call connection is modified if It is necessary to support the said second conversation codec.

It will be apparent that aspects of the present invention enable the continuous transfer mode of a call connection between different termination signaling points, modifying the connection between the starting signaling point and the original termination signaling point to reflect the new codec . The end-to-end connection is completed by establishing a connection between the original termination signaling point and the new termination signaling point based on the new codec. Preferably, this last connection is established after the first connection has been modified, although this need is not the case.

It will also be apparent that the method of the present invention is only applicable when the negotiation of the conversation codec between the original end signaling point and the final completion signaling point does not result in the first conversation codec. If the result is the first conversation codec, then there may be no need to modify the original connection.

Preferably, the step of notifying the start signaling point of the second conversation codec comprises sending an appropriate Call Control message (CC) from the original end signaling point to the starting signaling point. This CC message can be a Modification request message.

Preferably, the Call Control protocol is a Transport Independent Call Control (TICC-Transport Independent Call Control) protocol.

The present invention is applicable to the evolution of the existing telecommunication network such as mobile phone networks based on GSM, DAMPS, PDC, etc., as well as to future generation networks such as UMTS.

In accordance with another aspect of the present invention a telecommunications system is provided in which the Call Control protocol is independent of the bearer transport mechanism, the system comprising:

means for negotiating a first conversation codec between a system start signaling point and a first end signaling point;

a means for establishing a call connection on the transport mechanism between the start signaling point and said first end signaling point depending on said first conversation codec;

a means for subsequently negotiating a second different conversation codec between said first termination signaling point and a second new termination signaling point;

a means to notify the starting signaling point of the second conversation codec; Y

a means for establishing a call connection between the start signaling point and said second end signaling point, where said first end signaling point acts as an intermediate signaling point and where the first mentioned call connection is modified if necessary to support the said second conversation codec.

In accordance with another aspect of the present invention, a signaling point of a telecommunications network is provided in which the Call Control protocol is independent of the bearer transport mechanism, the signaling point comprising:

a processing means for negotiating a first conversation codec with a first termination signaling point;

a means for establishing a call connection on the transport mechanism between the start signaling point and said first end signaling point depending on said first conversation codec;

means for receiving a notification of a second conversation codec that has been negotiated between said first termination signaling point and a second new termination signaling point; Y

a means for modifying the first mentioned call connection if necessary to support said second conversation codec.

In accordance with another aspect of the present invention there is provided a method of establishing a call connection between the first and second mobile telephone networks, at least one of which comprises a Tandem Free Operation device (TFO). located outside the part or parts of the Radio Access Network of the mobile telephone network and in which the Call Control protocol is independent of the carrier's transport mechanism, the method comprising the steps of:

conducting a negotiation between said TFO device of one of the mobile telephone networks and a TFO device paired with the other mobile telephone network to determine a suitable conversation codec; Y

Notify the aforementioned Radio Access Network of the conversation codec determined by sending a Call Control message (CC - Call Control) from the associated TFO device.

It may be the case that each of the mobile telephone networks comprises a Tandem Free Operation (TFO) device located outside the part or parts of the Radio Access network of the mobile telephone network. However, this need may not be the case, and one of the mobile telephone networks may have a TFO device within the Radio Access Network.

The two mobile telephone networks can be coupled to each other by means of another PSTN. Preferably, said negotiation is carried out using Disagreement Resolution and Optimization Procedure in the TFO protocol, sending TFO messages using in-band signaling.

Preferably, said or each TFO device located outside a Radio Access network is located or located in an MSC Gateway (GMSC-MSC Gateway) that provides an interface between the mobile telephone network and foreign networks , for example a PSTN.

The present invention is particularly suitable for establishing a call connection between subscribers, one of which is a subscriber of a Universal Mobile Telecommunication System (UMTS). In this case, the Radio Access Network is preferably a UMTS Terrestrial Radio Access Network (UTRAN - UMTS Terrestrial Radio Access Network), the TFO device being located at the border of the UMTS core network.

In accordance with another aspect of the present invention there is provided an apparatus for establishing a call connection between the first and second mobile telephone networks, at least one of which comprises a Tandem Free Operation device (TFO). located outside the part or parts of the Radio Access Network of the mobile telephone network and in which the Call Control protocol is independent of the carrier's transport mechanism, the apparatus comprising:

a means for conducting a negotiation between said TFO device of one of the mobile telephone networks and a paired TFO device of the other mobile telephone network to determine a suitable conversation codec; Y

a means for notifying the aforementioned Radio Access Network of the determined conversation codec, by sending a Call Control message (CC - Call Control) from the associated TFO device.

According to another aspect of the present invention, a method of establishing a call connection between the first and second mobile telephone networks is provided, each of which comprises a Tandem Free Operation device (TFO) located outside of the part or parts of the Radio Access Network of the mobile telephone network and in which the Call Control protocol is independent of the carrier's transport mechanism, the mobile telephone networks being coupled by means of a Public Switched Telephone Network (PSTN - Public Switched Telephone Network), the method comprising the stage of:

carry out a negotiation between two mobile telephone terminals that subscribe to the first and second mobile telephone networks respectively to determine an appropriate conversation codec, where the negotiation is carried out using exchanged Call Control protocol signaling messages between the parts of the Radio Access Network and the respective TFO devices, and ISUP messages sent between the TFO devices.

For example, a list of codecs that are available for a mobile home telephone network can be sent from the TFO device of that network to the paired TFO device using an ISUP Initial Address Message (IAM). . The message may additionally include the codec preferred by the home mobile telephone network. A subsequent ISUP message sent in the opposite direction will indicate the type of codec selected by the terminating mobile telephone terminal. In order to enable this negotiation procedure, it may be necessary to modify the ISUP standard.

In accordance with another aspect of the present invention there is provided an apparatus for establishing a call connection between the first and second mobile telephone networks, each of which comprises a Tandem Free Operation (TFO) device located outside the part or parts of the Radio Access Network of the mobile telephone network and in which the Call Control protocol is independent of the carrier's transport mechanism, the mobile telephone networks being coupled by means of a Public Switched Telephone Network (PSTN), including the device:

a means for conducting a negotiation between two mobile telephone terminals that subscribe to the first and second mobile telephone networks respectively to determine an appropriate conversation codec, where the negotiation is carried out using Control protocol signaling messages Call exchanged between the parts of the Access Network and the respective TFO devices, and ISUP messages sent between the TFO devices.

Brief Description of the Drawings

Figure 1 illustrates a number of signaling points in a telecommunications network; Y

Figure 2 (case A) illustrates signaling flows between the signaling points of Figure 1 according to a first aspect of the invention;

Figure 2 (case B) illustrates signaling flows between the signaling points of Figure 1 according to a second aspect of the invention;

Figure 3 illustrates a telecommunications network comprising Media Gateway Controller and Media Gateway;

Figure 4 is a flow chart illustrating a generic capacity negotiation process used in the network of Figure 3.

Figure 5 illustrates signaling flows where an intermediate node is involved in the negotiation between a start node and an end node;

Figure 6 illustrates another set of signaling flows where an intermediate node is involved in the negotiation between a start node and an end node;

Figure 7 schematically illustrates a telecommunications system of known design;

Figures 8 schematically illustrate a telecommunications system;

Figure 9 illustrates a first set of signals associated with codec negotiation in the system of Figure 8 and in accordance with a first aspect of the present invention; Y

Figure 10 illustrates a second set of signals associated with codec negotiation in the system of Figure 8 and in accordance with a second aspect of the present invention.

Detailed Description of Certain Aspects

A portion of a telecommunications network comprising two signaling points called hereafter Nodes A and C is illustrated in Figure 1. These nodes may be for example telephone exchanges or switches and may belong to the same network operator or to different network operators. . In the present example, Node A is the Start Node to which a caller is connected (not shown) while Node C represents an End Node to which the call is connected. Each of the signaling points comprises a part of Call Control (CC) (Call Control) and a part of Bearer Control (BC) (Bearer Control), that is the Call Control and Bearer Control functionalities They are separated into two distinct protocol layers. The CC parts form a level of Call Control that is responsible for carrying out functions such as call forwarding as well as other routing and control functions. The parts of BC are responsible for establishing and sizing tubes between parts of BC to transport user plane data.

Considering now the level of BC, this comprises a carrier network that can be for example an IP network. Within the IP network there are one or more bearer switching points, although only one such point is illustrated in Figure 1 (Node B). For the IP network, these bearer switching points will be IP routers. It will be noted that where the carrier network is an ATM or AAL2 network, the bearer switching points would be ATM or AAL2 switches respectively.

In the event that a caller initiates a call, for example by picking up his telephone, the starting signal point Node A receives information from the source (ie, caller) at the CC level that defines the requirements of bearer resources. The starting signaling point determines, based on the source information and / or the capabilities of the signaling point and your home network, a list of possible options that should be negotiated with the completion signaling point Node C .

Node A then indicates the list of possible options in a message to the part of CC with which it wishes to negotiate a specific capacity (Node C in Figure 1). The message indicates the level of preference for each option.

Node C uses the priority levels specified by Node A for each option in the list to select the most preferred option it supports, that is, Node C does not choose an option if it supports another option in the list for which Node A has indicated a higher preference. Node C indicates the option selected to Node A.

If the results of this negotiation affect the bearer connection level, appropriate actions are taken at the bearer level to adapt to the negotiation result. This is done by the BC protocol. Two cases are possible: send the bearer modification (case A in Figure 2) or return the bearer modification (case B in Figure 2).

An example of a capacity that can be negotiated using the previous method is the codec capacity. Generally, the conversation is transcoded to PCM in the radio access network because this is the only conversation format allowed by traditional telephone networks. However, since transcoding significantly impairs the quality of the conversation, some cellular standards (eg GSM, PDC) specify methods to avoid transcoding when a connection between two compatible terminals (for example two GSM terminals with a codec has been established common). Since most of the current mobile terminals support several codecs, these methods also implement codec negotiation but have some significant drawbacks compared to the generic negotiation mechanism presented here.

Conversation codecs are tightly coupled to mobile environments (in PDC, codec negotiation is a service of the Mobile Application Part (MAP) protocol or does not optimize the use of hardware and transmission resources ( In GSM, codec negotiation is part of the Tandem Free Operation (TFO) protocol.Two transcoders (TRAUs) and a 64Kbps channel arc always assigned to a conversation connection regardless of whether the transcoding has place really or not.) Currently, different mobile standards deal with the problem in different ways and there has previously been no possibility of harmonizing solutions.

Another example of a capability that may require negotiation is security capabilities (voice encryption, data encryption, etc.). This is currently the great interest in public telecommunication networks and will occur to a greater extent in the near future. However, there are multiple ways to protect user plane information against unwanted observers. There are multiple encryption and encryption algorithms of currently deployed data and new ones that appear continuously. Therefore, a way to negotiate security capabilities in public telecommunication networks will be required shortly and this is provided by the present invention.

The above description has generally referred to a level of BC and a level of CC. Figure 3 illustrates in more detail a telecommunications system that has separate DC and BC levels. The CC level comprises a number of Media Gateway Controllers (MGC_A, MGC_B, MGC_C) while the BC level comprises a number of Media Gateway (MG_1 to MG_6). A first pair of Media Gateway (MG_1, MG_6) provides a carrier connection over an AAL2 network, while a second pair of Media Gateway (MG_2, MG_5) provides a carrier connection over an IP network. A third pair of Media Gateways (MG_3, MG_4) provide a carrier connection over an STM network. As can be seen from Figure 3, each of the Media Gateway Controllers is arranged to control two or three Media Gateways through an open interface (X-CP_1 to X-CP_3).

As described above, capacity negotiation takes place at the CC level, that is, between Media Gateway Controllers. In addition, X-CPs provide a means for the Media Gateway to "inform" its Media Gateway Controllers of the capabilities and options that the Media Gateway supports (for example capabilities = compressed conversation, options = list of codecs). The X-CPs also provide means in which a Media Gateway may be involved in a negotiation capacity, thus ensuring that a capacity selection is based not only on the capabilities specified for the Gateway but also on the actual availability of those capabilities. If this is not the case, there is a danger that a Media Gateway may not have the resources to support an option negotiated on its behalf by a Media Gateway Controller. Consider for example the case in which a Media Gateway Controller needs to provide an Intelligent Network (IN) service for a call and that requires in-band ads. If the Media Gateway Controller selects a codec by itself, based on a knowledge of the codecs supported by the Media Gateway, the Media Gateway may not actually have a TRAU available for that codec. The Media Gateway will therefore reject the connection, taking place a call drop.

To illustrate the solution to this problem, consider the situation in which a Media Gateway Controller receives a connection establishment message from a Media Gateway Controller that contains a list of possible options related to a specific negotiable capacity. The message indicates the level of preference for each option. The Receiving Media Gateway Controller reads the received list and removes from the list those options that it knows are not supported by the Media Gateway that has been selected for connection. The Media Gateway Controller then sends the modified list of options to the Media Gateway on the X-CP interface.

The receiving Media Gateway chooses the option that has the highest priority and that the Gateway is really capable of supporting. The Media Gateway then takes the resources it needs to support this option and communicates the option to the Media Gateway Controller over the X_CP interface. Upon receiving the selected option, the Media Gateway Controller signals the selection to the Media Gateway Gateway as described above.

The generic capacity negotiation process described above is also illustrated by the flow chart of Figure 4.

In many countries, local authorities ask operators to intercept calls for legal purposes. In order to allow legal interception, operators normally require access to user plane data (that is, the actual conversation or other data transported by the so-called "tube" at the bearer level) for each call within their networks . With the introduction of TICC, user plane data can be transported on a variety of technologies (for example, ATM, IP etc.) that allow the transport of encrypted data from the user plane. Therefore, unless the Legal Interception point can determine the security and codec characteristics (coding and decoding) of the user plane data, the call cannot be intercepted successfully. Therefore, the operator needs to know the encryption and encryption (security) algorithms used within each BC section (between two BC nodes) of a call.

On the other hand, transcoding and translation of security algorithms significantly reduce the quality of service of a call and introduce extra delays. Therefore, it is desirable to reduce the number (or avoid if possible) of transcoding points and translation points of the security algorithm required end-to-end for each call.

A mechanism to solve this problem will now be described with reference to the sequence of operations shown in Figure 5. The Call Control part of Node A (see Figure 1) transmits a message to the Call Control part of Node B which includes a list of options supported by Node A, along with a level of preference for each such option. Although the mechanism is particularly suitable for use in TICC, it could also be used as an independent protocol specifically adapted to carry out a negotiation, and this is indicated in Figure 5 by means of the initials UPCN (User Part Capability Negotiation) Capacity of the User Part).

The Call Control part of Node B then sends a message to the Call Control part of Node C, which includes a list of options supported by both Node A and Node B, along with a level of preference for each option. Node C then selects the supported option with the highest level of preference, and returns a message to Node B that includes the selected option. Node B in turn sends a message to Node A specifying the selected option.

Figure 5 then shows two possible sequences of operation, labeled as case A and case B. In Case A, if the bearer level is affected by the negotiation at the Call Control level described above, Node A takes appropriate action. at the carrier level. For example, it may be necessary for Node A to change the size of the tube that is needed to support the selected option. A BC message is then sent from Node A to Node B specifying whatever parameters are needed. Node B then analyzes the selected option and takes appropriate action at the bearer level if the bearer level is affected by the negotiation. A BC message is then sent from Node B to Node C specifying any parameters that are needed for the bearer level between Nodes B and C.

In Case B the sequence of operations at the bearer level begins at Node C and ends at Node A. Node C analyzes the selected option and takes the appropriate action at the bearer level, if the bearer level is affected For the negotiation. A BC message is then sent from Node C to Node B specifying whatever parameters are needed. Node B then analyzes the selected option and takes appropriate action at the bearer level if the bearer level is affected by the negotiation. A message from BC is then sent from Node B to Node A specifying any necessary parameters.

It should be understood that cases A and B are alternative, but other cases are also possible. In view of this, it should be noted that the BC protocol between Nodes A and B may be different from that between Nodes B and C, for example the protocols could be ATM and IP respectively. It may be necessary to modify the parameters for the bearer level connection between one pair of Nodes, but not between the other pair of Nodes.

It should also be noted that the negotiation described above may take place during the call establishment or during the call itself. The latter can apply, for example, where a user wishes to start using encryption during a call.

This procedure is capable of reducing the number of points in the network at which the generic capacity needs to be changed. In particular, the mechanism specified above can be used to minimize the number of transcoding points or security translation algorithm points for end-to-end calls that cross multiple CC sections. The solution applies to calls that cross one or multiple networks.

The GCN mechanism includes having the Start CC Node include the list of options with its preference level and the End Node select a supported option that uses the preference level indicated by the Start Node. The negotiation mechanism mentioned above is useful when the negotiation is carried out between two CC Nodes only. The proposed adaptation expands the GCN mechanism to cases in which more than two CC Nodes intervene in the negotiation. That is, cases in which the call crosses multiple CC Nodes that belong to one or more telecommunications operators. The adaptation involves the following sequence of operations:

The Startup CC Node sends its list of supported options with the level of preference associated with each one.

Transit CC Nodes analyze the list of options received, delete the unsupported options from the list and send the list to the next Node.

The End CC Node: analyzes the list of options received with their associated priorities and selects the supported option with the highest indicated priority.

In the case of legal interception, it allows the Legal Interception point (that is, the Node in which the interception is going to take place) to receive the security and coding characteristics of the user plane data to allow successful interception Of the call.

With the proposed new architecture of Figure 1, where the Call Control protocol is independent of the transport mechanism, codec negotiation must be carried out by the TICC for a call before a plane transport connection is established of user for this call. The reason is that the amount of transport resources required to support a call depends on the codec selected for this call (that is, the result of the codec negotiation procedure). In some cases, a codec is initially selected for a call but, due to a change in circumstances, this codec must be changed to a new codec at a later stage of the call. Two important situations where this happens are the following:

1) Interactions with Intelligent Network (IN) services for the purpose of providing announcements and redirection of calls to a called party either automatically or based on an entry from a caller. A first codec must be selected to provide the announcements. Later the call is sent / redirected to another party and a second codec is used.

2) Call Forwarding on No-Reply (CFNR), which is a so-called supplementary service. A first codec is negotiated between parts A and B, following which the call is sent to a part C because B fails to answer at a predefined time. A second codec is requested based on the requirements of part C.

It is recognized that, in both cases 1) and 2) above, as well as in other related situations, a call connection must be established on the transport mechanism depending on the first codec selected, to carry announcements, tones, etc. When the codec is changed then the call connection may need to be modified to support the new codec.

A signaling process will now be described with reference to Figure 6 for use in networks using the TICCl protocol and that is capable of establishing an end-to-end telephone connection for a conversation call, where the call originates from a Start signaling point Node A and is initially directed to a first node Node B or signaling point: The call is then routed to some other termination signaling point Node C. The process illustrated specifically refers to the request of an IN service, where a caller is initially connected to an IN network node that directs a previously recorded message to the caller before routing the call to a final end signaling point, or to a CFNR service, where a call is sent to a new termination signaling point after the call, connected to the initial termination signaling point, fails to answer the call.

The signaling sequence comprises the following sequential stages:

Node A establishes a call to Node B. Codec X is selected for this call.

A transport connection with adequate transport resources for the selected codec is established between Nodes A and B.

Node B diverts / redirects the call to Node C. Node C does not support codec X and selects codec Y.

Node B asks Node A to modify the codec choice for this call from codec X to codec Y.

If necessary, the transport connection between Node A and Node B is modified to fit codec Y. Figure 6 illustrates both the case in which the modification to the transport connection is made in the forward direction and the alternative case in which the modification is made in the backward direction.

A suitable transport connection for codec Y is established between Nodes B and C.

The TICC completes the call establishment (TICC ACM + TICC ANM).

The TICC is based on the ISUP, therefore the names of the signaling message are taken from the ISUP. However, ISUP does not include codec negotiation or codec modification procedures, so a new pair of TICC messages (request / confirmation of TICC modification) is required to provide codec modification capabilities.

It should be noted that, while Figure 6 illustrates only that the modification made to the transport connection from Node A to Node B can be made either in the forward or backward direction, other modifications of the transport connection may be also done well in the forward or backward direction. Also, the transport connections can be established either in the forward or backward direction.

The described mechanism can be used to establish a conversation call connection between more than three signaling points or nodes. For example, one or more transit nodes may be present between Node A of the starting signaling point and Node B of the original termination signaling point and / or between Node B of the original termination signaling point and Node C of the final completion signaling point. It will also be noted that the call can also be transferred or delayed from Node C to another signaling point (ie, Node D). This would imply that a negotiation determined whether codec Y is suitable for Node D, and if not a possible modification for the transport connection between Node A and Node B and between Node B and Node C. This process can be extended to any number of nodes.

Conventional Public Switched Telephone Networks (PSTN) digitally encode conversation data for transmission using Pulse Code Modulation (PCM). On the other hand, digital mobile telephone networks make use of more advanced coding techniques such as CELP coding and Adaptive Multi-Rate (AMR) coding, which achieves higher compression rates than can be achieved. be achieved with PCM. In many mobile networks, conversation coding and decoding is carried out in the mobile terminals themselves. On condition that a call is made between two mobile terminals both registered with the same network it may be possible to transmit coded conversation data from end to end.

In the event that a call is from a mobile terminal registered with a mobile network, to a terminal that is a subscriber of a "foreign" network, an end-to-end transmission of encrypted conversation data may not be possible, depending on the nature of the foreign network and any intermediate networks that connect the mobile home network with the foreign network (the same will occur of course where the call originates from the foreign network).

Consider the telecommunications system of Figure 7 that illustrates two third generation networks Universal Mobile Telecommunication System (UMTS) 1,2 that are coupled by means of a conventional PSTN / ISDN 3 network UMTS 1.2 networks comprise a UMTS Terrestrial Radio Access Network (

UTRAN) (UMTS Land Access Network) 4 each with Radio Network Controllers (RNCs) 5 and Base Transceiver Stations (BTSs) 6. A UTRAN 5 passes data Conversation between a mobile terminal (not shown) and a Mobile Switching Center (MSC) 7 that routes incoming and outgoing call connections.

Assume that a call originates from a subscriber of one of the UMTS 1 networks and is made to a subscriber of the other of the UMTS 2 networks. The call is routed through PSTN 3 using respective Transition 8 nodes - one of which is an MSCs Gateway (GMSCs) - of the UMTS 1,2 networks. As already noted, PSTN 3 uses PCM to encode conversation data. Now it is important that any conversation data transferred through PSTN 3 be in a form that can be understood by that network. This is necessary, for example, to allow the PSTN 3 to insert operator announcements into a conversation call, to carry out voice-pointing services, etc., as well as to allow the operator of the PSTN 3 to monitor the call. , for example, for security purposes. It is therefore necessary to "transcode" conversation data in the GMSCs 8 of the UMTS 1.2 networks before passing the data to the PSTN 3, that is, the conversation data is converted from a conversation coding format from a mobile network to PCM. Similarly, PCM data received in GMSCs 8 must be converted to the appropriate mobile network conversation encoding format

Transcoding consumes considerable treatment resources in the GMSC 8 and also causes a noticeable degradation in the quality of the conversation. In order to at least partially solve these disadvantages, Tandem Free Operation (TFO) devices can be introduced into the conversation connection in the GMSCs 8. Outgoing conversation data continues to become PCM, but at least The significant bits of each PCM sample are "stolen" by the TFO device. Stolen bits form a channel that has enough bandwidth (i.e. 8Kbits / sec) to transport the original encoded data. The TFO device in the terminating UMTS network reassembles the encoded data to send it to the associated UTRAN while the received PCM data is deleted (unless they have been modified by the PSTN, for example, by adding an operator announcement). In this way, the TFO makes PCM data available for PSTN 3, while still allowing end-to-end transmission of efficiently encoded conversation data.

In the event that intermediate devices within the PSTN / ISDN alter the PCM bit stream, the TFO devices detect the change and "fall back" to pass the PCM coded conversation between the TFO devices, that is , no longer transmit compressed voice data.

The conversation encodings available for a mobile network depend on the nature of the network and possibly the nature of a terminal using the network. It will be clear that the end-to-end use of a single encoder is only possible when both networks are capable of using the same encoder. Assuming that the GMSCs of two mobile communication networks are aware of the encoder capabilities of the networks to which they belong, they may negotiate and agree on a common codec. Actually, an appropriate protocol is provided in GSM recommendation 08.62 (version 7.0.0. Version 1998) of the ETSI.

A problem arises when trying to implement TFO in mobile phone networks such as those illustrated in Figure 7, where TFO devices are located on the edge of mobile phone networks, that is, outside the UTRAN (conventionally the TFO devices are located within radio access networks). There is currently no mechanism to exchange information, regarding negotiated codecs between TFO devices, between TFO devices and radio access networks, in such networks.

Figure 8 illustrates a modified system architecture in which the GMSCs of the two UMTS networks have TFO devices 9 incorporated, while Figure 9 illustrates signaling associated with call establishment between two nodes in respective UMTS mobile telephone networks. . The outermost nodes represent MSCs (according to nodes 7 of Figure 7), while the two nodes 8 in the middle are each connected to each other from the outermost node and the PSTN / ISDN 3 represents GMSCs.

The call establishment signaling within the UMTS networks is carried out at the Call Control level (for example, TICC), the call establishment being initiated by an Initial Address Message (IAM). sent from an MSC to the associated GMSC. This IAM uses a Generic Capabilities Negotiation (GCN) mechanism to determine a number of parameters for the call connection. In particular, the IAM contains a list of codecs supported by the startup UMTS network, as well as the preferred codec. The start side of the GMSC selects a codec from the transmitted codec list, and signals its selection back to the MSC in the message (codec 1 Selected). Subsequently, a call connection is established at the bearer level (for example, AAL2 or IP) with sufficient bandwidth to support the selected codec.

To allow end-to-end codec negotiation, it has been proposed to add the GCN to the ISUP protocol. This end-to-end codec negotiation will maximize the possibility that endpoints use the same type of codec. If the endpoints use the same voice coding algorithm, the TFO is able to pass the compressed voice through the PCM network without degrading the voice quality due to unnecessary transcoding. This also allows transmission savings by minimizing bearer requirements within the start and end networks that support compressed voice (for example, AAL2 or IP carrier transport).

Figure 9 illustrates the use of enhanced ISUP messages by GCN to fill the "gap" signaling between the two UMTS networks. In the terminating UMTS network, an IAM is sent using TICC from the GMSC to the MSC. The MSC in this case accepts the use of codec 1 and points this back to the GMSC again using TICC. Subsequently, the bearer level connection is established in the terminating UMTS network. The PSTN transmits the acceptance of the codec to the GMSC on the start side. As the originally proposed codec has been accepted, there is no need to change the bearer level connection in the startup UMTS network. However, if there is a change in the codec, this must be sent from the GMSC from the start side to the MSC, so that the bearer level connection can be modified, for example, to increase the bandwidth of the connection . In Figure 9, the ACM indicates an Address Complete Message and the ANM indicates a Answer Message.

Figure 10 illustrates a second embodiment of the invention. This solution is based on the optional Mismatch Resolution codec and the Optimization procedure in the TFO protocol to detect incompatible codecs. When the TFO detects codec incompatibility, you can activate a codec modification procedure to modify the codec used by the terminals. The TFO specifies the rules for resolving the codec disagreement (that is, which codec to select). The TFO protocol then activates the TICC signaling to modify the codec used in the call, in order to allow incompatible codecs in the two end terminals. A change from the codec initially suggested by the startup UMTS network may require a modification of the bearer level connection established in any UMTS network.

The above-described embodiments unnecessarily minimize the transcoding of the conversation due to intermediate PCM networks as well as allow the optimal location of the user plane equipment (e.g., transcoding units) and / or the user plane resources (e.g., bandwidth) in order to support the service level of a particular call on public telecommunication networks.

5 Codec type negotiation will be required in existing networks and in new networks that only support voice coded by PCM in order to minimize unnecessary transcoding in the conversation path. The quality of the conversation between UMTS and GSM subscribers will not be unnecessarily degraded when their calls go through existing PCM core networks that have TFO support. When the GCN mechanism is introduced in the TICC protocol, it is likely to be transported by a transparent method (APM

10 User.

It will be apparent to the person skilled in the art that various modifications can be made to the embodiments described above without departing from the scope of the present invention.

Claims (10)

1. A method of negotiating protocol options between the first (A), second (B) and third (C) nodes in a telecommunications network using separate Call Control and Bearer Control protocols, the method comprising: transmit a first call control message from the first node (A) to the second node (B) specifying the protocol options supported by the first node (A), transmit a second call control message from the second node (B) to the third node (C) specifying the protocol options supported by the two nodes first (A) and second (B); Y select a protocol option from the protocol options specified in the second control message.

2.

 A method according to claim 1, wherein said step of selecting a protocol option from the protocol options specified in the second control message is carried out by the third node (C).

3.

A method according to claim 1 or 2, wherein each of said call control messages includes a level of preference associated with each of the specified protocol options.

Four.

A method according to any one of claims 1 to 3, wherein the first node (A) is a starting node.

5.

A method according to any one of claims 1 to 4, wherein the third node (C) is an end node.

6.

A method according to any one of claims 1 to 5, which also includes determining whether the carrier level between the first (A) and second (B) nodes is affected by said selection stage, and whether the carrier level is affected by taking actions to modify the bearer level parameters between the first and second nodes.

7.

A method according to any one of claims 1 to 6, which also includes determining whether the bearer level between the second (B) and third (C) nodes is affected by said selection stage, and whether the bearer level It is affected by taking actions to modify the bearer level parameters between the second and third nodes.

8.

The method of claim 1, wherein the first call control message corresponds to a first list of options supported by the first node (A), and wherein the second call control message corresponds to a second list of calls. options that are generated by deleting the options of the first list that are not supported by the second node (B).

9.

The method of claim 3, wherein selecting a protocol option from the protocol options specified in the second control message is carried out by analyzing the options received with their associated preference levels and selecting the option with the preference level indicated higher.

10.

The method of claim 2, wherein the third node (C) returns a message to the second node (B), which includes the selected protocol option and wherein said second node (B) in turn sends a message to the first node (A) specifying the protocol option selected.