JP5118757B2 - Function negotiation in telecommunications networks - Google Patents

Abstract

A method of negotiating a call capability between signaling points in a telecommunications system. The method comprises sending a capability preference or prioritised list of preferences from an originating signaling point to a terminating signaling point or signaling transfer point, at the call control level. A capability acceptance is returned from the terminating signaling point or signaling transfer point to the originating signaling point at the Call Control level, if the terminating signaling point or signaling transfer point accepts a preference sent by the originating signaling point.

Description

  The present invention relates to functional negotiation in a telecommunications network, and more particularly to negotiation of a suitable call codec, although this is not the case in all cases.

  Currently, telecommunications networks are mostly in the signaling system no. 2 as mechanisms for controlling call connections and for handling the transfer of signaling information between the signaling points of the network. 7 (SS7). Typically, one or more application parts and user parts of any signaling point utilize SS7 to communicate with peer application parts and user parts of another signaling point. Examples of user parts are ISUP (ISDN User Part) and TUP (Telephony User Part), and examples of application parts are INAP (Intelligent Network Application Part) and MAP (Mobile Application Part). The conventional SS7 protocol stack includes message transfer parts MTP1, MTP2, and MTP3, which handle the format of signaling messages transported through the physical layer and various routing functions.

  Recently, there is significant interest in the telecommunications community to use non-standard (ie, different from the telecommunications industry conventional) signaling transport mechanisms in place of traditional SS7 mechanisms in telecommunications networks. This reason is related to both efficiency gains and potential cost savings. For example, the transport of signaling information between signaling points using an Internet Protocol (IP) network has been actively studied. An IP network has the advantage that the transmission resources can be used effectively by using packet switching, and that this technology is widespread (compared to a special telecommunications technology) and is low in cost. There is also interest in using other transport mechanisms such as AAL1 / 2/5, FR.

  The ISUP standard handles call connection setup and control in telecommunications networks and is closely related to the SS7 signaling transport mechanism. Therefore, it is not suitable for using other non-standard transport technologies such as IP and AAL2. For this reason, several standardization organizations such as ITU-T, ETSI, and ANSI are currently studying specifications that are signaling protocol specifications related to call control and that are independent of the underlying transport mechanism. This means that a protocol different from the protocol that is a bearer control function is used. Bearer control only establishes “pipe” parameters (including start and end points) to be used when transporting user plane data between nodes and is limited to the transport mechanism. ing. On the other hand, a new protocol called TICC (Transport Independent Call Control) has a call control function. This functionality includes services activated to handle calls between callers and callers (eg, call forwarding), services related to general routing of user plane data, and the like.

  As a result of building a new network architecture by separating the call level and the bearer control level, an open interface was created between the call control entity and the bearer control entity. In this interface, the two entities are called a media gateway controller and a media gateway, respectively. In the following description, this open interface is called X-CP. Examples include IETF MEGACO work and ITU Research Committee 16 (SG16) H.264. 248 work.

  Traditionally, fixed telephone networks use pulse code modulation to transport user plane data (voice, facsimile, etc.) between network nodes. On the other hand, current cellular networks use one or more coders / decoders (called “codecs”) to compress voice signals for effective transmission over the atmospheric interface and within cellular networks. ing. If a telephone call connection spans two networks (or terminals) and they support different call codecs or multiple call codecs, negotiation can be made between the terminals to determine the appropriate codec . If this negotiation is not performed, code conversion (that is, conversion from one call code to another call code) may have to be performed at an interface between networks. Since code conversion uses a large amount of resources, the call quality is extremely lowered and processing time is delayed. Therefore, codec negotiation is the preferred option.

  In addition to codec negotiation, other functions and parameters may have to be negotiated in conventional telecommunications networks. For example, it may be better to negotiate security functions such as voice encryption and data encryption between terminals or nodes of a telecommunication network.

  According to a first aspect of the present invention, there is provided a method for negotiating a call function between signaling points in a telecommunications system, wherein a function preference or prioritized preference list from an originating signaling point to a terminating signaling point or a signaling transfer point And at the call control level, and if the terminating signaling point or signaling transfer point accepts the preference sent by the originating signaling point, call function acceptance from the terminating signaling point or signaling forwarding point to the originating signaling point. And returning the method.

  In some cases, an acceptance message may not be returned, such as when the terminating signaling point or signaling transfer point does not accept the feature preference (or any of the preference lists) sent by the originating signaling point, in which case the default feature Is assumed by both points. Alternatively, a default message indicating that the default codec is used may be returned. If none of the codecs reach agreement, the call can be released because the network is incompatible.

  The present invention is particularly suitable for negotiating call codec functionality between signaling transfer points in different telecommunications networks. For example, in a Japanese telecommunications system, the present invention can be used to negotiate whether to use VSELP, PSI-CELP, or μ-law coding (μ-law coding is the default coding). However, the present invention can also be applied to negotiate other functions such as security functions (such as voice encryption and data encryption).

  Although TICC can be used as a protocol for executing the negotiation, a specific protocol (that is, user plane function negotiation protocol) can also be used at the CC level.

  When the call control level and the bearer control level are controlled by different protocols and the selection affects the bearer level, the signaling point responds to the selection of the function at the call control level by notifying the bearer control level. If it is appropriate to send a notification at the bearer level between bearer switching points, it can be sent later to establish an appropriate bearer level resource.

  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 present at the bearer control level.

  Although certain options supported for bearer control level functionality are also known at the call control level, the call control level does not necessarily know that these options at the bearer level are currently available. . Thus, preferably, when a feature preference or prioritized preference list is received at a terminating signaling point or signaling transfer point, the call control level performs negotiation with the bearer control level to determine option availability at the bearer control level. More preferably, this negotiation is performed between the call control level media gateway controller and the bearer control level media gateway.

  When a prioritized preference list is sent from the originating signaling point to the media gateway controller, the controller preferably modifies the list to remove preferences that are known not to be supported by the corresponding 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 secure the resources necessary for the preference content and notify the media gateway controller of the preference content. The media gateway controller can now return the function acceptance to the originating media gateway controller.

  According to a second aspect of the present invention, a signaling point arranged to negotiate a call function with another signaling point in a telecommunication system, wherein the function preference or prioritized preference list is transmitted as a termination signaling point or signaling transfer. The means for transmitting to the point at the call control level and the function acceptance sent when the terminating signaling point or signaling forwarding point accepts the preference sent from the originating signaling point are called from the terminating signaling point or signaling forwarding point. And a means for receiving at a control level.

  According to a third aspect of the present invention, a media gateway controller of a telecommunications system receives a function preference or prioritized preference list for connections set up via the telecommunications system from peer media gateway controllers. Means for communicating with the media gateway corresponding to the media gateway controller and determining the availability of the preference content received at the media gateway; and a function preference acceptance message to the peer media gateway controller according to the availability determined at the media gateway. And a means for returning the media gateway controller.

  According to a fourth aspect of the invention, a media gateway of a telecommunications system, means for receiving from a media gateway controller a function preference or prioritized preference list for connections set up via the telecommunications system; There is provided a media gateway comprising: means for selecting preference contents according to availability of preference contents in the media gateway; and means for transmitting the selected preference contents to the media gateway controller.

  According to a fifth aspect of the present invention, there is a method for negotiating protocol options between first, second, and third nodes in a telecommunications network using separate call control protocols and bearer control protocols. A first call control message specifying protocol options supported by the first node from the first node to the second node, and a protocol supported by both the first and second nodes. Transmitting a second call control message specifying the option from the second node to the third node, and selecting a protocol option from the protocol options specified in the second control message. A featured method is provided.

  The step of selecting a protocol option from the protocol options specified in the second control message can be performed at the third node.

  Each call control message may include a preference level corresponding to each designated protocol option.

  The first node can be the originating node and the third node can be the terminating node.

  The method determines whether the bearer level between the first node and the second node is affected by the selecting step, and if the bearer level is affected, the first node and the second node The method may further include performing an action to modify a bearer level parameter between

  The method also determines whether the bearer level between the second node and the third node is affected by the selecting step, and if the bearer level is affected, the second node and the third node A step of modifying a bearer level parameter with the other nodes may be further included.

  According to a sixth aspect of the present invention, there is provided a method for setting up a call call connection in a telecommunications system in which the call control protocol is independent of the bearer transport mechanism, the outgoing signaling point of the system and the first termination signaling. Negotiating a first call codec with a point and establishing a call connection via a transport mechanism between an originating signaling point and the first termination signaling point according to the first call codec And then negotiating a second different call codec between the first termination signaling point and the second new termination signaling point, and notifying the originating signaling point of the second call codec Step and outgoing Establishing a call connection between a nulling point and the second termination signaling point, wherein the first call signaling point acts as an intermediate signaling point, and the first call connection is the second call connection point And a step modified as needed to support a call codec.

  Embodiments of the present invention facilitate the transfer of call connections between different terminating signaling points by modifying the connection between the originating signaling point and the original terminating signaling point to reflect the new codec. . The final end-to-end connection is completed by establishing a connection between the original termination signaling point and the new termination signaling point according to the new codec. Preferably, even if not necessary, the latter connection is established after modifying the first connection.

  Further, the method of the present invention is applied only when the first call codec is not obtained as a result of the call codec negotiation performed between the original termination signaling point and the final termination signaling point. If the result is the first call codec, it may not be necessary to modify the original connection.

  Preferably, notifying the originating signaling point of the second call codec includes transmitting an appropriate call control (CC) message from the original terminating signaling point to the originating signaling point. This CC message may be a “correction request message”.

  Preferably, the call control protocol is a transport independent call control (TICC) protocol.

  The present invention can be applied to the evolution of existing telecommunications networks such as mobile networks based on GSM, DAMPS, PDC, etc., and can also be applied to future generation networks such as UMTS.

  According to a seventh aspect of the present invention, there is provided a telecommunications system in which a call control protocol is independent of a bearer transport mechanism, wherein a first communication point between a system outgoing signaling point and a first termination signaling point is Means for negotiating the first call codec, means for establishing a call connection via a transport mechanism between the originating signaling point and the first terminating signaling point according to the first call codec, Means for later negotiating a second different call codec between the terminating signaling point and the second new terminating signaling point; means for notifying the originating signaling point of the second calling codec; and originating signaling point And the second termination signaling point Means for establishing a call connection, wherein said first outgoing signaling point operates as an intermediate signaling point, and wherein said first call connection is modified as necessary to support said second call codec And a system characterized by comprising:

  According to an eighth aspect of the present invention, there is provided a telecommunications network signaling point whose call control protocol is independent of the bearer transport mechanism, wherein the first termination signaling point and the first call codec are negotiated Means for establishing a call connection via a transport mechanism between an originating signaling point and the first terminating signaling point according to the first call codec, the first terminating signaling point and a second Means for receiving a notification of a second call codec negotiated with a new termination signaling point, and means for modifying said first call connection as needed to support said second call codec And providing a signaling point characterized by comprising: .

  According to a ninth aspect of the present invention, there is provided a method for setting up a call connection between a first and a second mobile telephone network, wherein either network is located outside the radio access network portion of the mobile network A method including a tandem free operation (TFO) device, wherein a call control protocol is independent of a bearer transport mechanism, and the TFO device of one mobile network and the other mobile to determine an appropriate call codec Performing negotiation with a peer TFO device of the network, and notifying the determined call codec to the radio access network by transmitting a call control (CC) message from the corresponding TFO device. It is characterized by including The law provides.

  Each mobile network may include a tandem free operation (TFO) device outside the radio access network portion of the mobile network. However, this is not always the case, and in either mobile network there may be a TFO in the radio access network.

  The two mobile phone networks can be connected to each other via the PSTN. Preferably, the negotiation is performed using a TFO protocol codec mismatch resolution and optimization procedure, and the TFO message is transmitted using in-band signaling.

  Preferably, the or each TFO device installed outside the radio access network is installed in a gateway MSC (GMSC) that provides an interface between a mobile network and an external network (such as PSTN).

  The present invention is particularly suitable for setting up a call connection between subscribers, any of which are subscribers of a Universal Mobile Telecommunications System (UMTS) network. In this case, the radio access network is preferably a UMTS terminal terrestrial radio access network (UTRAN), and the TFO device of that network is installed at the boundary of the UMTS core network.

  According to a tenth aspect of the present invention, an apparatus for setting up a call connection between a first and a second mobile telephone network, wherein at least one of the networks is installed outside a radio access network portion of the mobile network. In order to determine the appropriate call codec, the TFO device of one mobile network and the other are used to determine the appropriate call codec, including a tandem free operation (TFO) device and a call control protocol independent of the bearer transport mechanism. Means for performing negotiation with a peer TFO device of the mobile network, means for notifying the determined call codec to the radio access network by transmitting a call control (CC) message from the corresponding TFO device; Characterized by comprising To provide that equipment.

  According to an eleventh aspect of the present invention, there is provided a method for setting up a call connection between first and second mobile telephone networks, each network being tandem free installed outside the radio access network portion of the mobile network. An operation (TFO) device, wherein the call control protocol is independent of the bearer transport mechanism, and both mobile networks are connected via a public switched telephone network (PSTN), comprising a suitable call codec To negotiate between two mobile terminals subscribing to the first and second mobile networks respectively, the negotiation being exchanged between the radio access network part and each TFO device Call control protocol signal Provides a method characterized by comprising the steps, performed using an ISUP message sent between grayed messages and TFO device.

  For example, a list of codecs available on the originating mobile network can be transmitted from the TFO device of that network to a peer TFO device using an ISUP initial address message (IAM). The message may further include a codec that the originating mobile network prefers. Subsequent ISUP messages sent in the reverse direction indicate the codec type selected by the terminating mobile terminal. In order to be able to use this negotiation procedure, it may be necessary to modify the ISUP standard.

  According to a twelfth aspect of the present invention, an apparatus for setting up a call connection between a first and a second mobile telephone network, each network being tandem free installed outside a radio access network portion of the mobile network An operation (TFO) device, the call control protocol is independent of the bearer transport mechanism, and both mobile networks are connected via a public switched telephone network (PSTN), and the appropriate call codec To negotiate between two mobile terminals subscribing to the first and second mobile networks, respectively, the negotiation being exchanged between the radio access network part and each TFO device Call control protocol signaling Means performed using ISUP messages sent between the message and TFO devices, provides an apparatus, characterized in that it comprises a.

FIG. 2 shows a plurality of signaling points in a telecommunications network. FIG. 6 is a diagram illustrating a signaling flow (Case A) between the signaling points of FIG. 1 according to the first embodiment of the present invention and a signaling flow (Case B) between the signaling points of FIG. 1 according to the second embodiment of the present invention. . 1 illustrates a telecommunications network including a media gateway controller and a media gateway. FIG. FIG. 4 is a flow diagram illustrating a generic function negotiation process used in the network of FIG. It is a figure which shows the signaling flow in which an intermediate node is related to the negotiation between a transmission node and a termination | terminus node. It is a figure which shows the signaling flow in which an intermediate node is related to the negotiation between a transmission node and a termination | terminus node. FIG. 6 shows another set of signaling flows in which an intermediate node is involved in negotiation between a source node and a terminal node. FIG. 6 shows another set of signaling flows in which an intermediate node is involved in negotiation between a source node and a terminal node. 1 schematically illustrates a telecommunications system of known design. It is a figure which shows a telecommunications system typically. FIG. 9 is a diagram illustrating a first signal set according to the first embodiment of the present invention related to codec negotiation in the system of FIG. 8; FIG. 9 is a diagram illustrating a second signal set according to the second embodiment of the present invention related to codec negotiation in the system of FIG. 8;

  FIG. 1 shows a portion of a telecommunications network that includes two signaling points, referred to as node A and node C in the following description. These nodes are telephone exchanges or exchanges, and may belong to the same network operator or different network operators. In the following example, node A represents the calling node to which the caller (not shown) is connected, and node C represents the end node to which the caller is connected. Each signaling point is composed of a call control (CC) portion and a bearer control (BC) portion. That is, the call control function and the bearer control function are separated into two different protocol layers. The CC portion forms the call control level and performs functions such as call forwarding and other routing and control functions. The BC part establishes a pipe between the BC parts, determines its size, and transports user plane data.

  The BC level will be described. This level is configured by a bearer network such as an IP network. Although only one bearer switching point (Node B) is shown in FIG. 1, there are one or more bearer switching points in the IP network. In the case of an IP network, these bearer switching points are IP routers. When the bearer network is an ATM network or an AAL2 network, the bearer switching point is an ATM switch or an AAL2 switch, respectively.

  When the caller initiates a call, such as picking up the handset, Node A, the originating signaling point, receives information at the CC level from the source (caller). This information defines bearer resource requirements. The originating signaling point determines the available option list that needs to be negotiated with the terminating signaling point, Node C, based on either (or both) the source information or the signaling point and its home network capabilities.

  Next, node A inserts the available option list into the message and sends a signal to the CC part negotiating a specific function (node C in FIG. 1). This message indicates the preference level of each option.

  Node C uses the priority level specified by node A for each option in the list and selects the option with the highest preference level among the supported options. That is, node C selects only the highest preference option, even if node A supports another option in the list that shows a high preference. Node C sends the selected option to node A.

  If the result of this negotiation affects the bearer connection level, an appropriate action is performed at the bearer level so as to conform to the result of the negotiation. This is done with the BC protocol. There are two cases: forward bearer correction (Case A in FIG. 2) and reverse bearer correction (Case B in FIG. 2).

  An example of a function that can be negotiated using the above method is a codec function. Generally, in a radio access network, calls are transcoded to PCM. This is because this is the only call format that can be used in a conventional fixed telephone network. However, since the code conversion process drastically degrades the call quality, some cellular standards (GSM, PDC, etc.) have two compatible terminals (for example, two GSM terminals used in a common codec). The specification determines how to avoid transcoding once the connection is established. Although most current mobile terminals support several codecs, these methods also allow codec negotiation, but have some significant deficiencies compared to the generic negotiation mechanism shown below.

  The call codec is tightly coupled to the mobile environment (in PDC, codec negotiation is a service of the MAP (Mobile Application Part) protocol) or has not optimized the use of hardware and transmission resources (in GSM, Codec negotiation is part of tandem free operation). Regardless of whether or not code conversion is actually performed, two code conversion units (TRAU) and a 64 Kbps channel are always allocated for call connection. Currently, various mobile standards deal with this problem in different ways, and so far there has been no possibility of adjusting these solutions.

  Another example of a function that needs to be negotiated is a security function (voice encryption, data encryption, etc.). Currently, this is a significant concern in public telecommunications networks and will become even more important in the near future. However, there are several ways to protect a user's plain information from unwanted third parties. There are a plurality of encryption algorithms, and data encryption algorithms currently being developed and new algorithms are being introduced one after another. Therefore, a method for negotiating security functions in a public telecommunications network will be needed in the near future, and this method is provided by the present invention.

  The above description generally refers to the BC level and the CC level. FIG. 3 illustrates in more detail the telecommunications system divided into CC level and BC level. The CC level includes a plurality of media gateway controllers (MGC_A, MGC_B, MGC_C), and the BC level includes a plurality of media gateways (MG_1 to MG_6). The first pair of media gateways (MG_1, MG_6) is a bearer connection via an AAL2 network, and the second pair of media gateways (MG_2, MG_5) is a bearer connection via an IP network. The third pair of media gateways (MG_3, MG_4) is a bearer connection via the STM network. As is apparent from FIG. 3, each media gateway controller is arranged to control two or three media gateways via open interfaces (X-CP_1 to X-CP_3).

  As already explained above, function negotiation takes place at the CC level, ie between the media gateway controllers. Furthermore, the X-CP is also used as a means for the media gateway to “notify” the media gateway controller that controls the media gateway to support the functions and options (eg, function = compressed call, option = codec list). ). Also, the media gateway can participate in the negotiation via the X-CP, so that the function selection depends not only on the functions specified in the gateway but also on whether these functions are currently available. Otherwise, there is a risk that the media gateway does not have resources to support the options that the media gateway controller negotiates for the media gateway. For example, consider the case where a media gateway controller needs to provide an intelligent network (IN) service for calls and this service requires in-band announcement information. If the media gateway controller alone selects a codec according to the knowledge of the codec supported by the media gateway, the media gateway may not currently have TRAU available for this codec. Therefore, the media gateway rejects this connection and the call is aborted.

  To illustrate the solution to this problem, consider a case where a media gateway controller receives a connection setup message from a peer media gateway controller that includes an option list for a specific negotiable feature. The message indicates the preference level for each option. The receiving media gateway controller reads the received list and removes from the list any options that are known not to be supported by the media gateway selected for the connection. The media gateway controller then sends the modified option list to the media gateway via the X-CP interface.

  The receiving media gateway selects the option with the highest priority and which the gateway can currently support. The media gateway then acquires the necessary resources to support this option and sends the option over the X-CP interface to the media gateway controller. Upon receipt of the selected option, the media gateway controller sends the selected content to the peer media gateway controller as described above.

  The generic function negotiation described above will be described in more detail with reference to the flowchart of FIG.

  In many countries, merchants are required to intercept calls for legal purposes from the country's authorities. In order to enable legitimate interception, the merchant usually has to user plane data (ie actual calls or other data transmitted by so-called “pipes” performed at the bearer level) for each call in the network. Require access to everything. With the introduction of TICC, user plane data can be transported via various techniques (ATM, IP, etc.), thereby enabling transport of encrypted user plane data. Therefore, if the security and codec (encryption and decryption) characteristics of the user plane data cannot be determined at the legitimate intercept reception point, the call cannot be intercepted normally. Therefore, the merchant needs knowledge of the codec and encryption (security) algorithm used in each BC section of the call (between two BC nodes).

  On the other hand, the code conversion and conversion processing of the security algorithm greatly reduces the service quality of the call and generates an extra delay. Therefore, it is desirable to reduce the number of conversion points required end-to-end for each call (avoid if possible) and reduce the conversion of security algorithm points.

  A mechanism for solving this problem will be described with reference to the operation sequence of FIG. The call control part of node A (see FIG. 1) sends a message to the call control part of node B. This message includes a list of options supported by node A and the preference level for each option. This mechanism is particularly suitable for use with TICC, but can also be used with independent protocols specially adapted for performing negotiations. This is indicated by the initial UPCN (User Part Capability Negotiation) in FIG.

  Next, the call control side of node B sends a message to the call control side of node C. This includes a list of options supported by both node A and node B and the preference level for each option. Node C selects the support option with the highest preference level and returns a message containing the selected option to Node B. Upon receiving this message, node B sends a message specifying the selection option to node A.

  Next, FIG. 5 shows two operation sequences labeled Case A and Case B. In case A, if the bearer level is affected by the call control level negotiation described above, node A performs the appropriate action at the bearer level. For example, node A may have to change the size of pipes needed to support the selected option. A BC message is then sent from node A to node B, specifying which parameters are required. Node B then analyzes the selected option and, if the bearer level is affected by the negotiation, performs the appropriate action at the bearer level. The BC message is then sent from node B to node C, specifying between the node B and node C what parameters are required at the bearer level.

  In case B, the bearer level operation sequence starts at node C and ends at node A. Node C analyzes the selected option and performs the appropriate processing at the bearer level if the bearer level is affected by the negotiation. A BC message is then sent from node C to node B, specifying the necessary parameters. Node B analyzes the selected option and performs the appropriate processing at the bearer level if the bearer level is affected by the negotiation. A BC message is then sent from node A to node B, specifying the necessary parameters.

  Case A and case B are alternatives, but other cases are possible. In this regard, the BC protocol between node A and node B may be different from the protocol between node B and node C. For example, the protocols may be ATM and IP, respectively. Even if it is necessary to modify the bearer level connection parameter between one node pair, it may not be necessary to modify between the other node pairs.

  The above negotiation can be performed at the time of call establishment or at the time of call. The latter is applicable, for example, when the user tries to start using the cipher when calling.

  This procedure can reduce the number of points in the network that need to change the generic function. In particular, by using the mechanism specified above, it is possible to minimize the conversion processing of the number of code conversion points or security algorithm points for end-to-end calls over a plurality of CC sections. This solution can be applied to calls across one or more networks.

The GCN mechanism also includes the function of incorporating the option list and its preference level into the initiating CC node, and having the end node select a support option using the preference level indicated by the originating node. The negotiation mechanism described later is useful only when negotiation is performed between two CC nodes. In the proposed adaptation method, this GCN mechanism is extended when a plurality of CC nodes intervene in the negotiation. In other words, this is a case where the call extends over a plurality of CC nodes belonging to one or a plurality of telecommunications carriers. This adaptation method includes the following operation sequence.
1. The initiating CC node sends a list of support options and a preference level corresponding to each option.
2. The relay CC node analyzes the received option list, removes the unsupported option from the list, and forwards the list to the next node.
3. The terminating CC node analyzes the received option list and the corresponding priority and selects the highest priority support option shown.

  In the case of the legitimate intercept, the security and coding characteristics of the user plane data can be received at the legitimate intercept reception point (that is, the node where the intercept is performed) so that the call can be intercepted normally.

In the proposed new architecture shown in FIG. 1, the call control protocol is independent of the transport mechanism. In this case, the user plane transport connection cannot be established for the call unless the TICC performs codec negotiation for the call. The reason is that the amount of transport resources required to support a call depends on the codec selected for that call (ie, the result of the codec negotiation procedure). In some cases, the codec is initially selected for the call, but due to environmental changes, this codec must be changed to a new codec in the later phase of the call. Two important cases in which this situation occurs are as follows.
1) An interaction with an intelligent network (IN) service that makes announcement information and re-forwards the call to the called party, either automatically or according to input from the calling party. In order to perform notification information, it is necessary to select the first codec. Later, the call is forwarded / re-forwarded to the other side and the second codec is used.
2) This is a so-called supplementary service called CFNR (Call Forwarding on No-Reply). After negotiating the first codec between the A side and the B side, the call is forwarded to the C side so that B does not answer within a predefined time. The second codec is required according to C side requirements.

  In the above cases 1) and 2) and other related cases, it is necessary to establish a call connection with the transport mechanism by the first selected codec, and to perform notification information, signal sound inspection, and the like. If the codec is changed later, the call connection may need to be modified to support the new codec.

  In the following, referring to FIG. 6, a signaling process used in a network using the TICC protocol and capable of establishing a telephone call end-to-end telephone connection will be described. Here, the call is generated at node A, the originating signaling point, and is initially transferred to the first node, Node B, or the signaling point. This call is then forwarded to another termination signaling point, Node C. The illustrated process relates in particular to the invocation of the IN service, in which case the caller is first connected to the IN network node, which plays the prerecorded message to the caller and then terminates the call at the final termination. It is transferred to the signaling point. The illustrated process is also related to CFNR service, in which case the caller connected to the first terminating signaling point forwards the call to the new terminating signaling point if it does not answer the call.

The signaling sequence is composed of the following sequential steps.
1. Node A establishes a call to Node B. Codec X is selected for this call.
2. A transport connection having transport resources appropriate for the selected codec is set up between node A and node B.
3. Node B forwards / reforwards the call to Node C. Node C does not support codec X and selects codec Y.
4). Node B requests Node A to modify the codec selection for this call from codec X to codec Y.
5. The transport connection between the node A and the node B is adjusted to the codec Y as necessary. FIG. 6 shows a case where the correction to the transport connection is made in the forward direction and an alternative case where the correction is made in the reverse direction.
6). A transport connection suitable for codec Y is established between node B and node C.
7). The TICC completes the call establishment (TICC ACM + TICC ANM).

  Since TICC is based on ISUP, the signaling message name is taken from ISUP. However, since the ISUP does not include a codec negotiation procedure or a codec modification procedure, a new TICC message pair (“TICC modification. Request / confirm”) is required to perform the codec modification function.

  FIG. 6 shows that only the corrections made for the transport connection from node A to node B are made in the forward or reverse direction, but other transport connection modifications are also made in the forward or reverse direction. be able to. Transport connections can also be established in the forward or reverse direction.

  Using the mechanisms described above, call calling connections between four or more signaling points or nodes can be established. For example, between the originating signaling point Node A and the original terminating signaling point Node B, or the original terminating signaling point Node B and the final terminating signaling point Node C (or its) Both) there may be one or more relay nodes. The call can also be forwarded or relayed from node C to yet another signaling point (ie, node D). However, in this case, it is necessary to negotiate to determine whether the codec Y is appropriate for the node D. If not, the transport between the node A and the node B and between the node B and the node C is necessary. The connection may need to be modified. This process can be extended to any number of nodes.

  A conventional public switched telephone network (PSTN) digitally encodes call data for transfer using pulse code modulation (PCM). Digital mobile telephone networks, on the other hand, use more advanced coding techniques such as CELP and adaptive multi-rate (AMR) coding to achieve higher compression rates than PCM. In many mobile networks, call coding and decoding are performed by the mobile terminal itself. When a call is made between two mobile terminals registered in the same network, it is possible to transmit coded call data end to end.

  When a call is placed from a mobile terminal registered in the mobile network to a terminal that is a subscriber to an “external” network, the end-to-end transmission of the coded call data is dependent on the characteristics of the external network and the originating mobile network. Depending on the nature of the intermediate network connecting to the external network, this may not be possible (of course, this is also true when the call originates on the external network).

  Considering the telecommunications system of FIG. 7, this system is composed of two third generation universal mobile communication (UMTS) networks 1 and 2, which are the conventional PSTN / ISDN network 3. It is connected by. Each of the UMTS networks 1 and 2 includes a UMTS terminal terrestrial radio access network (UTRAN) 4, which is composed of a radio network controller (RNC) 5 and a radio base station (BTS) 6. The UTRAN 5 delivers compressed call data between a mobile terminal (not shown) and a mobile switching center (MSC) 7 that transfers the input call connection and the output call connection.

  Assume that a call originates from one UMTS network 1 subscriber and is sent to another UMTS network 2 subscriber. The call is forwarded via PSTN 3 using each relay node 8 of UMTS networks 1 and 2 (which is either a gateway MSC (GMSC)). As already explained above, PSTN 3 encodes the call data using PCM. It is important that the call data transferred via the PSTN 3 is in a format that can be understood by the network. For example, it is necessary for PSTN3 to insert the notification information of the merchant in the call and to execute a voice prompt service, etc., and also to allow the PSTN3 merchant to monitor the call for security purposes. It is. Therefore, it is necessary to “code convert” the call data at the GMSC 8 of the UMTS networks 1 and 2 before passing the data to the PSTN 3. That is, the call data is converted from the mobile network call coding format to PCM. Similarly, PCM data received by the GMSC 8 needs to be converted into a corresponding mobile network call coding format.

  When code conversion is executed, a large amount of processing resources are consumed in the GMSC 8, and a decrease in call quality is conspicuous. To compensate for some of these shortcomings, a tandem free operation (TFO) device can be introduced into the call connection in GMSC8. The outgoing call data is still converted to PCM, but the least significant bit of each PCM sample is “wiretapped” by the TFO device. The channel bandwidth formed by these eavesdropping bits is sufficiently large (ie, 8 Kbit / s) to transmit the original encoded data. The TFO device in the terminating UMTS network assembles the coded data and sends it to the corresponding UTRAN, but the received PCM data is discarded (the data has been modified by the PSTN, such as the addition of vendor notice information) Unless it is). In this way, TFO allows PSTN 3 to use PCM data, but end-to-end transmission of effectively coded call data can still be performed.

  When an intermediate device in the PSTN / ISDN changes the PCM bitstream, the TFO device detects the change and “falls back” to pass the PCM coded call between the TFO devices. That is, after that, compressed audio data is not delivered.

  The call codec that can be used by the mobile network differs depending on the characteristics of the network and, in some cases, the characteristics of the terminal using the network. Obviously, a single codec can be used end-to-end only when both networks use the same codec. Assuming that the GMSCs of the two mobile communication networks have information on the codec capabilities of the connected network, they can negotiate and agree to use a common codec. Indeed, the ETSI recommendation GSM 08.62 (version 7.0.0, release 1998) provides a suitable protocol.

  When the TFO device is installed in the boundary part of the mobile network, that is, outside the UTRAN as shown in FIG. 7 (TFO device is conventionally installed in the radio access network) A problem occurs. At present, there is no exchange information regarding codecs negotiated between TFO devices and between TFO devices and radio access networks in such networks.

  FIG. 8 shows a modified system architecture in which two UMTS network GMSCs incorporate a TFO device 9. On the other hand, FIG. 9 shows signaling for call setup between two nodes in each UMTS mobile telephone network. Node 1 and node 4 represent the MSC, and the two intermediate nodes 2 and 3 represent the GMSC.

  Call setup signaling within the UMTS network is performed at the call control (eg, TICC) level. This call setup is initiated by an initial address message (IAM) sent from the MSC to the corresponding GMSC. This IAM uses a generic function negotiation (GCN) mechanism to determine the number of parameters for the call connection. In particular, the IAM includes a codec list and preferred codecs supported by the originating UMTS network. The originating GMSC selects a codec from the transmitted codec list and returns the selected content to the MSC in the form of “message (selected codec 1)”. The call connection is then established with a bearer level (eg, AAL2 or IP) with sufficient bandwidth to support the selected codec.

  In order to be able to perform end-to-end codec negotiation, it has been proposed to add GCN to the ISUP protocol. By performing this end-to-end codec negotiation, the probability that both end points use the same codec type increases. If both endpoints use the same speech coding algorithm, the TFO can pass the compressed speech over the PCM network, and there is no degradation in speech quality due to unnecessary conversion processing. In addition, since the bearer requirements in the originating network and the terminating network that support compressed voice are also reduced (for example, AAL2 or IP bearer transport), the transmission amount can be reduced.

  FIG. 9 illustrates the use of a GCN Enhancement ISUP message to close a signaling “gap” between two UMTS networks. In a terminating UMTS network, an IAM is sent from the GMSC to the MSC using TICC. In this case, the MSC accepts the use of Codec 1 and sends back to that effect to the GMSC using the TICC. A bearer level connection is then established in the terminating UMTS network. The PSTN relays the codec acceptance to the originating GMSC. Since the originally proposed codec has been accepted, there is no need to change the bearer level connection in the originating UMTS network. However, when there is a change in the codec, it is necessary to transmit that fact from the originating GMSC to the MSC. As a result, the bearer level connection can be modified and the bandwidth of the connection can be expanded. In FIG. 9, ACM indicates an address complete message, and ANM indicates a response message.

  FIG. 10 illustrates a second embodiment of the present invention. This solution uses an optional codec mismatch resolution and optimization procedure provided in the TFO protocol to detect incompatible codecs. When the TFO detects codec incompatibility, a codec modification procedure can be initiated to modify the codec used by the terminal. The TFO specifies rules for resolving codec mismatches (ie, which codec to select). The TFO protocol then initiates TICC signaling, modifies the codec used in the call, and makes a compatible codec available at the two end terminals. If the originating UMTS network changes from the first indicated codec to another codec, the bearer level connection established in both UMTS networks may have to be modified.

  The embodiment described above minimizes unnecessary call code conversion by intermediate PCM networks and also supports user plane devices (eg, code conversion devices) to support specific call service levels in public telecommunications networks. ) Or user plane resources (eg bandwidth) (or both) allocations are optimized.

  If the existing network and the new network support only PCM coded voice to reduce unnecessary transcoding in the call path as much as possible, codec type negotiation is required. UMTS and GSM subscriber call quality is not necessarily degraded when a call passes through an existing PCM core network with TFO support. Incorporating the GCN mechanism into the TICC protocol is often done in a transparent manner (APM users).

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

Claims (21)

A method of negotiating protocol options between first, second and third nodes in a telecommunications network using separate call control protocols and bearer control protocols comprising:
And transmitting the protocol options the first node supports a first call control message to specify, to the second node from the first node,
Comprising the steps of both said second node and the first node and the second call control message specifying the support to Help Lot Col option, and transmits from said second node to said third node,
Selecting one of the protocol options specified in the second control message,
Determining whether a bearer level between the first node, the second node, and the third node is affected by the selected protocol option;
If it is determined that the bearer level between the first node and the second node is affected, modify a bearer level parameter between the first node and the second node; If a bearer level between a second node and the third node is determined to be affected, modifying a bearer level parameter between the second node and the third node;
A method comprising the steps of: The The method of claim 1, characterized in that it is performed in the third node to select a protocol option from the protocol options specified in the second control message. 3. A method according to claim 1 or 2, wherein each call control message includes a preference level corresponding to each specified protocol option. The method according to any one of claims 1 to 3, wherein the first node is a source node. The method according to any one of claims 1 to 4, wherein the third node is a terminal node. A method for setting up a call call connection in a telecommunications system in which the call control protocol is independent of the transport mechanism, comprising:
Negotiating a first call codec between an outgoing signaling point of the system and a first terminating signaling point;
Establishing a first call connection via a transport mechanism between an originating signaling point and the first terminating signaling point according to the first call codec;
Between the second terminating signaling point of the first end signaling point and the new, the steps to negotiate the second passage story codec different from the first speech codec,
And notifying the second speech codec to the originating signaling point,
Comprising the steps of establishing a second call connection between said second end signaling point and the originating signaling point, the first end signaling point acts as an intermediate signaling point, and wherein the step of establishing,
And correcting said first call connection if it is necessary to support the second speech codec,
A method comprising the steps of: The method of claim 6, further comprising a step that correct the second call connection Osamu based on the second speech codec. Said step of notifying the second speech codec to the originating signaling point, characterized by being performed by transmitting the first appropriate call control (CC) message from the terminating signaling point to the originating signaling point The method according to claim 6 or 7 . 9. The method of claim 8 , wherein the call control protocol is a transport independent call control (TICC) protocol and the CC message is a "modify.request" message. A telecommunication system wherein the call control protocol is independent of the transport mechanism,
Means for negotiating a first call codec between an outgoing signaling point of the system and a first terminating signaling point;
Means for establishing a first call connection via a transport mechanism between an originating signaling point and the first terminating signaling point according to the first call codec;
Between the second terminating signaling point of the first end signaling point and the new, the second through story codec different from the first speech codec, and means for negotiating later,
It means for notifying the second speech codec to the originating signaling point,
And means for establishing a second call connection between said second end signaling point and the originating signaling point, the first end signaling point acts as an intermediate signaling point, and wherein said means for establishing,
It means for modifying said first call connection if the second required to support a speech codec of,
A system comprising: A telecommunication network signaling point where the call control protocol is independent of the transport mechanism,
Processing means for negotiating a first call signaling codec with a first termination signaling point;
Means for establishing a call connection between the signaling point and the first terminating signaling point via a transport mechanism according to the first call codec;
It means for receiving a notification of the second speech codec that has been negotiated between the second terminating signaling point of the first end signaling point and the new,
If it is necessary to support the second speech codec, and means for modifying said call connection,
A signaling point comprising: A method for setting up a call connection between a first and a second mobile network, wherein at least one of the first and second mobile networks is located outside a radio access network. The call control protocol is independent of the bearer transport mechanism , comprising :
To determine the speech codec, performing a negotiation with the peer · TFO device T FO device and the other mobile network of one of the mobile network,
Call codec determined by negotiation between the TFO device and the peer TFO device if the determined call codec is incompatible with the call codec currently used in at least one of the mobile networks Informing the radio access network of the determined call codec by sending a call control (CC) message indicating:
Modifying a call codec currently used in at least one of the mobile networks;
A method comprising the steps of: The method of claim 12 , wherein each mobile network includes a tandem free operation (TFO) device outside a radio access network portion of the mobile network. Two mobile Rene Ttowaku The method of claim 12 or 13, characterized in that linked to one another via the PSTN. The negotiation is performed using a codec mismatch resolution and optimization procedure of TFO protocol, TFO messages according to any one of claims 12 to 14, wherein the sent using in-band signaling the method of. The or each TFO device installed outside the radio access network, any one of claims 12 to 15, characterized in that it is installed in the gateway MSC (GMSC) that provides an interface between the mobile network and an external network 2. The method according to item 1. Wherein the tandem free operation (TFO) devices mobile network that is installed outside the radio access network portion is Ri universal mobile communication system (UMTS) networks der, the radio access network is a UMTS terminal terrestrial radio access network The method according to any one of claims 12 to 16 . An apparatus for setting up a call connection between a first and a second mobile network, wherein at least one of the first and second mobile networks is a tandem free operation (TFO) apparatus installed outside a radio access network The device wherein the call control protocol is independent of the bearer transport mechanism,
To determine the speech codec, and means for performing a negotiation between peer · TFO device of the TFO devices and other mobile networks of one of the mobile network,
Call codec determined by negotiation between the TFO device and the peer TFO device if the determined call codec is incompatible with the call codec currently used in at least one of the mobile networks Means for notifying the radio access network of the determined call codec by transmitting a call control (CC) message indicating:
Means for modifying a call codec currently used in at least one of the mobile networks;
A device comprising: A method of setting up a call connection between a first and a second mobile network, each of the first and second mobile networks including a tandem free operation (TFO) device installed outside the radio access network. , call control protocol is independent of the bearer transport mechanism, both mobile network is linked via a public switched telephone network (PSTN), a said method,
To determine the speech codec, comprising the steps of performing a negotiation between the mobile terminals subscribing to the first mobile terminal that subscribes to a mobile network and the second mobile network, the negotiation It includes a ISUP messages sent between the TFO devices to notify the call control protocol signaling messages exchanged between the radio access network to determine the speech codec and the corresponding TFO device, the determined speech codec are executed using, said step of executing,
Modifying a call codec currently used in at least one of the mobile networks if the determined call codec is incompatible with a call codec currently used in at least one of the mobile networks;
A method comprising the steps of: A list of codecs available in the outgoing mobile network is transmitted from the TFO device of the network to the peer TFO device using an ISUP initial address message (IAM), and the selected codec is returned to the outgoing network in a subsequent ISUP message. The method according to claim 19 . An apparatus for setting up a call connection between a first and a second mobile network, each of the first and second mobile networks including a tandem free operation (TFO) apparatus installed outside the radio access network , call control protocol is independent of the bearer transport mechanism, both mobile network are connected via a public switched telephone network (PSTN), a said device,
Means for performing negotiation between a mobile terminal subscribing to the first mobile network and a mobile terminal subscribing to the second mobile network to determine a call codec , wherein the negotiation It includes a ISUP messages sent between the TFO devices to notify the call control protocol signaling messages exchanged between the radio access network to determine the speech codec and the corresponding TFO device, the determined speech codec are executed using the said means for executing,
Means for modifying a call codec currently used in at least one of the mobile networks if the determined call codec is incompatible with a call codec currently used in at least one of the mobile networks;
The apparatus characterized by including .

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